Adding upstream version 115.8.0esr.upstream/115.8.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

253 files changed, 35629 insertions, 0 deletions

diff --git a/third_party/libwebrtc/net/dcsctp/BUILD.gn b/third_party/libwebrtc/net/dcsctp/BUILD.gn
new file mode 100644
index 0000000000..8b38a65ca1
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/BUILD.gn
@@ -0,0 +1,26 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+if (rtc_include_tests) {
+  rtc_test("dcsctp_unittests") {
+    testonly = true
+    deps = [
+      "../../test:test_main",
+      "common:dcsctp_common_unittests",
+      "fuzzers:dcsctp_fuzzers_unittests",
+      "packet:dcsctp_packet_unittests",
+      "public:dcsctp_public_unittests",
+      "rx:dcsctp_rx_unittests",
+      "socket:dcsctp_socket_unittests",
+      "timer:dcsctp_timer_unittests",
+      "tx:dcsctp_tx_unittests",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/net/dcsctp/OWNERS b/third_party/libwebrtc/net/dcsctp/OWNERS
new file mode 100644
index 0000000000..06a0f86179
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/OWNERS
@@ -0,0 +1,2 @@
+boivie@webrtc.org
+orphis@webrtc.org
diff --git a/third_party/libwebrtc/net/dcsctp/common/BUILD.gn b/third_party/libwebrtc/net/dcsctp/common/BUILD.gn
new file mode 100644
index 0000000000..78fa0d307e
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/common/BUILD.gn
@@ -0,0 +1,67 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("internal_types") {
+  deps = [
+    "../../../rtc_base:strong_alias",
+    "../public:types",
+  ]
+  sources = [ "internal_types.h" ]
+}
+
+rtc_source_set("math") {
+  deps = []
+  sources = [ "math.h" ]
+}
+
+rtc_source_set("sequence_numbers") {
+  deps = [
+    ":internal_types",
+    "../../../rtc_base:rtc_numerics",
+  ]
+  sources = [ "sequence_numbers.h" ]
+}
+
+rtc_source_set("str_join") {
+  deps = [ "../../../rtc_base:stringutils" ]
+  sources = [ "str_join.h" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("dcsctp_common_unittests") {
+    testonly = true
+
+    defines = []
+    deps = [
+      ":math",
+      ":sequence_numbers",
+      ":str_join",
+      "../../../api:array_view",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:gunit_helpers",
+      "../../../test:test_support",
+    ]
+    sources = [
+      "math_test.cc",
+      "sequence_numbers_test.cc",
+      "str_join_test.cc",
+    ]
+  }
+}
+
+rtc_library("handover_testing") {
+  deps = [ "../public:socket" ]
+  testonly = true
+  sources = [
+    "handover_testing.cc",
+    "handover_testing.h",
+  ]
+}
diff --git a/third_party/libwebrtc/net/dcsctp/common/handover_testing.cc b/third_party/libwebrtc/net/dcsctp/common/handover_testing.cc
new file mode 100644
index 0000000000..1081766ea5
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/common/handover_testing.cc
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/common/handover_testing.h"
+
+namespace dcsctp {
+namespace {
+// Default transformer function does nothing - dcSCTP does not implement
+// state serialization that could be tested by setting
+// `g_handover_state_transformer_for_test`.
+void NoTransformation(DcSctpSocketHandoverState*) {}
+}  // namespace
+
+void (*g_handover_state_transformer_for_test)(DcSctpSocketHandoverState*) =
+    NoTransformation;
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/common/handover_testing.h b/third_party/libwebrtc/net/dcsctp/common/handover_testing.h
new file mode 100644
index 0000000000..396016afec
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/common/handover_testing.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_COMMON_HANDOVER_TESTING_H_
+#define NET_DCSCTP_COMMON_HANDOVER_TESTING_H_
+
+#include "net/dcsctp/public/dcsctp_handover_state.h"
+
+namespace dcsctp {
+// This global function is to facilitate testing of the socket handover state
+// (`DcSctpSocketHandoverState`) serialization. dcSCTP library users have to
+// implement state serialization if it's needed. To test the serialization one
+// can set a custom `g_handover_state_transformer_for_test` at startup, link to
+// the dcSCTP tests and run the resulting binary. Custom function can serialize
+// and deserialize the passed state. All dcSCTP handover tests call
+// `g_handover_state_transformer_for_test`. If some part of the state is
+// serialized incorrectly or is forgotten, high chance that it will fail the
+// tests.
+extern void (*g_handover_state_transformer_for_test)(
+    DcSctpSocketHandoverState*);
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_COMMON_HANDOVER_TESTING_H_
diff --git a/third_party/libwebrtc/net/dcsctp/common/internal_types.h b/third_party/libwebrtc/net/dcsctp/common/internal_types.h
new file mode 100644
index 0000000000..2354b92cc4
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/common/internal_types.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_COMMON_INTERNAL_TYPES_H_
+#define NET_DCSCTP_COMMON_INTERNAL_TYPES_H_
+
+#include <functional>
+#include <utility>
+
+#include "net/dcsctp/public/types.h"
+#include "rtc_base/strong_alias.h"
+
+namespace dcsctp {
+
+// Stream Sequence Number (SSN)
+using SSN = webrtc::StrongAlias<class SSNTag, uint16_t>;
+
+// Message Identifier (MID)
+using MID = webrtc::StrongAlias<class MIDTag, uint32_t>;
+
+// Fragment Sequence Number (FSN)
+using FSN = webrtc::StrongAlias<class FSNTag, uint32_t>;
+
+// Transmission Sequence Number (TSN)
+using TSN = webrtc::StrongAlias<class TSNTag, uint32_t>;
+
+// Reconfiguration Request Sequence Number
+using ReconfigRequestSN =
+    webrtc::StrongAlias<class ReconfigRequestSNTag, uint32_t>;
+
+// Verification Tag, used for packet validation.
+using VerificationTag = webrtc::StrongAlias<class VerificationTagTag, uint32_t>;
+
+// Tie Tag, used as a nonce when connecting.
+using TieTag = webrtc::StrongAlias<class TieTagTag, uint64_t>;
+
+}  // namespace dcsctp
+#endif  // NET_DCSCTP_COMMON_INTERNAL_TYPES_H_
diff --git a/third_party/libwebrtc/net/dcsctp/common/math.h b/third_party/libwebrtc/net/dcsctp/common/math.h
new file mode 100644
index 0000000000..12f690ed57
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/common/math.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_COMMON_MATH_H_
+#define NET_DCSCTP_COMMON_MATH_H_
+
+namespace dcsctp {
+
+// Rounds up `val` to the nearest value that is divisible by four. Frequently
+// used to e.g. pad chunks or parameters to an even 32-bit offset.
+template <typename IntType>
+IntType RoundUpTo4(IntType val) {
+  return (val + 3) & ~3;
+}
+
+// Similarly, rounds down `val` to the nearest value that is divisible by four.
+template <typename IntType>
+IntType RoundDownTo4(IntType val) {
+  return val & ~3;
+}
+
+// Returns true if `val` is divisible by four.
+template <typename IntType>
+bool IsDivisibleBy4(IntType val) {
+  return (val & 3) == 0;
+}
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_COMMON_MATH_H_
diff --git a/third_party/libwebrtc/net/dcsctp/common/math_test.cc b/third_party/libwebrtc/net/dcsctp/common/math_test.cc
new file mode 100644
index 0000000000..f95dfbdb55
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/common/math_test.cc
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/common/math.h"
+
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(MathUtilTest, CanRoundUpTo4) {
+  // Signed numbers
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(-5)), -4);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(-4)), -4);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(-3)), 0);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(-2)), 0);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(-1)), 0);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(0)), 0);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(1)), 4);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(2)), 4);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(3)), 4);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(4)), 4);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(5)), 8);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(6)), 8);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(7)), 8);
+  EXPECT_EQ(RoundUpTo4(static_cast<int>(8)), 8);
+  EXPECT_EQ(RoundUpTo4(static_cast<int64_t>(10000000000)), 10000000000);
+  EXPECT_EQ(RoundUpTo4(static_cast<int64_t>(10000000001)), 10000000004);
+
+  // Unsigned numbers
+  EXPECT_EQ(RoundUpTo4(static_cast<unsigned int>(0)), 0u);
+  EXPECT_EQ(RoundUpTo4(static_cast<unsigned int>(1)), 4u);
+  EXPECT_EQ(RoundUpTo4(static_cast<unsigned int>(2)), 4u);
+  EXPECT_EQ(RoundUpTo4(static_cast<unsigned int>(3)), 4u);
+  EXPECT_EQ(RoundUpTo4(static_cast<unsigned int>(4)), 4u);
+  EXPECT_EQ(RoundUpTo4(static_cast<unsigned int>(5)), 8u);
+  EXPECT_EQ(RoundUpTo4(static_cast<unsigned int>(6)), 8u);
+  EXPECT_EQ(RoundUpTo4(static_cast<unsigned int>(7)), 8u);
+  EXPECT_EQ(RoundUpTo4(static_cast<unsigned int>(8)), 8u);
+  EXPECT_EQ(RoundUpTo4(static_cast<uint64_t>(10000000000)), 10000000000u);
+  EXPECT_EQ(RoundUpTo4(static_cast<uint64_t>(10000000001)), 10000000004u);
+}
+
+TEST(MathUtilTest, CanRoundDownTo4) {
+  // Signed numbers
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(-5)), -8);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(-4)), -4);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(-3)), -4);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(-2)), -4);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(-1)), -4);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(0)), 0);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(1)), 0);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(2)), 0);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(3)), 0);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(4)), 4);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(5)), 4);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(6)), 4);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(7)), 4);
+  EXPECT_EQ(RoundDownTo4(static_cast<int>(8)), 8);
+  EXPECT_EQ(RoundDownTo4(static_cast<int64_t>(10000000000)), 10000000000);
+  EXPECT_EQ(RoundDownTo4(static_cast<int64_t>(10000000001)), 10000000000);
+
+  // Unsigned numbers
+  EXPECT_EQ(RoundDownTo4(static_cast<unsigned int>(0)), 0u);
+  EXPECT_EQ(RoundDownTo4(static_cast<unsigned int>(1)), 0u);
+  EXPECT_EQ(RoundDownTo4(static_cast<unsigned int>(2)), 0u);
+  EXPECT_EQ(RoundDownTo4(static_cast<unsigned int>(3)), 0u);
+  EXPECT_EQ(RoundDownTo4(static_cast<unsigned int>(4)), 4u);
+  EXPECT_EQ(RoundDownTo4(static_cast<unsigned int>(5)), 4u);
+  EXPECT_EQ(RoundDownTo4(static_cast<unsigned int>(6)), 4u);
+  EXPECT_EQ(RoundDownTo4(static_cast<unsigned int>(7)), 4u);
+  EXPECT_EQ(RoundDownTo4(static_cast<unsigned int>(8)), 8u);
+  EXPECT_EQ(RoundDownTo4(static_cast<uint64_t>(10000000000)), 10000000000u);
+  EXPECT_EQ(RoundDownTo4(static_cast<uint64_t>(10000000001)), 10000000000u);
+}
+
+TEST(MathUtilTest, IsDivisibleBy4) {
+  // Signed numbers
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(-4)), true);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(-3)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(-2)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(-1)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(0)), true);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(1)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(2)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(3)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(4)), true);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(5)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(6)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(7)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int>(8)), true);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int64_t>(10000000000)), true);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<int64_t>(10000000001)), false);
+
+  // Unsigned numbers
+  EXPECT_EQ(IsDivisibleBy4(static_cast<unsigned int>(0)), true);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<unsigned int>(1)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<unsigned int>(2)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<unsigned int>(3)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<unsigned int>(4)), true);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<unsigned int>(5)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<unsigned int>(6)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<unsigned int>(7)), false);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<unsigned int>(8)), true);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<uint64_t>(10000000000)), true);
+  EXPECT_EQ(IsDivisibleBy4(static_cast<uint64_t>(10000000001)), false);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/common/sequence_numbers.h b/third_party/libwebrtc/net/dcsctp/common/sequence_numbers.h
new file mode 100644
index 0000000000..d324fb223a
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/common/sequence_numbers.h
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_COMMON_SEQUENCE_NUMBERS_H_
+#define NET_DCSCTP_COMMON_SEQUENCE_NUMBERS_H_
+
+#include <cstdint>
+#include <limits>
+#include <utility>
+
+#include "net/dcsctp/common/internal_types.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace dcsctp {
+
+// UnwrappedSequenceNumber handles wrapping sequence numbers and unwraps them to
+// an int64_t value space, to allow wrapped sequence numbers to be easily
+// compared for ordering.
+//
+// Sequence numbers are expected to be monotonically increasing, but they do not
+// need to be unwrapped in order, as long as the difference to the previous one
+// is not larger than half the range of the wrapped sequence number.
+//
+// The WrappedType must be a webrtc::StrongAlias type.
+template <typename WrappedType>
+class UnwrappedSequenceNumber {
+ public:
+  static_assert(
+      !std::numeric_limits<typename WrappedType::UnderlyingType>::is_signed,
+      "The wrapped type must be unsigned");
+  static_assert(
+      std::numeric_limits<typename WrappedType::UnderlyingType>::max() <
+          std::numeric_limits<int64_t>::max(),
+      "The wrapped type must be less than the int64_t value space");
+
+  // The unwrapper is a sort of factory and converts wrapped sequence numbers to
+  // unwrapped ones.
+  class Unwrapper {
+   public:
+    Unwrapper() = default;
+    Unwrapper(const Unwrapper&) = default;
+    Unwrapper& operator=(const Unwrapper&) = default;
+
+    // Given a wrapped `value`, and with knowledge of its current last seen
+    // largest number, will return a value that can be compared using normal
+    // operators, such as less-than, greater-than etc.
+    //
+    // This will also update the Unwrapper's state, to track the last seen
+    // largest value.
+    UnwrappedSequenceNumber<WrappedType> Unwrap(WrappedType value) {
+      return UnwrappedSequenceNumber<WrappedType>(unwrapper_.Unwrap(*value));
+    }
+
+    // Similar to `Unwrap`, but will not update the Unwrappers's internal state.
+    UnwrappedSequenceNumber<WrappedType> PeekUnwrap(WrappedType value) const {
+      return UnwrappedSequenceNumber<WrappedType>(
+          unwrapper_.PeekUnwrap(*value));
+    }
+
+    // Resets the Unwrapper to its pristine state. Used when a sequence number
+    // is to be reset to zero.
+    void Reset() { unwrapper_.Reset(); }
+
+   private:
+    webrtc::SeqNumUnwrapper<typename WrappedType::UnderlyingType> unwrapper_;
+  };
+
+  // Returns the wrapped value this type represents.
+  WrappedType Wrap() const {
+    return static_cast<WrappedType>(value_ % kValueLimit);
+  }
+
+  template <typename H>
+  friend H AbslHashValue(H state,
+                         const UnwrappedSequenceNumber<WrappedType>& hash) {
+    return H::combine(std::move(state), hash.value_);
+  }
+
+  bool operator==(const UnwrappedSequenceNumber<WrappedType>& other) const {
+    return value_ == other.value_;
+  }
+  bool operator!=(const UnwrappedSequenceNumber<WrappedType>& other) const {
+    return value_ != other.value_;
+  }
+  bool operator<(const UnwrappedSequenceNumber<WrappedType>& other) const {
+    return value_ < other.value_;
+  }
+  bool operator>(const UnwrappedSequenceNumber<WrappedType>& other) const {
+    return value_ > other.value_;
+  }
+  bool operator>=(const UnwrappedSequenceNumber<WrappedType>& other) const {
+    return value_ >= other.value_;
+  }
+  bool operator<=(const UnwrappedSequenceNumber<WrappedType>& other) const {
+    return value_ <= other.value_;
+  }
+
+  // Const accessors for underlying value.
+  constexpr const int64_t* operator->() const { return &value_; }
+  constexpr const int64_t& operator*() const& { return value_; }
+  constexpr const int64_t&& operator*() const&& { return std::move(value_); }
+  constexpr const int64_t& value() const& { return value_; }
+  constexpr const int64_t&& value() const&& { return std::move(value_); }
+  constexpr explicit operator const int64_t&() const& { return value_; }
+
+  // Increments the value.
+  void Increment() { ++value_; }
+
+  // Returns the next value relative to this sequence number.
+  UnwrappedSequenceNumber<WrappedType> next_value() const {
+    return UnwrappedSequenceNumber<WrappedType>(value_ + 1);
+  }
+
+  // Returns a new sequence number based on `value`, and adding `delta` (which
+  // may be negative).
+  static UnwrappedSequenceNumber<WrappedType> AddTo(
+      UnwrappedSequenceNumber<WrappedType> value,
+      int delta) {
+    return UnwrappedSequenceNumber<WrappedType>(value.value_ + delta);
+  }
+
+  // Returns the absolute difference between `lhs` and `rhs`.
+  static typename WrappedType::UnderlyingType Difference(
+      UnwrappedSequenceNumber<WrappedType> lhs,
+      UnwrappedSequenceNumber<WrappedType> rhs) {
+    return (lhs.value_ > rhs.value_) ? (lhs.value_ - rhs.value_)
+                                     : (rhs.value_ - lhs.value_);
+  }
+
+ private:
+  explicit UnwrappedSequenceNumber(int64_t value) : value_(value) {}
+  static constexpr int64_t kValueLimit =
+      static_cast<int64_t>(1)
+      << std::numeric_limits<typename WrappedType::UnderlyingType>::digits;
+
+  int64_t value_;
+};
+
+// Unwrapped Transmission Sequence Numbers (TSN)
+using UnwrappedTSN = UnwrappedSequenceNumber<TSN>;
+
+// Unwrapped Stream Sequence Numbers (SSN)
+using UnwrappedSSN = UnwrappedSequenceNumber<SSN>;
+
+// Unwrapped Message Identifier (MID)
+using UnwrappedMID = UnwrappedSequenceNumber<MID>;
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_COMMON_SEQUENCE_NUMBERS_H_
diff --git a/third_party/libwebrtc/net/dcsctp/common/sequence_numbers_test.cc b/third_party/libwebrtc/net/dcsctp/common/sequence_numbers_test.cc
new file mode 100644
index 0000000000..c4842f089e
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/common/sequence_numbers_test.cc
@@ -0,0 +1,202 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/common/sequence_numbers.h"
+
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+using Wrapped = webrtc::StrongAlias<class WrappedTag, uint16_t>;
+using TestSequence = UnwrappedSequenceNumber<Wrapped>;
+
+TEST(SequenceNumbersTest, SimpleUnwrapping) {
+  TestSequence::Unwrapper unwrapper;
+
+  TestSequence s0 = unwrapper.Unwrap(Wrapped(0));
+  TestSequence s1 = unwrapper.Unwrap(Wrapped(1));
+  TestSequence s2 = unwrapper.Unwrap(Wrapped(2));
+  TestSequence s3 = unwrapper.Unwrap(Wrapped(3));
+
+  EXPECT_LT(s0, s1);
+  EXPECT_LT(s0, s2);
+  EXPECT_LT(s0, s3);
+  EXPECT_LT(s1, s2);
+  EXPECT_LT(s1, s3);
+  EXPECT_LT(s2, s3);
+
+  EXPECT_EQ(TestSequence::Difference(s1, s0), 1);
+  EXPECT_EQ(TestSequence::Difference(s2, s0), 2);
+  EXPECT_EQ(TestSequence::Difference(s3, s0), 3);
+
+  EXPECT_GT(s1, s0);
+  EXPECT_GT(s2, s0);
+  EXPECT_GT(s3, s0);
+  EXPECT_GT(s2, s1);
+  EXPECT_GT(s3, s1);
+  EXPECT_GT(s3, s2);
+
+  s0.Increment();
+  EXPECT_EQ(s0, s1);
+  s1.Increment();
+  EXPECT_EQ(s1, s2);
+  s2.Increment();
+  EXPECT_EQ(s2, s3);
+
+  EXPECT_EQ(TestSequence::AddTo(s0, 2), s3);
+}
+
+TEST(SequenceNumbersTest, MidValueUnwrapping) {
+  TestSequence::Unwrapper unwrapper;
+
+  TestSequence s0 = unwrapper.Unwrap(Wrapped(0x7FFE));
+  TestSequence s1 = unwrapper.Unwrap(Wrapped(0x7FFF));
+  TestSequence s2 = unwrapper.Unwrap(Wrapped(0x8000));
+  TestSequence s3 = unwrapper.Unwrap(Wrapped(0x8001));
+
+  EXPECT_LT(s0, s1);
+  EXPECT_LT(s0, s2);
+  EXPECT_LT(s0, s3);
+  EXPECT_LT(s1, s2);
+  EXPECT_LT(s1, s3);
+  EXPECT_LT(s2, s3);
+
+  EXPECT_EQ(TestSequence::Difference(s1, s0), 1);
+  EXPECT_EQ(TestSequence::Difference(s2, s0), 2);
+  EXPECT_EQ(TestSequence::Difference(s3, s0), 3);
+
+  EXPECT_GT(s1, s0);
+  EXPECT_GT(s2, s0);
+  EXPECT_GT(s3, s0);
+  EXPECT_GT(s2, s1);
+  EXPECT_GT(s3, s1);
+  EXPECT_GT(s3, s2);
+
+  s0.Increment();
+  EXPECT_EQ(s0, s1);
+  s1.Increment();
+  EXPECT_EQ(s1, s2);
+  s2.Increment();
+  EXPECT_EQ(s2, s3);
+
+  EXPECT_EQ(TestSequence::AddTo(s0, 2), s3);
+}
+
+TEST(SequenceNumbersTest, WrappedUnwrapping) {
+  TestSequence::Unwrapper unwrapper;
+
+  TestSequence s0 = unwrapper.Unwrap(Wrapped(0xFFFE));
+  TestSequence s1 = unwrapper.Unwrap(Wrapped(0xFFFF));
+  TestSequence s2 = unwrapper.Unwrap(Wrapped(0x0000));
+  TestSequence s3 = unwrapper.Unwrap(Wrapped(0x0001));
+
+  EXPECT_LT(s0, s1);
+  EXPECT_LT(s0, s2);
+  EXPECT_LT(s0, s3);
+  EXPECT_LT(s1, s2);
+  EXPECT_LT(s1, s3);
+  EXPECT_LT(s2, s3);
+
+  EXPECT_EQ(TestSequence::Difference(s1, s0), 1);
+  EXPECT_EQ(TestSequence::Difference(s2, s0), 2);
+  EXPECT_EQ(TestSequence::Difference(s3, s0), 3);
+
+  EXPECT_GT(s1, s0);
+  EXPECT_GT(s2, s0);
+  EXPECT_GT(s3, s0);
+  EXPECT_GT(s2, s1);
+  EXPECT_GT(s3, s1);
+  EXPECT_GT(s3, s2);
+
+  s0.Increment();
+  EXPECT_EQ(s0, s1);
+  s1.Increment();
+  EXPECT_EQ(s1, s2);
+  s2.Increment();
+  EXPECT_EQ(s2, s3);
+
+  EXPECT_EQ(TestSequence::AddTo(s0, 2), s3);
+}
+
+TEST(SequenceNumbersTest, WrapAroundAFewTimes) {
+  TestSequence::Unwrapper unwrapper;
+
+  TestSequence s0 = unwrapper.Unwrap(Wrapped(0));
+  TestSequence prev = s0;
+
+  for (uint32_t i = 1; i < 65536 * 3; i++) {
+    uint16_t wrapped = static_cast<uint16_t>(i);
+    TestSequence si = unwrapper.Unwrap(Wrapped(wrapped));
+
+    EXPECT_LT(s0, si);
+    EXPECT_LT(prev, si);
+    prev = si;
+  }
+}
+
+TEST(SequenceNumbersTest, IncrementIsSameAsWrapped) {
+  TestSequence::Unwrapper unwrapper;
+
+  TestSequence s0 = unwrapper.Unwrap(Wrapped(0));
+
+  for (uint32_t i = 1; i < 65536 * 2; i++) {
+    uint16_t wrapped = static_cast<uint16_t>(i);
+    TestSequence si = unwrapper.Unwrap(Wrapped(wrapped));
+
+    s0.Increment();
+    EXPECT_EQ(s0, si);
+  }
+}
+
+TEST(SequenceNumbersTest, UnwrappingLargerNumberIsAlwaysLarger) {
+  TestSequence::Unwrapper unwrapper;
+
+  for (uint32_t i = 1; i < 65536 * 2; i++) {
+    uint16_t wrapped = static_cast<uint16_t>(i);
+    TestSequence si = unwrapper.Unwrap(Wrapped(wrapped));
+
+    EXPECT_GT(unwrapper.Unwrap(Wrapped(wrapped + 1)), si);
+    EXPECT_GT(unwrapper.Unwrap(Wrapped(wrapped + 5)), si);
+    EXPECT_GT(unwrapper.Unwrap(Wrapped(wrapped + 10)), si);
+    EXPECT_GT(unwrapper.Unwrap(Wrapped(wrapped + 100)), si);
+  }
+}
+
+TEST(SequenceNumbersTest, UnwrappingSmallerNumberIsAlwaysSmaller) {
+  TestSequence::Unwrapper unwrapper;
+
+  for (uint32_t i = 1; i < 65536 * 2; i++) {
+    uint16_t wrapped = static_cast<uint16_t>(i);
+    TestSequence si = unwrapper.Unwrap(Wrapped(wrapped));
+
+    EXPECT_LT(unwrapper.Unwrap(Wrapped(wrapped - 1)), si);
+    EXPECT_LT(unwrapper.Unwrap(Wrapped(wrapped - 5)), si);
+    EXPECT_LT(unwrapper.Unwrap(Wrapped(wrapped - 10)), si);
+    EXPECT_LT(unwrapper.Unwrap(Wrapped(wrapped - 100)), si);
+  }
+}
+
+TEST(SequenceNumbersTest, DifferenceIsAbsolute) {
+  TestSequence::Unwrapper unwrapper;
+
+  TestSequence this_value = unwrapper.Unwrap(Wrapped(10));
+  TestSequence other_value = TestSequence::AddTo(this_value, 100);
+
+  EXPECT_EQ(TestSequence::Difference(this_value, other_value), 100);
+  EXPECT_EQ(TestSequence::Difference(other_value, this_value), 100);
+
+  TestSequence minus_value = TestSequence::AddTo(this_value, -100);
+
+  EXPECT_EQ(TestSequence::Difference(this_value, minus_value), 100);
+  EXPECT_EQ(TestSequence::Difference(minus_value, this_value), 100);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/common/str_join.h b/third_party/libwebrtc/net/dcsctp/common/str_join.h
new file mode 100644
index 0000000000..04517827b7
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/common/str_join.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_COMMON_STR_JOIN_H_
+#define NET_DCSCTP_COMMON_STR_JOIN_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+template <typename Range>
+std::string StrJoin(const Range& seq, absl::string_view delimiter) {
+  rtc::StringBuilder sb;
+  int idx = 0;
+
+  for (const typename Range::value_type& elem : seq) {
+    if (idx > 0) {
+      sb << delimiter;
+    }
+    sb << elem;
+
+    ++idx;
+  }
+  return sb.Release();
+}
+
+template <typename Range, typename Functor>
+std::string StrJoin(const Range& seq,
+                    absl::string_view delimiter,
+                    const Functor& fn) {
+  rtc::StringBuilder sb;
+  int idx = 0;
+
+  for (const typename Range::value_type& elem : seq) {
+    if (idx > 0) {
+      sb << delimiter;
+    }
+    fn(sb, elem);
+
+    ++idx;
+  }
+  return sb.Release();
+}
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_COMMON_STR_JOIN_H_
diff --git a/third_party/libwebrtc/net/dcsctp/common/str_join_test.cc b/third_party/libwebrtc/net/dcsctp/common/str_join_test.cc
new file mode 100644
index 0000000000..dbfd92c1cf
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/common/str_join_test.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/common/str_join.h"
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(StrJoinTest, CanJoinStringsFromVector) {
+  std::vector<std::string> strings = {"Hello", "World"};
+  std::string s = StrJoin(strings, " ");
+  EXPECT_EQ(s, "Hello World");
+}
+
+TEST(StrJoinTest, CanJoinNumbersFromArray) {
+  std::array<int, 3> numbers = {1, 2, 3};
+  std::string s = StrJoin(numbers, ",");
+  EXPECT_EQ(s, "1,2,3");
+}
+
+TEST(StrJoinTest, CanFormatElementsWhileJoining) {
+  std::vector<std::pair<std::string, std::string>> pairs = {
+      {"hello", "world"}, {"foo", "bar"}, {"fum", "gazonk"}};
+  std::string s = StrJoin(pairs, ",",
+                          [&](rtc::StringBuilder& sb,
+                              const std::pair<std::string, std::string>& p) {
+                            sb << p.first << "=" << p.second;
+                          });
+  EXPECT_EQ(s, "hello=world,foo=bar,fum=gazonk");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/fuzzers/BUILD.gn b/third_party/libwebrtc/net/dcsctp/fuzzers/BUILD.gn
new file mode 100644
index 0000000000..302c828684
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/fuzzers/BUILD.gn
@@ -0,0 +1,50 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("dcsctp_fuzzers") {
+  testonly = true
+  deps = [
+    "../../../api:array_view",
+    "../../../api/task_queue:task_queue",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../common:math",
+    "../packet:chunk",
+    "../packet:error_cause",
+    "../packet:parameter",
+    "../public:socket",
+    "../public:types",
+    "../socket:dcsctp_socket",
+  ]
+  sources = [
+    "dcsctp_fuzzers.cc",
+    "dcsctp_fuzzers.h",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("dcsctp_fuzzers_unittests") {
+    testonly = true
+
+    deps = [
+      ":dcsctp_fuzzers",
+      "../../../api:array_view",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:gunit_helpers",
+      "../../../rtc_base:logging",
+      "../../../test:test_support",
+      "../packet:sctp_packet",
+      "../public:socket",
+      "../socket:dcsctp_socket",
+      "../testing:testing_macros",
+    ]
+    sources = [ "dcsctp_fuzzers_test.cc" ]
+  }
+}
diff --git a/third_party/libwebrtc/net/dcsctp/fuzzers/dcsctp_fuzzers.cc b/third_party/libwebrtc/net/dcsctp/fuzzers/dcsctp_fuzzers.cc
new file mode 100644
index 0000000000..e8fcacffa0
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/fuzzers/dcsctp_fuzzers.cc
@@ -0,0 +1,461 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/fuzzers/dcsctp_fuzzers.h"
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "net/dcsctp/common/math.h"
+#include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/chunk/shutdown_chunk.h"
+#include "net/dcsctp/packet/error_cause/protocol_violation_cause.h"
+#include "net/dcsctp/packet/error_cause/user_initiated_abort_cause.h"
+#include "net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/state_cookie_parameter.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/socket/dcsctp_socket.h"
+#include "net/dcsctp/socket/state_cookie.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+namespace dcsctp_fuzzers {
+namespace {
+static constexpr int kRandomValue = FuzzerCallbacks::kRandomValue;
+static constexpr size_t kMinInputLength = 5;
+static constexpr size_t kMaxInputLength = 1024;
+
+// A starting state for the socket, when fuzzing.
+enum class StartingState : int {
+  kConnectNotCalled,
+  // When socket initiating Connect
+  kConnectCalled,
+  kReceivedInitAck,
+  kReceivedCookieAck,
+  // When socket initiating Shutdown
+  kShutdownCalled,
+  kReceivedShutdownAck,
+  // When peer socket initiated Connect
+  kReceivedInit,
+  kReceivedCookieEcho,
+  // When peer initiated Shutdown
+  kReceivedShutdown,
+  kReceivedShutdownComplete,
+  kNumberOfStates,
+};
+
+// State about the current fuzzing iteration
+class FuzzState {
+ public:
+  explicit FuzzState(rtc::ArrayView<const uint8_t> data) : data_(data) {}
+
+  uint8_t GetByte() {
+    uint8_t value = 0;
+    if (offset_ < data_.size()) {
+      value = data_[offset_];
+      ++offset_;
+    }
+    return value;
+  }
+
+  TSN GetNextTSN() { return TSN(tsn_++); }
+  MID GetNextMID() { return MID(mid_++); }
+
+  bool empty() const { return offset_ >= data_.size(); }
+
+ private:
+  uint32_t tsn_ = kRandomValue;
+  uint32_t mid_ = 0;
+  rtc::ArrayView<const uint8_t> data_;
+  size_t offset_ = 0;
+};
+
+void SetSocketState(DcSctpSocketInterface& socket,
+                    FuzzerCallbacks& socket_cb,
+                    StartingState state) {
+  // We'll use another temporary peer socket for the establishment.
+  FuzzerCallbacks peer_cb;
+  DcSctpSocket peer("peer", peer_cb, nullptr, {});
+
+  switch (state) {
+    case StartingState::kConnectNotCalled:
+      return;
+    case StartingState::kConnectCalled:
+      socket.Connect();
+      return;
+    case StartingState::kReceivedInitAck:
+      socket.Connect();
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // INIT
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // INIT_ACK
+      return;
+    case StartingState::kReceivedCookieAck:
+      socket.Connect();
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // INIT
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // INIT_ACK
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // COOKIE_ECHO
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // COOKIE_ACK
+      return;
+    case StartingState::kShutdownCalled:
+      socket.Connect();
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // INIT
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // INIT_ACK
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // COOKIE_ECHO
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // COOKIE_ACK
+      socket.Shutdown();
+      return;
+    case StartingState::kReceivedShutdownAck:
+      socket.Connect();
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // INIT
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // INIT_ACK
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // COOKIE_ECHO
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // COOKIE_ACK
+      socket.Shutdown();
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // SHUTDOWN
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // SHUTDOWN_ACK
+      return;
+    case StartingState::kReceivedInit:
+      peer.Connect();
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // INIT
+      return;
+    case StartingState::kReceivedCookieEcho:
+      peer.Connect();
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // INIT
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // INIT_ACK
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // COOKIE_ECHO
+      return;
+    case StartingState::kReceivedShutdown:
+      socket.Connect();
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // INIT
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // INIT_ACK
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // COOKIE_ECHO
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // COOKIE_ACK
+      peer.Shutdown();
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // SHUTDOWN
+      return;
+    case StartingState::kReceivedShutdownComplete:
+      socket.Connect();
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // INIT
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // INIT_ACK
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // COOKIE_ECHO
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // COOKIE_ACK
+      peer.Shutdown();
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // SHUTDOWN
+      peer.ReceivePacket(socket_cb.ConsumeSentPacket());  // SHUTDOWN_ACK
+      socket.ReceivePacket(peer_cb.ConsumeSentPacket());  // SHUTDOWN_COMPLETE
+      return;
+    case StartingState::kNumberOfStates:
+      RTC_CHECK(false);
+      return;
+  }
+}
+
+void MakeDataChunk(FuzzState& state, SctpPacket::Builder& b) {
+  DataChunk::Options options;
+  options.is_unordered = IsUnordered(state.GetByte() != 0);
+  options.is_beginning = Data::IsBeginning(state.GetByte() != 0);
+  options.is_end = Data::IsEnd(state.GetByte() != 0);
+  b.Add(DataChunk(state.GetNextTSN(), StreamID(state.GetByte()),
+                  SSN(state.GetByte()), PPID(53), std::vector<uint8_t>(10),
+                  options));
+}
+
+void MakeInitChunk(FuzzState& state, SctpPacket::Builder& b) {
+  Parameters::Builder builder;
+  builder.Add(ForwardTsnSupportedParameter());
+
+  b.Add(InitChunk(VerificationTag(kRandomValue), 10000, 1000, 1000,
+                  TSN(kRandomValue), builder.Build()));
+}
+
+void MakeInitAckChunk(FuzzState& state, SctpPacket::Builder& b) {
+  Parameters::Builder builder;
+  builder.Add(ForwardTsnSupportedParameter());
+
+  uint8_t state_cookie[] = {1, 2, 3, 4, 5};
+  Parameters::Builder params_builder =
+      Parameters::Builder().Add(StateCookieParameter(state_cookie));
+
+  b.Add(InitAckChunk(VerificationTag(kRandomValue), 10000, 1000, 1000,
+                     TSN(kRandomValue), builder.Build()));
+}
+
+void MakeSackChunk(FuzzState& state, SctpPacket::Builder& b) {
+  std::vector<SackChunk::GapAckBlock> gap_ack_blocks;
+  uint16_t last_end = 0;
+  while (gap_ack_blocks.size() < 20) {
+    uint8_t delta_start = state.GetByte();
+    if (delta_start < 0x80) {
+      break;
+    }
+    uint8_t delta_end = state.GetByte();
+
+    uint16_t start = last_end + delta_start;
+    uint16_t end = start + delta_end;
+    last_end = end;
+    gap_ack_blocks.emplace_back(start, end);
+  }
+
+  TSN cum_ack_tsn(kRandomValue + state.GetByte());
+  b.Add(SackChunk(cum_ack_tsn, 10000, std::move(gap_ack_blocks), {}));
+}
+
+void MakeHeartbeatRequestChunk(FuzzState& state, SctpPacket::Builder& b) {
+  uint8_t info[] = {1, 2, 3, 4, 5};
+  b.Add(HeartbeatRequestChunk(
+      Parameters::Builder().Add(HeartbeatInfoParameter(info)).Build()));
+}
+
+void MakeHeartbeatAckChunk(FuzzState& state, SctpPacket::Builder& b) {
+  std::vector<uint8_t> info(8);
+  b.Add(HeartbeatRequestChunk(
+      Parameters::Builder().Add(HeartbeatInfoParameter(info)).Build()));
+}
+
+void MakeAbortChunk(FuzzState& state, SctpPacket::Builder& b) {
+  b.Add(AbortChunk(
+      /*filled_in_verification_tag=*/true,
+      Parameters::Builder().Add(UserInitiatedAbortCause("Fuzzing")).Build()));
+}
+
+void MakeErrorChunk(FuzzState& state, SctpPacket::Builder& b) {
+  b.Add(ErrorChunk(
+      Parameters::Builder().Add(ProtocolViolationCause("Fuzzing")).Build()));
+}
+
+void MakeCookieEchoChunk(FuzzState& state, SctpPacket::Builder& b) {
+  std::vector<uint8_t> cookie(StateCookie::kCookieSize);
+  b.Add(CookieEchoChunk(cookie));
+}
+
+void MakeCookieAckChunk(FuzzState& state, SctpPacket::Builder& b) {
+  b.Add(CookieAckChunk());
+}
+
+void MakeShutdownChunk(FuzzState& state, SctpPacket::Builder& b) {
+  b.Add(ShutdownChunk(state.GetNextTSN()));
+}
+
+void MakeShutdownAckChunk(FuzzState& state, SctpPacket::Builder& b) {
+  b.Add(ShutdownAckChunk());
+}
+
+void MakeShutdownCompleteChunk(FuzzState& state, SctpPacket::Builder& b) {
+  b.Add(ShutdownCompleteChunk(false));
+}
+
+void MakeReConfigChunk(FuzzState& state, SctpPacket::Builder& b) {
+  std::vector<StreamID> streams = {StreamID(state.GetByte())};
+  Parameters::Builder params_builder =
+      Parameters::Builder().Add(OutgoingSSNResetRequestParameter(
+          ReconfigRequestSN(kRandomValue), ReconfigRequestSN(kRandomValue),
+          state.GetNextTSN(), streams));
+  b.Add(ReConfigChunk(params_builder.Build()));
+}
+
+void MakeForwardTsnChunk(FuzzState& state, SctpPacket::Builder& b) {
+  std::vector<ForwardTsnChunk::SkippedStream> skipped_streams;
+  for (;;) {
+    uint8_t stream = state.GetByte();
+    if (skipped_streams.size() > 20 || stream < 0x80) {
+      break;
+    }
+    skipped_streams.emplace_back(StreamID(stream), SSN(state.GetByte()));
+  }
+  b.Add(ForwardTsnChunk(state.GetNextTSN(), std::move(skipped_streams)));
+}
+
+void MakeIDataChunk(FuzzState& state, SctpPacket::Builder& b) {
+  DataChunk::Options options;
+  options.is_unordered = IsUnordered(state.GetByte() != 0);
+  options.is_beginning = Data::IsBeginning(state.GetByte() != 0);
+  options.is_end = Data::IsEnd(state.GetByte() != 0);
+  b.Add(IDataChunk(state.GetNextTSN(), StreamID(state.GetByte()),
+                   state.GetNextMID(), PPID(53), FSN(0),
+                   std::vector<uint8_t>(10), options));
+}
+
+void MakeIForwardTsnChunk(FuzzState& state, SctpPacket::Builder& b) {
+  std::vector<ForwardTsnChunk::SkippedStream> skipped_streams;
+  for (;;) {
+    uint8_t stream = state.GetByte();
+    if (skipped_streams.size() > 20 || stream < 0x80) {
+      break;
+    }
+    skipped_streams.emplace_back(StreamID(stream), SSN(state.GetByte()));
+  }
+  b.Add(IForwardTsnChunk(state.GetNextTSN(), std::move(skipped_streams)));
+}
+
+class RandomFuzzedChunk : public Chunk {
+ public:
+  explicit RandomFuzzedChunk(FuzzState& state) : state_(state) {}
+
+  void SerializeTo(std::vector<uint8_t>& out) const override {
+    size_t bytes = state_.GetByte();
+    for (size_t i = 0; i < bytes; ++i) {
+      out.push_back(state_.GetByte());
+    }
+  }
+
+  std::string ToString() const override { return std::string("RANDOM_FUZZED"); }
+
+ private:
+  FuzzState& state_;
+};
+
+void MakeChunkWithRandomContent(FuzzState& state, SctpPacket::Builder& b) {
+  b.Add(RandomFuzzedChunk(state));
+}
+
+std::vector<uint8_t> GeneratePacket(FuzzState& state) {
+  DcSctpOptions options;
+  // Setting a fixed limit to not be dependent on the defaults, which may
+  // change.
+  options.mtu = 2048;
+  SctpPacket::Builder builder(VerificationTag(kRandomValue), options);
+
+  // The largest expected serialized chunk, as created by fuzzers.
+  static constexpr size_t kMaxChunkSize = 256;
+
+  for (int i = 0; i < 5 && builder.bytes_remaining() > kMaxChunkSize; ++i) {
+    switch (state.GetByte()) {
+      case 1:
+        MakeDataChunk(state, builder);
+        break;
+      case 2:
+        MakeInitChunk(state, builder);
+        break;
+      case 3:
+        MakeInitAckChunk(state, builder);
+        break;
+      case 4:
+        MakeSackChunk(state, builder);
+        break;
+      case 5:
+        MakeHeartbeatRequestChunk(state, builder);
+        break;
+      case 6:
+        MakeHeartbeatAckChunk(state, builder);
+        break;
+      case 7:
+        MakeAbortChunk(state, builder);
+        break;
+      case 8:
+        MakeErrorChunk(state, builder);
+        break;
+      case 9:
+        MakeCookieEchoChunk(state, builder);
+        break;
+      case 10:
+        MakeCookieAckChunk(state, builder);
+        break;
+      case 11:
+        MakeShutdownChunk(state, builder);
+        break;
+      case 12:
+        MakeShutdownAckChunk(state, builder);
+        break;
+      case 13:
+        MakeShutdownCompleteChunk(state, builder);
+        break;
+      case 14:
+        MakeReConfigChunk(state, builder);
+        break;
+      case 15:
+        MakeForwardTsnChunk(state, builder);
+        break;
+      case 16:
+        MakeIDataChunk(state, builder);
+        break;
+      case 17:
+        MakeIForwardTsnChunk(state, builder);
+        break;
+      case 18:
+        MakeChunkWithRandomContent(state, builder);
+        break;
+      default:
+        break;
+    }
+  }
+  std::vector<uint8_t> packet = builder.Build();
+  return packet;
+}
+}  // namespace
+
+void FuzzSocket(DcSctpSocketInterface& socket,
+                FuzzerCallbacks& cb,
+                rtc::ArrayView<const uint8_t> data) {
+  if (data.size() < kMinInputLength || data.size() > kMaxInputLength) {
+    return;
+  }
+  if (data[0] >= static_cast<int>(StartingState::kNumberOfStates)) {
+    return;
+  }
+
+  // Set the socket in a specified valid starting state
+  SetSocketState(socket, cb, static_cast<StartingState>(data[0]));
+
+  FuzzState state(data.subview(1));
+
+  while (!state.empty()) {
+    switch (state.GetByte()) {
+      case 1:
+        // Generate a valid SCTP packet (based on fuzz data) and "receive it".
+        socket.ReceivePacket(GeneratePacket(state));
+        break;
+      case 2:
+        socket.Connect();
+        break;
+      case 3:
+        socket.Shutdown();
+        break;
+      case 4:
+        socket.Close();
+        break;
+      case 5: {
+        StreamID streams[] = {StreamID(state.GetByte())};
+        socket.ResetStreams(streams);
+      } break;
+      case 6: {
+        uint8_t flags = state.GetByte();
+        SendOptions options;
+        options.unordered = IsUnordered(flags & 0x01);
+        options.max_retransmissions =
+            (flags & 0x02) != 0 ? absl::make_optional(0) : absl::nullopt;
+        options.lifecycle_id = LifecycleId(42);
+        size_t payload_exponent = (flags >> 2) % 16;
+        size_t payload_size = static_cast<size_t>(1) << payload_exponent;
+        socket.Send(DcSctpMessage(StreamID(state.GetByte()), PPID(53),
+                                  std::vector<uint8_t>(payload_size)),
+                    options);
+        break;
+      }
+      case 7: {
+        // Expire an active timeout/timer.
+        uint8_t timeout_idx = state.GetByte();
+        absl::optional<TimeoutID> timeout_id = cb.ExpireTimeout(timeout_idx);
+        if (timeout_id.has_value()) {
+          socket.HandleTimeout(*timeout_id);
+        }
+        break;
+      }
+      default:
+        break;
+    }
+  }
+}
+}  // namespace dcsctp_fuzzers
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/fuzzers/dcsctp_fuzzers.h b/third_party/libwebrtc/net/dcsctp/fuzzers/dcsctp_fuzzers.h
new file mode 100644
index 0000000000..90cfa35099
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/fuzzers/dcsctp_fuzzers.h
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_FUZZERS_DCSCTP_FUZZERS_H_
+#define NET_DCSCTP_FUZZERS_DCSCTP_FUZZERS_H_
+
+#include <deque>
+#include <memory>
+#include <set>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+
+namespace dcsctp {
+namespace dcsctp_fuzzers {
+
+// A fake timeout used during fuzzing.
+class FuzzerTimeout : public Timeout {
+ public:
+  explicit FuzzerTimeout(std::set<TimeoutID>& active_timeouts)
+      : active_timeouts_(active_timeouts) {}
+
+  void Start(DurationMs duration_ms, TimeoutID timeout_id) override {
+    // Start is only allowed to be called on stopped or expired timeouts.
+    if (timeout_id_.has_value()) {
+      // It has been started before, but maybe it expired. Ensure that it's not
+      // running at least.
+      RTC_DCHECK(active_timeouts_.find(*timeout_id_) == active_timeouts_.end());
+    }
+    timeout_id_ = timeout_id;
+    RTC_DCHECK(active_timeouts_.insert(timeout_id).second);
+  }
+
+  void Stop() override {
+    // Stop is only allowed to be called on active timeouts. Not stopped or
+    // expired.
+    RTC_DCHECK(timeout_id_.has_value());
+    RTC_DCHECK(active_timeouts_.erase(*timeout_id_) == 1);
+    timeout_id_ = absl::nullopt;
+  }
+
+  // A set of all active timeouts, managed by `FuzzerCallbacks`.
+  std::set<TimeoutID>& active_timeouts_;
+  // If present, the timout is active and will expire reported as `timeout_id`.
+  absl::optional<TimeoutID> timeout_id_;
+};
+
+class FuzzerCallbacks : public DcSctpSocketCallbacks {
+ public:
+  static constexpr int kRandomValue = 42;
+  void SendPacket(rtc::ArrayView<const uint8_t> data) override {
+    sent_packets_.emplace_back(std::vector<uint8_t>(data.begin(), data.end()));
+  }
+  std::unique_ptr<Timeout> CreateTimeout(
+      webrtc::TaskQueueBase::DelayPrecision precision) override {
+    // The fuzzer timeouts don't implement |precision|.
+    return std::make_unique<FuzzerTimeout>(active_timeouts_);
+  }
+  TimeMs TimeMillis() override { return TimeMs(42); }
+  uint32_t GetRandomInt(uint32_t low, uint32_t high) override {
+    return kRandomValue;
+  }
+  void OnMessageReceived(DcSctpMessage message) override {}
+  void OnError(ErrorKind error, absl::string_view message) override {}
+  void OnAborted(ErrorKind error, absl::string_view message) override {}
+  void OnConnected() override {}
+  void OnClosed() override {}
+  void OnConnectionRestarted() override {}
+  void OnStreamsResetFailed(rtc::ArrayView<const StreamID> outgoing_streams,
+                            absl::string_view reason) override {}
+  void OnStreamsResetPerformed(
+      rtc::ArrayView<const StreamID> outgoing_streams) override {}
+  void OnIncomingStreamsReset(
+      rtc::ArrayView<const StreamID> incoming_streams) override {}
+
+  std::vector<uint8_t> ConsumeSentPacket() {
+    if (sent_packets_.empty()) {
+      return {};
+    }
+    std::vector<uint8_t> ret = sent_packets_.front();
+    sent_packets_.pop_front();
+    return ret;
+  }
+
+  // Given an index among the active timeouts, will expire that one.
+  absl::optional<TimeoutID> ExpireTimeout(size_t index) {
+    if (index < active_timeouts_.size()) {
+      auto it = active_timeouts_.begin();
+      std::advance(it, index);
+      TimeoutID timeout_id = *it;
+      active_timeouts_.erase(it);
+      return timeout_id;
+    }
+    return absl::nullopt;
+  }
+
+ private:
+  // Needs to be ordered, to allow fuzzers to expire timers.
+  std::set<TimeoutID> active_timeouts_;
+  std::deque<std::vector<uint8_t>> sent_packets_;
+};
+
+// Given some fuzzing `data` will send packets to the socket as well as calling
+// API methods.
+void FuzzSocket(DcSctpSocketInterface& socket,
+                FuzzerCallbacks& cb,
+                rtc::ArrayView<const uint8_t> data);
+
+}  // namespace dcsctp_fuzzers
+}  // namespace dcsctp
+#endif  // NET_DCSCTP_FUZZERS_DCSCTP_FUZZERS_H_
diff --git a/third_party/libwebrtc/net/dcsctp/fuzzers/dcsctp_fuzzers_test.cc b/third_party/libwebrtc/net/dcsctp/fuzzers/dcsctp_fuzzers_test.cc
new file mode 100644
index 0000000000..c7d2cd7c99
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/fuzzers/dcsctp_fuzzers_test.cc
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/fuzzers/dcsctp_fuzzers.h"
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/socket/dcsctp_socket.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/logging.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace dcsctp_fuzzers {
+namespace {
+
+// This is a testbed where fuzzed data that cause issues can be evaluated and
+// crashes reproduced. Use `xxd -i ./crash-abc` to generate `data` below.
+TEST(DcsctpFuzzersTest, PassesTestbed) {
+  uint8_t data[] = {0x07, 0x09, 0x00, 0x01, 0x11, 0xff, 0xff};
+
+  FuzzerCallbacks cb;
+  DcSctpOptions options;
+  options.disable_checksum_verification = true;
+  DcSctpSocket socket("A", cb, nullptr, options);
+
+  FuzzSocket(socket, cb, data);
+}
+
+}  // namespace
+}  // namespace dcsctp_fuzzers
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/BUILD.gn b/third_party/libwebrtc/net/dcsctp/packet/BUILD.gn
new file mode 100644
index 0000000000..08bdb0f5a5
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/BUILD.gn
@@ -0,0 +1,331 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+group("packet") {
+  deps = [ ":bounded_io" ]
+}
+
+rtc_source_set("bounded_io") {
+  deps = [
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+  ]
+  sources = [
+    "bounded_byte_reader.h",
+    "bounded_byte_writer.h",
+  ]
+}
+
+rtc_library("tlv_trait") {
+  deps = [
+    ":bounded_io",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings:strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+  sources = [
+    "tlv_trait.cc",
+    "tlv_trait.h",
+  ]
+}
+
+rtc_source_set("data") {
+  deps = [
+    "../../../rtc_base:checks",
+    "../common:internal_types",
+    "../public:types",
+  ]
+  sources = [ "data.h" ]
+}
+
+rtc_library("crc32c") {
+  deps = [
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "//third_party/crc32c",
+  ]
+  sources = [
+    "crc32c.cc",
+    "crc32c.h",
+  ]
+}
+
+rtc_library("parameter") {
+  deps = [
+    ":bounded_io",
+    ":data",
+    ":tlv_trait",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:stringutils",
+    "../common:internal_types",
+    "../common:math",
+    "../common:str_join",
+    "../public:types",
+  ]
+  sources = [
+    "parameter/add_incoming_streams_request_parameter.cc",
+    "parameter/add_incoming_streams_request_parameter.h",
+    "parameter/add_outgoing_streams_request_parameter.cc",
+    "parameter/add_outgoing_streams_request_parameter.h",
+    "parameter/forward_tsn_supported_parameter.cc",
+    "parameter/forward_tsn_supported_parameter.h",
+    "parameter/heartbeat_info_parameter.cc",
+    "parameter/heartbeat_info_parameter.h",
+    "parameter/incoming_ssn_reset_request_parameter.cc",
+    "parameter/incoming_ssn_reset_request_parameter.h",
+    "parameter/outgoing_ssn_reset_request_parameter.cc",
+    "parameter/outgoing_ssn_reset_request_parameter.h",
+    "parameter/parameter.cc",
+    "parameter/parameter.h",
+    "parameter/reconfiguration_response_parameter.cc",
+    "parameter/reconfiguration_response_parameter.h",
+    "parameter/ssn_tsn_reset_request_parameter.cc",
+    "parameter/ssn_tsn_reset_request_parameter.h",
+    "parameter/state_cookie_parameter.cc",
+    "parameter/state_cookie_parameter.h",
+    "parameter/supported_extensions_parameter.cc",
+    "parameter/supported_extensions_parameter.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("error_cause") {
+  deps = [
+    ":data",
+    ":parameter",
+    ":tlv_trait",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:stringutils",
+    "../common:internal_types",
+    "../common:math",
+    "../common:str_join",
+    "../packet:bounded_io",
+    "../public:types",
+  ]
+  sources = [
+    "error_cause/cookie_received_while_shutting_down_cause.cc",
+    "error_cause/cookie_received_while_shutting_down_cause.h",
+    "error_cause/error_cause.cc",
+    "error_cause/error_cause.h",
+    "error_cause/invalid_mandatory_parameter_cause.cc",
+    "error_cause/invalid_mandatory_parameter_cause.h",
+    "error_cause/invalid_stream_identifier_cause.cc",
+    "error_cause/invalid_stream_identifier_cause.h",
+    "error_cause/missing_mandatory_parameter_cause.cc",
+    "error_cause/missing_mandatory_parameter_cause.h",
+    "error_cause/no_user_data_cause.cc",
+    "error_cause/no_user_data_cause.h",
+    "error_cause/out_of_resource_error_cause.cc",
+    "error_cause/out_of_resource_error_cause.h",
+    "error_cause/protocol_violation_cause.cc",
+    "error_cause/protocol_violation_cause.h",
+    "error_cause/restart_of_an_association_with_new_address_cause.cc",
+    "error_cause/restart_of_an_association_with_new_address_cause.h",
+    "error_cause/stale_cookie_error_cause.cc",
+    "error_cause/stale_cookie_error_cause.h",
+    "error_cause/unrecognized_chunk_type_cause.cc",
+    "error_cause/unrecognized_chunk_type_cause.h",
+    "error_cause/unrecognized_parameter_cause.cc",
+    "error_cause/unrecognized_parameter_cause.h",
+    "error_cause/unresolvable_address_cause.cc",
+    "error_cause/unresolvable_address_cause.h",
+    "error_cause/user_initiated_abort_cause.cc",
+    "error_cause/user_initiated_abort_cause.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("chunk") {
+  deps = [
+    ":data",
+    ":error_cause",
+    ":parameter",
+    ":tlv_trait",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:stringutils",
+    "../common:math",
+    "../common:str_join",
+    "../packet:bounded_io",
+  ]
+  sources = [
+    "chunk/abort_chunk.cc",
+    "chunk/abort_chunk.h",
+    "chunk/chunk.cc",
+    "chunk/chunk.h",
+    "chunk/cookie_ack_chunk.cc",
+    "chunk/cookie_ack_chunk.h",
+    "chunk/cookie_echo_chunk.cc",
+    "chunk/cookie_echo_chunk.h",
+    "chunk/data_chunk.cc",
+    "chunk/data_chunk.h",
+    "chunk/data_common.h",
+    "chunk/error_chunk.cc",
+    "chunk/error_chunk.h",
+    "chunk/forward_tsn_chunk.cc",
+    "chunk/forward_tsn_chunk.h",
+    "chunk/forward_tsn_common.h",
+    "chunk/heartbeat_ack_chunk.cc",
+    "chunk/heartbeat_ack_chunk.h",
+    "chunk/heartbeat_request_chunk.cc",
+    "chunk/heartbeat_request_chunk.h",
+    "chunk/idata_chunk.cc",
+    "chunk/idata_chunk.h",
+    "chunk/iforward_tsn_chunk.cc",
+    "chunk/iforward_tsn_chunk.h",
+    "chunk/init_ack_chunk.cc",
+    "chunk/init_ack_chunk.h",
+    "chunk/init_chunk.cc",
+    "chunk/init_chunk.h",
+    "chunk/reconfig_chunk.cc",
+    "chunk/reconfig_chunk.h",
+    "chunk/sack_chunk.cc",
+    "chunk/sack_chunk.h",
+    "chunk/shutdown_ack_chunk.cc",
+    "chunk/shutdown_ack_chunk.h",
+    "chunk/shutdown_chunk.cc",
+    "chunk/shutdown_chunk.h",
+    "chunk/shutdown_complete_chunk.cc",
+    "chunk/shutdown_complete_chunk.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("chunk_validators") {
+  deps = [
+    ":chunk",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+  ]
+  sources = [
+    "chunk_validators.cc",
+    "chunk_validators.h",
+  ]
+}
+
+rtc_library("sctp_packet") {
+  deps = [
+    ":bounded_io",
+    ":chunk",
+    ":crc32c",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:stringutils",
+    "../common:internal_types",
+    "../common:math",
+    "../public:types",
+  ]
+  sources = [
+    "sctp_packet.cc",
+    "sctp_packet.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory:memory",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("dcsctp_packet_unittests") {
+    testonly = true
+
+    deps = [
+      ":bounded_io",
+      ":chunk",
+      ":chunk_validators",
+      ":crc32c",
+      ":error_cause",
+      ":parameter",
+      ":sctp_packet",
+      ":tlv_trait",
+      "../../../api:array_view",
+      "../../../rtc_base:buffer",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:gunit_helpers",
+      "../../../test:test_support",
+      "../common:internal_types",
+      "../common:math",
+      "../public:types",
+      "../testing:testing_macros",
+    ]
+    sources = [
+      "bounded_byte_reader_test.cc",
+      "bounded_byte_writer_test.cc",
+      "chunk/abort_chunk_test.cc",
+      "chunk/cookie_ack_chunk_test.cc",
+      "chunk/cookie_echo_chunk_test.cc",
+      "chunk/data_chunk_test.cc",
+      "chunk/error_chunk_test.cc",
+      "chunk/forward_tsn_chunk_test.cc",
+      "chunk/heartbeat_ack_chunk_test.cc",
+      "chunk/heartbeat_request_chunk_test.cc",
+      "chunk/idata_chunk_test.cc",
+      "chunk/iforward_tsn_chunk_test.cc",
+      "chunk/init_ack_chunk_test.cc",
+      "chunk/init_chunk_test.cc",
+      "chunk/reconfig_chunk_test.cc",
+      "chunk/sack_chunk_test.cc",
+      "chunk/shutdown_ack_chunk_test.cc",
+      "chunk/shutdown_chunk_test.cc",
+      "chunk/shutdown_complete_chunk_test.cc",
+      "chunk_validators_test.cc",
+      "crc32c_test.cc",
+      "error_cause/cookie_received_while_shutting_down_cause_test.cc",
+      "error_cause/invalid_mandatory_parameter_cause_test.cc",
+      "error_cause/invalid_stream_identifier_cause_test.cc",
+      "error_cause/missing_mandatory_parameter_cause_test.cc",
+      "error_cause/no_user_data_cause_test.cc",
+      "error_cause/out_of_resource_error_cause_test.cc",
+      "error_cause/protocol_violation_cause_test.cc",
+      "error_cause/restart_of_an_association_with_new_address_cause_test.cc",
+      "error_cause/stale_cookie_error_cause_test.cc",
+      "error_cause/unrecognized_chunk_type_cause_test.cc",
+      "error_cause/unrecognized_parameter_cause_test.cc",
+      "error_cause/unresolvable_address_cause_test.cc",
+      "error_cause/user_initiated_abort_cause_test.cc",
+      "parameter/add_incoming_streams_request_parameter_test.cc",
+      "parameter/add_outgoing_streams_request_parameter_test.cc",
+      "parameter/forward_tsn_supported_parameter_test.cc",
+      "parameter/incoming_ssn_reset_request_parameter_test.cc",
+      "parameter/outgoing_ssn_reset_request_parameter_test.cc",
+      "parameter/parameter_test.cc",
+      "parameter/reconfiguration_response_parameter_test.cc",
+      "parameter/ssn_tsn_reset_request_parameter_test.cc",
+      "parameter/state_cookie_parameter_test.cc",
+      "parameter/supported_extensions_parameter_test.cc",
+      "sctp_packet_test.cc",
+      "tlv_trait_test.cc",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+}
diff --git a/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_reader.h b/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_reader.h
new file mode 100644
index 0000000000..603ed6ac33
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_reader.h
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef NET_DCSCTP_PACKET_BOUNDED_BYTE_READER_H_
+#define NET_DCSCTP_PACKET_BOUNDED_BYTE_READER_H_
+
+#include <cstdint>
+
+#include "api/array_view.h"
+
+namespace dcsctp {
+
+// TODO(boivie): These generic functions - and possibly this entire class -
+// could be a candidate to have added to rtc_base/. They should use compiler
+// intrinsics as well.
+namespace internal {
+// Loads a 8-bit unsigned word at `data`.
+inline uint8_t LoadBigEndian8(const uint8_t* data) {
+  return data[0];
+}
+
+// Loads a 16-bit unsigned word at `data`.
+inline uint16_t LoadBigEndian16(const uint8_t* data) {
+  return (data[0] << 8) | data[1];
+}
+
+// Loads a 32-bit unsigned word at `data`.
+inline uint32_t LoadBigEndian32(const uint8_t* data) {
+  return (data[0] << 24) | (data[1] << 16) | (data[2] << 8) | data[3];
+}
+}  // namespace internal
+
+// BoundedByteReader wraps an ArrayView and divides it into two parts; A fixed
+// size - which is the template parameter - and a variable size, which is what
+// remains in `data` after the `FixedSize`.
+//
+// The BoundedByteReader provides methods to load/read big endian numbers from
+// the FixedSize portion of the buffer, and these are read with static bounds
+// checking, to avoid out-of-bounds accesses without a run-time penalty.
+//
+// The variable sized portion can either be used to create sub-readers, which
+// themselves would provide compile-time bounds-checking, or the entire variable
+// sized portion can be retrieved as an ArrayView.
+template <int FixedSize>
+class BoundedByteReader {
+ public:
+  explicit BoundedByteReader(rtc::ArrayView<const uint8_t> data) : data_(data) {
+    RTC_CHECK(data.size() >= FixedSize);
+  }
+
+  template <size_t offset>
+  uint8_t Load8() const {
+    static_assert(offset + sizeof(uint8_t) <= FixedSize, "Out-of-bounds");
+    return internal::LoadBigEndian8(&data_[offset]);
+  }
+
+  template <size_t offset>
+  uint16_t Load16() const {
+    static_assert(offset + sizeof(uint16_t) <= FixedSize, "Out-of-bounds");
+    static_assert((offset % sizeof(uint16_t)) == 0, "Unaligned access");
+    return internal::LoadBigEndian16(&data_[offset]);
+  }
+
+  template <size_t offset>
+  uint32_t Load32() const {
+    static_assert(offset + sizeof(uint32_t) <= FixedSize, "Out-of-bounds");
+    static_assert((offset % sizeof(uint32_t)) == 0, "Unaligned access");
+    return internal::LoadBigEndian32(&data_[offset]);
+  }
+
+  template <size_t SubSize>
+  BoundedByteReader<SubSize> sub_reader(size_t variable_offset) const {
+    RTC_CHECK(FixedSize + variable_offset + SubSize <= data_.size());
+
+    rtc::ArrayView<const uint8_t> sub_span =
+        data_.subview(FixedSize + variable_offset, SubSize);
+    return BoundedByteReader<SubSize>(sub_span);
+  }
+
+  size_t variable_data_size() const { return data_.size() - FixedSize; }
+
+  rtc::ArrayView<const uint8_t> variable_data() const {
+    return data_.subview(FixedSize, data_.size() - FixedSize);
+  }
+
+ private:
+  const rtc::ArrayView<const uint8_t> data_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_BOUNDED_BYTE_READER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_reader_test.cc b/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_reader_test.cc
new file mode 100644
index 0000000000..2fb4a86785
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_reader_test.cc
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(BoundedByteReaderTest, CanLoadData) {
+  uint8_t data[14] = {1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4};
+
+  BoundedByteReader<8> reader(data);
+  EXPECT_EQ(reader.variable_data_size(), 6U);
+  EXPECT_EQ(reader.Load32<0>(), 0x01020304U);
+  EXPECT_EQ(reader.Load32<4>(), 0x05060708U);
+  EXPECT_EQ(reader.Load16<4>(), 0x0506U);
+  EXPECT_EQ(reader.Load8<4>(), 0x05U);
+  EXPECT_EQ(reader.Load8<5>(), 0x06U);
+
+  BoundedByteReader<6> sub = reader.sub_reader<6>(0);
+  EXPECT_EQ(sub.Load16<0>(), 0x0900U);
+  EXPECT_EQ(sub.Load32<0>(), 0x09000102U);
+  EXPECT_EQ(sub.Load16<4>(), 0x0304U);
+
+  EXPECT_THAT(reader.variable_data(), ElementsAre(9, 0, 1, 2, 3, 4));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_writer.h b/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_writer.h
new file mode 100644
index 0000000000..d754549e4f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_writer.h
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef NET_DCSCTP_PACKET_BOUNDED_BYTE_WRITER_H_
+#define NET_DCSCTP_PACKET_BOUNDED_BYTE_WRITER_H_
+
+#include <algorithm>
+
+#include "api/array_view.h"
+
+namespace dcsctp {
+
+// TODO(boivie): These generic functions - and possibly this entire class -
+// could be a candidate to have added to rtc_base/. They should use compiler
+// intrinsics as well.
+namespace internal {
+// Stores a 8-bit unsigned word at `data`.
+inline void StoreBigEndian8(uint8_t* data, uint8_t val) {
+  data[0] = val;
+}
+
+// Stores a 16-bit unsigned word at `data`.
+inline void StoreBigEndian16(uint8_t* data, uint16_t val) {
+  data[0] = val >> 8;
+  data[1] = val;
+}
+
+// Stores a 32-bit unsigned word at `data`.
+inline void StoreBigEndian32(uint8_t* data, uint32_t val) {
+  data[0] = val >> 24;
+  data[1] = val >> 16;
+  data[2] = val >> 8;
+  data[3] = val;
+}
+}  // namespace internal
+
+// BoundedByteWriter wraps an ArrayView and divides it into two parts; A fixed
+// size - which is the template parameter - and a variable size, which is what
+// remains in `data` after the `FixedSize`.
+//
+// The BoundedByteWriter provides methods to write big endian numbers to the
+// FixedSize portion of the buffer, and these are written with static bounds
+// checking, to avoid out-of-bounds accesses without a run-time penalty.
+//
+// The variable sized portion can either be used to create sub-writers, which
+// themselves would provide compile-time bounds-checking, or data can be copied
+// to it.
+template <int FixedSize>
+class BoundedByteWriter {
+ public:
+  explicit BoundedByteWriter(rtc::ArrayView<uint8_t> data) : data_(data) {
+    RTC_CHECK(data.size() >= FixedSize);
+  }
+
+  template <size_t offset>
+  void Store8(uint8_t value) {
+    static_assert(offset + sizeof(uint8_t) <= FixedSize, "Out-of-bounds");
+    internal::StoreBigEndian8(&data_[offset], value);
+  }
+
+  template <size_t offset>
+  void Store16(uint16_t value) {
+    static_assert(offset + sizeof(uint16_t) <= FixedSize, "Out-of-bounds");
+    static_assert((offset % sizeof(uint16_t)) == 0, "Unaligned access");
+    internal::StoreBigEndian16(&data_[offset], value);
+  }
+
+  template <size_t offset>
+  void Store32(uint32_t value) {
+    static_assert(offset + sizeof(uint32_t) <= FixedSize, "Out-of-bounds");
+    static_assert((offset % sizeof(uint32_t)) == 0, "Unaligned access");
+    internal::StoreBigEndian32(&data_[offset], value);
+  }
+
+  template <size_t SubSize>
+  BoundedByteWriter<SubSize> sub_writer(size_t variable_offset) {
+    RTC_CHECK(FixedSize + variable_offset + SubSize <= data_.size());
+
+    return BoundedByteWriter<SubSize>(
+        data_.subview(FixedSize + variable_offset, SubSize));
+  }
+
+  void CopyToVariableData(rtc::ArrayView<const uint8_t> source) {
+    size_t copy_size = std::min(source.size(), data_.size() - FixedSize);
+    if (source.data() == nullptr || copy_size == 0) {
+      return;
+    }
+    memcpy(data_.data() + FixedSize, source.data(), copy_size);
+  }
+
+ private:
+  rtc::ArrayView<uint8_t> data_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_BOUNDED_BYTE_WRITER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_writer_test.cc b/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_writer_test.cc
new file mode 100644
index 0000000000..3cea0a2f7c
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/bounded_byte_writer_test.cc
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(BoundedByteWriterTest, CanWriteData) {
+  std::vector<uint8_t> data(14);
+
+  BoundedByteWriter<8> writer(data);
+  writer.Store32<0>(0x01020304);
+  writer.Store16<4>(0x0506);
+  writer.Store8<6>(0x07);
+  writer.Store8<7>(0x08);
+
+  uint8_t variable_data[] = {0, 0, 0, 0, 3, 0};
+  writer.CopyToVariableData(variable_data);
+
+  BoundedByteWriter<6> sub = writer.sub_writer<6>(0);
+  sub.Store32<0>(0x09000000);
+  sub.Store16<2>(0x0102);
+
+  BoundedByteWriter<2> sub2 = writer.sub_writer<2>(4);
+  sub2.Store8<1>(0x04);
+
+  EXPECT_THAT(data, ElementsAre(1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/abort_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/abort_chunk.cc
new file mode 100644
index 0000000000..8348eb96a9
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/abort_chunk.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/abort_chunk.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.7
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 6    |Reserved     |T|           Length              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /                   zero or more Error Causes                   /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int AbortChunk::kType;
+
+absl::optional<AbortChunk> AbortChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  absl::optional<Parameters> error_causes =
+      Parameters::Parse(reader->variable_data());
+  if (!error_causes.has_value()) {
+    return absl::nullopt;
+  }
+  uint8_t flags = reader->Load8<1>();
+  bool filled_in_verification_tag = (flags & (1 << kFlagsBitT)) == 0;
+  return AbortChunk(filled_in_verification_tag, *std::move(error_causes));
+}
+
+void AbortChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  rtc::ArrayView<const uint8_t> error_causes = error_causes_.data();
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, error_causes.size());
+  writer.Store8<1>(filled_in_verification_tag_ ? 0 : (1 << kFlagsBitT));
+  writer.CopyToVariableData(error_causes);
+}
+
+std::string AbortChunk::ToString() const {
+  return "ABORT";
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/abort_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/abort_chunk.h
new file mode 100644
index 0000000000..1408a75e80
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/abort_chunk.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_ABORT_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_ABORT_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.7
+struct AbortChunkConfig : ChunkConfig {
+  static constexpr int kType = 6;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class AbortChunk : public Chunk, public TLVTrait<AbortChunkConfig> {
+ public:
+  static constexpr int kType = AbortChunkConfig::kType;
+
+  AbortChunk(bool filled_in_verification_tag, Parameters error_causes)
+      : filled_in_verification_tag_(filled_in_verification_tag),
+        error_causes_(std::move(error_causes)) {}
+
+  AbortChunk(AbortChunk&& other) = default;
+  AbortChunk& operator=(AbortChunk&& other) = default;
+
+  static absl::optional<AbortChunk> Parse(rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  bool filled_in_verification_tag() const {
+    return filled_in_verification_tag_;
+  }
+
+  const Parameters& error_causes() const { return error_causes_; }
+
+ private:
+  static constexpr int kFlagsBitT = 0;
+  bool filled_in_verification_tag_;
+  Parameters error_causes_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_ABORT_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/abort_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/abort_chunk_test.cc
new file mode 100644
index 0000000000..c1f3a4d5b9
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/abort_chunk_test.cc
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/abort_chunk.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/error_cause/user_initiated_abort_cause.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(AbortChunkTest, FromCapture) {
+  /*
+  ABORT chunk
+      Chunk type: ABORT (6)
+      Chunk flags: 0x00
+      Chunk length: 8
+      User initiated ABORT cause
+  */
+
+  uint8_t data[] = {0x06, 0x00, 0x00, 0x08, 0x00, 0x0c, 0x00, 0x04};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(AbortChunk chunk, AbortChunk::Parse(data));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      UserInitiatedAbortCause cause,
+      chunk.error_causes().get<UserInitiatedAbortCause>());
+
+  EXPECT_EQ(cause.upper_layer_abort_reason(), "");
+}
+
+TEST(AbortChunkTest, SerializeAndDeserialize) {
+  AbortChunk chunk(/*filled_in_verification_tag=*/true,
+                   Parameters::Builder()
+                       .Add(UserInitiatedAbortCause("Close called"))
+                       .Build());
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(AbortChunk deserialized,
+                              AbortChunk::Parse(serialized));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      UserInitiatedAbortCause cause,
+      deserialized.error_causes().get<UserInitiatedAbortCause>());
+
+  EXPECT_EQ(cause.upper_layer_abort_reason(), "Close called");
+}
+
+// Validates that AbortChunk doesn't make any alignment assumptions.
+TEST(AbortChunkTest, SerializeAndDeserializeOneChar) {
+  AbortChunk chunk(
+      /*filled_in_verification_tag=*/true,
+      Parameters::Builder().Add(UserInitiatedAbortCause("!")).Build());
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(AbortChunk deserialized,
+                              AbortChunk::Parse(serialized));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      UserInitiatedAbortCause cause,
+      deserialized.error_causes().get<UserInitiatedAbortCause>());
+
+  EXPECT_EQ(cause.upper_layer_abort_reason(), "!");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/chunk.cc
new file mode 100644
index 0000000000..832ab82288
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/chunk.cc
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/chunk.h"
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/math.h"
+#include "net/dcsctp/packet/chunk/abort_chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/error_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/init_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/init_chunk.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_complete_chunk.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+template <class Chunk>
+bool ParseAndPrint(uint8_t chunk_type,
+                   rtc::ArrayView<const uint8_t> data,
+                   rtc::StringBuilder& sb) {
+  if (chunk_type == Chunk::kType) {
+    absl::optional<Chunk> c = Chunk::Parse(data);
+    if (c.has_value()) {
+      sb << c->ToString();
+    } else {
+      sb << "Failed to parse chunk of type " << chunk_type;
+    }
+    return true;
+  }
+  return false;
+}
+
+std::string DebugConvertChunkToString(rtc::ArrayView<const uint8_t> data) {
+  rtc::StringBuilder sb;
+
+  if (data.empty()) {
+    sb << "Failed to parse chunk due to empty data";
+  } else {
+    uint8_t chunk_type = data[0];
+    if (!ParseAndPrint<DataChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<InitChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<InitAckChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<SackChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<HeartbeatRequestChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<HeartbeatAckChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<AbortChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<ErrorChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<CookieEchoChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<CookieAckChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<ShutdownChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<ShutdownAckChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<ShutdownCompleteChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<ReConfigChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<ForwardTsnChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<IDataChunk>(chunk_type, data, sb) &&
+        !ParseAndPrint<IForwardTsnChunk>(chunk_type, data, sb)) {
+      sb << "Unhandled chunk type: " << static_cast<int>(chunk_type);
+    }
+  }
+  return sb.Release();
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/chunk.h
new file mode 100644
index 0000000000..687aa1daa1
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/chunk.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_CHUNK_H_
+
+#include <stddef.h>
+#include <sys/types.h>
+
+#include <cstdint>
+#include <iterator>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// Base class for all SCTP chunks
+class Chunk {
+ public:
+  Chunk() {}
+  virtual ~Chunk() = default;
+
+  // Chunks can contain data payloads that shouldn't be copied unnecessarily.
+  Chunk(Chunk&& other) = default;
+  Chunk& operator=(Chunk&& other) = default;
+  Chunk(const Chunk&) = delete;
+  Chunk& operator=(const Chunk&) = delete;
+
+  // Serializes the chunk to `out`, growing it as necessary.
+  virtual void SerializeTo(std::vector<uint8_t>& out) const = 0;
+
+  // Returns a human readable description of this chunk and its parameters.
+  virtual std::string ToString() const = 0;
+};
+
+// Introspects the chunk in `data` and returns a human readable textual
+// representation of it, to be used in debugging.
+std::string DebugConvertChunkToString(rtc::ArrayView<const uint8_t> data);
+
+struct ChunkConfig {
+  static constexpr int kTypeSizeInBytes = 1;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_ack_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_ack_chunk.cc
new file mode 100644
index 0000000000..4839969ccf
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_ack_chunk.cc
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.12
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 11   |Chunk  Flags   |     Length = 4                |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int CookieAckChunk::kType;
+
+absl::optional<CookieAckChunk> CookieAckChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  if (!ParseTLV(data).has_value()) {
+    return absl::nullopt;
+  }
+  return CookieAckChunk();
+}
+
+void CookieAckChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  AllocateTLV(out);
+}
+
+std::string CookieAckChunk::ToString() const {
+  return "COOKIE-ACK";
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_ack_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_ack_chunk.h
new file mode 100644
index 0000000000..f7d4a33f7d
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_ack_chunk.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_COOKIE_ACK_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_COOKIE_ACK_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.12
+struct CookieAckChunkConfig : ChunkConfig {
+  static constexpr int kType = 11;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class CookieAckChunk : public Chunk, public TLVTrait<CookieAckChunkConfig> {
+ public:
+  static constexpr int kType = CookieAckChunkConfig::kType;
+
+  CookieAckChunk() {}
+
+  static absl::optional<CookieAckChunk> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_COOKIE_ACK_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_ack_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_ack_chunk_test.cc
new file mode 100644
index 0000000000..3f560c6fef
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_ack_chunk_test.cc
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(CookieAckChunkTest, FromCapture) {
+  /*
+  COOKIE_ACK chunk
+      Chunk type: COOKIE_ACK (11)
+      Chunk flags: 0x00
+      Chunk length: 4
+  */
+
+  uint8_t data[] = {0x0b, 0x00, 0x00, 0x04};
+
+  EXPECT_TRUE(CookieAckChunk::Parse(data).has_value());
+}
+
+TEST(CookieAckChunkTest, SerializeAndDeserialize) {
+  CookieAckChunk chunk;
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(CookieAckChunk deserialized,
+                              CookieAckChunk::Parse(serialized));
+  EXPECT_EQ(deserialized.ToString(), "COOKIE-ACK");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_echo_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_echo_chunk.cc
new file mode 100644
index 0000000000..a01d0b13c4
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_echo_chunk.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.11
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 10   |Chunk  Flags   |         Length                |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                     Cookie                                    /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int CookieEchoChunk::kType;
+
+absl::optional<CookieEchoChunk> CookieEchoChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  return CookieEchoChunk(reader->variable_data());
+}
+
+void CookieEchoChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, cookie_.size());
+  writer.CopyToVariableData(cookie_);
+}
+
+std::string CookieEchoChunk::ToString() const {
+  return "COOKIE-ECHO";
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_echo_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_echo_chunk.h
new file mode 100644
index 0000000000..8cb80527f8
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_echo_chunk.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_COOKIE_ECHO_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_COOKIE_ECHO_CHUNK_H_
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.11
+struct CookieEchoChunkConfig : ChunkConfig {
+  static constexpr int kType = 10;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class CookieEchoChunk : public Chunk, public TLVTrait<CookieEchoChunkConfig> {
+ public:
+  static constexpr int kType = CookieEchoChunkConfig::kType;
+
+  explicit CookieEchoChunk(rtc::ArrayView<const uint8_t> cookie)
+      : cookie_(cookie.begin(), cookie.end()) {}
+
+  static absl::optional<CookieEchoChunk> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  rtc::ArrayView<const uint8_t> cookie() const { return cookie_; }
+
+ private:
+  std::vector<uint8_t> cookie_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_COOKIE_ECHO_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_echo_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_echo_chunk_test.cc
new file mode 100644
index 0000000000..d06e0a6439
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/cookie_echo_chunk_test.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(CookieEchoChunkTest, FromCapture) {
+  /*
+  COOKIE_ECHO chunk (Cookie length: 256 bytes)
+      Chunk type: COOKIE_ECHO (10)
+      Chunk flags: 0x00
+      Chunk length: 260
+      Cookie: 12345678
+  */
+
+  uint8_t data[] = {0x0a, 0x00, 0x00, 0x08, 0x12, 0x34, 0x56, 0x78};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(CookieEchoChunk chunk,
+                              CookieEchoChunk::Parse(data));
+
+  EXPECT_THAT(chunk.cookie(), ElementsAre(0x12, 0x34, 0x56, 0x78));
+}
+
+TEST(CookieEchoChunkTest, SerializeAndDeserialize) {
+  uint8_t cookie[] = {1, 2, 3, 4};
+  CookieEchoChunk chunk(cookie);
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(CookieEchoChunk deserialized,
+                              CookieEchoChunk::Parse(serialized));
+
+  EXPECT_THAT(deserialized.cookie(), ElementsAre(1, 2, 3, 4));
+  EXPECT_EQ(deserialized.ToString(), "COOKIE-ECHO");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/data_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/data_chunk.cc
new file mode 100644
index 0000000000..769be2db91
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/data_chunk.cc
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/chunk/data_common.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.1
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 0    | Reserved|U|B|E|    Length                     |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                              TSN                              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |      Stream Identifier S      |   Stream Sequence Number n    |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  Payload Protocol Identifier                  |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /                 User Data (seq n of Stream S)                 /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int DataChunk::kType;
+
+absl::optional<DataChunk> DataChunk::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  uint8_t flags = reader->Load8<1>();
+  TSN tsn(reader->Load32<4>());
+  StreamID stream_identifier(reader->Load16<8>());
+  SSN ssn(reader->Load16<10>());
+  PPID ppid(reader->Load32<12>());
+
+  Options options;
+  options.is_end = Data::IsEnd((flags & (1 << kFlagsBitEnd)) != 0);
+  options.is_beginning =
+      Data::IsBeginning((flags & (1 << kFlagsBitBeginning)) != 0);
+  options.is_unordered = IsUnordered((flags & (1 << kFlagsBitUnordered)) != 0);
+  options.immediate_ack =
+      ImmediateAckFlag((flags & (1 << kFlagsBitImmediateAck)) != 0);
+
+  return DataChunk(tsn, stream_identifier, ssn, ppid,
+                   std::vector<uint8_t>(reader->variable_data().begin(),
+                                        reader->variable_data().end()),
+                   options);
+}
+
+void DataChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, payload().size());
+
+  writer.Store8<1>(
+      (*options().is_end ? (1 << kFlagsBitEnd) : 0) |
+      (*options().is_beginning ? (1 << kFlagsBitBeginning) : 0) |
+      (*options().is_unordered ? (1 << kFlagsBitUnordered) : 0) |
+      (*options().immediate_ack ? (1 << kFlagsBitImmediateAck) : 0));
+  writer.Store32<4>(*tsn());
+  writer.Store16<8>(*stream_id());
+  writer.Store16<10>(*ssn());
+  writer.Store32<12>(*ppid());
+
+  writer.CopyToVariableData(payload());
+}
+
+std::string DataChunk::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "DATA, type=" << (options().is_unordered ? "unordered" : "ordered")
+     << "::"
+     << (*options().is_beginning && *options().is_end
+             ? "complete"
+             : *options().is_beginning ? "first"
+                                       : *options().is_end ? "last" : "middle")
+     << ", tsn=" << *tsn() << ", sid=" << *stream_id() << ", ssn=" << *ssn()
+     << ", ppid=" << *ppid() << ", length=" << payload().size();
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/data_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/data_chunk.h
new file mode 100644
index 0000000000..12bb05f2c4
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/data_chunk.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_DATA_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_DATA_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/chunk/data_common.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.1
+struct DataChunkConfig : ChunkConfig {
+  static constexpr int kType = 0;
+  static constexpr size_t kHeaderSize = 16;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class DataChunk : public AnyDataChunk, public TLVTrait<DataChunkConfig> {
+ public:
+  static constexpr int kType = DataChunkConfig::kType;
+
+  // Exposed to allow the retransmission queue to make room for the correct
+  // header size.
+  static constexpr size_t kHeaderSize = DataChunkConfig::kHeaderSize;
+
+  DataChunk(TSN tsn,
+            StreamID stream_id,
+            SSN ssn,
+            PPID ppid,
+            std::vector<uint8_t> payload,
+            const Options& options)
+      : AnyDataChunk(tsn,
+                     stream_id,
+                     ssn,
+                     MID(0),
+                     FSN(0),
+                     ppid,
+                     std::move(payload),
+                     options) {}
+
+  DataChunk(TSN tsn, Data&& data, bool immediate_ack)
+      : AnyDataChunk(tsn, std::move(data), immediate_ack) {}
+
+  static absl::optional<DataChunk> Parse(rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_DATA_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/data_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/data_chunk_test.cc
new file mode 100644
index 0000000000..def99ceb23
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/data_chunk_test.cc
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(DataChunkTest, FromCapture) {
+  /*
+  DATA chunk(ordered, complete segment, TSN: 1426601532, SID: 2, SSN: 1,
+    PPID: 53, payload length: 4 bytes)
+      Chunk type: DATA (0)
+      Chunk flags: 0x03
+      Chunk length: 20
+      Transmission sequence number: 1426601532
+      Stream identifier: 0x0002
+      Stream sequence number: 1
+      Payload protocol identifier: WebRTC Binary (53)
+  */
+
+  uint8_t data[] = {0x00, 0x03, 0x00, 0x14, 0x55, 0x08, 0x36, 0x3c, 0x00, 0x02,
+                    0x00, 0x01, 0x00, 0x00, 0x00, 0x35, 0x00, 0x01, 0x02, 0x03};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(DataChunk chunk, DataChunk::Parse(data));
+  EXPECT_EQ(*chunk.tsn(), 1426601532u);
+  EXPECT_EQ(*chunk.stream_id(), 2u);
+  EXPECT_EQ(*chunk.ssn(), 1u);
+  EXPECT_EQ(*chunk.ppid(), 53u);
+  EXPECT_TRUE(*chunk.options().is_beginning);
+  EXPECT_TRUE(*chunk.options().is_end);
+  EXPECT_FALSE(*chunk.options().is_unordered);
+  EXPECT_FALSE(*chunk.options().immediate_ack);
+  EXPECT_THAT(chunk.payload(), ElementsAre(0x0, 0x1, 0x2, 0x3));
+}
+
+TEST(DataChunkTest, SerializeAndDeserialize) {
+  DataChunk chunk(TSN(123), StreamID(456), SSN(789), PPID(9090),
+                  /*payload=*/{1, 2, 3, 4, 5},
+                  /*options=*/{});
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(DataChunk deserialized,
+                              DataChunk::Parse(serialized));
+  EXPECT_EQ(*chunk.tsn(), 123u);
+  EXPECT_EQ(*chunk.stream_id(), 456u);
+  EXPECT_EQ(*chunk.ssn(), 789u);
+  EXPECT_EQ(*chunk.ppid(), 9090u);
+  EXPECT_THAT(chunk.payload(), ElementsAre(1, 2, 3, 4, 5));
+
+  EXPECT_EQ(deserialized.ToString(),
+            "DATA, type=ordered::middle, tsn=123, sid=456, ssn=789, ppid=9090, "
+            "length=5");
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/data_common.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/data_common.h
new file mode 100644
index 0000000000..b67efeee1e
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/data_common.h
@@ -0,0 +1,97 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_DATA_COMMON_H_
+#define NET_DCSCTP_PACKET_CHUNK_DATA_COMMON_H_
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/data.h"
+
+namespace dcsctp {
+
+// Base class for DataChunk and IDataChunk
+class AnyDataChunk : public Chunk {
+ public:
+  // Represents the "immediate ack" flag on DATA/I-DATA, from RFC7053.
+  using ImmediateAckFlag = webrtc::StrongAlias<class ImmediateAckFlagTag, bool>;
+
+  // Data chunk options.
+  // See https://tools.ietf.org/html/rfc4960#section-3.3.1
+  struct Options {
+    Data::IsEnd is_end = Data::IsEnd(false);
+    Data::IsBeginning is_beginning = Data::IsBeginning(false);
+    IsUnordered is_unordered = IsUnordered(false);
+    ImmediateAckFlag immediate_ack = ImmediateAckFlag(false);
+  };
+
+  TSN tsn() const { return tsn_; }
+
+  Options options() const {
+    Options options;
+    options.is_end = data_.is_end;
+    options.is_beginning = data_.is_beginning;
+    options.is_unordered = data_.is_unordered;
+    options.immediate_ack = immediate_ack_;
+    return options;
+  }
+
+  StreamID stream_id() const { return data_.stream_id; }
+  SSN ssn() const { return data_.ssn; }
+  MID message_id() const { return data_.message_id; }
+  FSN fsn() const { return data_.fsn; }
+  PPID ppid() const { return data_.ppid; }
+  rtc::ArrayView<const uint8_t> payload() const { return data_.payload; }
+
+  // Extracts the Data from the chunk, as a destructive action.
+  Data extract() && { return std::move(data_); }
+
+  AnyDataChunk(TSN tsn,
+               StreamID stream_id,
+               SSN ssn,
+               MID message_id,
+               FSN fsn,
+               PPID ppid,
+               std::vector<uint8_t> payload,
+               const Options& options)
+      : tsn_(tsn),
+        data_(stream_id,
+              ssn,
+              message_id,
+              fsn,
+              ppid,
+              std::move(payload),
+              options.is_beginning,
+              options.is_end,
+              options.is_unordered),
+        immediate_ack_(options.immediate_ack) {}
+
+  AnyDataChunk(TSN tsn, Data data, bool immediate_ack)
+      : tsn_(tsn), data_(std::move(data)), immediate_ack_(immediate_ack) {}
+
+ protected:
+  // Bits in `flags` header field.
+  static constexpr int kFlagsBitEnd = 0;
+  static constexpr int kFlagsBitBeginning = 1;
+  static constexpr int kFlagsBitUnordered = 2;
+  static constexpr int kFlagsBitImmediateAck = 3;
+
+ private:
+  TSN tsn_;
+  Data data_;
+  ImmediateAckFlag immediate_ack_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_DATA_COMMON_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/error_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/error_chunk.cc
new file mode 100644
index 0000000000..baac0c5588
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/error_chunk.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/error_chunk.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 9    | Chunk  Flags  |           Length              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /                    one or more Error Causes                   /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int ErrorChunk::kType;
+
+absl::optional<ErrorChunk> ErrorChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  absl::optional<Parameters> error_causes =
+      Parameters::Parse(reader->variable_data());
+  if (!error_causes.has_value()) {
+    return absl::nullopt;
+  }
+  return ErrorChunk(*std::move(error_causes));
+}
+
+void ErrorChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  rtc::ArrayView<const uint8_t> error_causes = error_causes_.data();
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, error_causes.size());
+  writer.CopyToVariableData(error_causes);
+}
+
+std::string ErrorChunk::ToString() const {
+  return "ERROR";
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/error_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/error_chunk.h
new file mode 100644
index 0000000000..96122cff6a
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/error_chunk.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_ERROR_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_ERROR_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10
+struct ErrorChunkConfig : ChunkConfig {
+  static constexpr int kType = 9;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 4;
+};
+
+class ErrorChunk : public Chunk, public TLVTrait<ErrorChunkConfig> {
+ public:
+  static constexpr int kType = ErrorChunkConfig::kType;
+
+  explicit ErrorChunk(Parameters error_causes)
+      : error_causes_(std::move(error_causes)) {}
+
+  ErrorChunk(ErrorChunk&& other) = default;
+  ErrorChunk& operator=(ErrorChunk&& other) = default;
+
+  static absl::optional<ErrorChunk> Parse(rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  const Parameters& error_causes() const { return error_causes_; }
+
+ private:
+  Parameters error_causes_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_ERROR_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/error_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/error_chunk_test.cc
new file mode 100644
index 0000000000..f2b8be1edc
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/error_chunk_test.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/error_chunk.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(ErrorChunkTest, FromCapture) {
+  /*
+   ERROR chunk
+      Chunk type: ERROR (9)
+      Chunk flags: 0x00
+      Chunk length: 12
+      Unrecognized chunk type cause (Type: 73 (unknown))
+  */
+
+  uint8_t data[] = {0x09, 0x00, 0x00, 0x0c, 0x00, 0x06,
+                    0x00, 0x08, 0x49, 0x00, 0x00, 0x04};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(ErrorChunk chunk, ErrorChunk::Parse(data));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      UnrecognizedChunkTypeCause cause,
+      chunk.error_causes().get<UnrecognizedChunkTypeCause>());
+
+  EXPECT_THAT(cause.unrecognized_chunk(), ElementsAre(0x49, 0x00, 0x00, 0x04));
+}
+
+TEST(ErrorChunkTest, SerializeAndDeserialize) {
+  ErrorChunk chunk(Parameters::Builder()
+                       .Add(UnrecognizedChunkTypeCause({1, 2, 3, 4}))
+                       .Build());
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(ErrorChunk deserialized,
+                              ErrorChunk::Parse(serialized));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      UnrecognizedChunkTypeCause cause,
+      deserialized.error_causes().get<UnrecognizedChunkTypeCause>());
+
+  EXPECT_THAT(cause.unrecognized_chunk(), ElementsAre(1, 2, 3, 4));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_chunk.cc
new file mode 100644
index 0000000000..e432114c50
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_chunk.cc
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc3758#section-3.2
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 192  |  Flags = 0x00 |        Length = Variable      |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                      New Cumulative TSN                       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |         Stream-1              |       Stream Sequence-1       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               /
+//  /                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |         Stream-N              |       Stream Sequence-N       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int ForwardTsnChunk::kType;
+
+absl::optional<ForwardTsnChunk> ForwardTsnChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  TSN new_cumulative_tsn(reader->Load32<4>());
+
+  size_t streams_skipped =
+      reader->variable_data_size() / kSkippedStreamBufferSize;
+
+  std::vector<SkippedStream> skipped_streams;
+  skipped_streams.reserve(streams_skipped);
+  for (size_t i = 0; i < streams_skipped; ++i) {
+    BoundedByteReader<kSkippedStreamBufferSize> sub_reader =
+        reader->sub_reader<kSkippedStreamBufferSize>(i *
+                                                     kSkippedStreamBufferSize);
+
+    StreamID stream_id(sub_reader.Load16<0>());
+    SSN ssn(sub_reader.Load16<2>());
+    skipped_streams.emplace_back(stream_id, ssn);
+  }
+  return ForwardTsnChunk(new_cumulative_tsn, std::move(skipped_streams));
+}
+
+void ForwardTsnChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  rtc::ArrayView<const SkippedStream> skipped = skipped_streams();
+  size_t variable_size = skipped.size() * kSkippedStreamBufferSize;
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, variable_size);
+
+  writer.Store32<4>(*new_cumulative_tsn());
+  for (size_t i = 0; i < skipped.size(); ++i) {
+    BoundedByteWriter<kSkippedStreamBufferSize> sub_writer =
+        writer.sub_writer<kSkippedStreamBufferSize>(i *
+                                                    kSkippedStreamBufferSize);
+    sub_writer.Store16<0>(*skipped[i].stream_id);
+    sub_writer.Store16<2>(*skipped[i].ssn);
+  }
+}
+
+std::string ForwardTsnChunk::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "FORWARD-TSN, new_cumulative_tsn=" << *new_cumulative_tsn();
+  for (const auto& skipped : skipped_streams()) {
+    sb << ", skip " << skipped.stream_id.value() << ":" << *skipped.ssn;
+  }
+  return sb.str();
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_chunk.h
new file mode 100644
index 0000000000..b9ef666f41
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_chunk.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_FORWARD_TSN_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_FORWARD_TSN_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc3758#section-3.2
+struct ForwardTsnChunkConfig : ChunkConfig {
+  static constexpr int kType = 192;
+  static constexpr size_t kHeaderSize = 8;
+  static constexpr size_t kVariableLengthAlignment = 4;
+};
+
+class ForwardTsnChunk : public AnyForwardTsnChunk,
+                        public TLVTrait<ForwardTsnChunkConfig> {
+ public:
+  static constexpr int kType = ForwardTsnChunkConfig::kType;
+
+  ForwardTsnChunk(TSN new_cumulative_tsn,
+                  std::vector<SkippedStream> skipped_streams)
+      : AnyForwardTsnChunk(new_cumulative_tsn, std::move(skipped_streams)) {}
+
+  static absl::optional<ForwardTsnChunk> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+ private:
+  static constexpr size_t kSkippedStreamBufferSize = 4;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_FORWARD_TSN_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_chunk_test.cc
new file mode 100644
index 0000000000..51f97f2396
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_chunk_test.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(ForwardTsnChunkTest, FromCapture) {
+  /*
+  FORWARD_TSN chunk(Cumulative TSN: 1905748778)
+      Chunk type: FORWARD_TSN (192)
+      Chunk flags: 0x00
+      Chunk length: 8
+      New cumulative TSN: 1905748778
+  */
+
+  uint8_t data[] = {0xc0, 0x00, 0x00, 0x08, 0x71, 0x97, 0x6b, 0x2a};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(ForwardTsnChunk chunk,
+                              ForwardTsnChunk::Parse(data));
+  EXPECT_EQ(*chunk.new_cumulative_tsn(), 1905748778u);
+}
+
+TEST(ForwardTsnChunkTest, SerializeAndDeserialize) {
+  ForwardTsnChunk chunk(
+      TSN(123), {ForwardTsnChunk::SkippedStream(StreamID(1), SSN(23)),
+                 ForwardTsnChunk::SkippedStream(StreamID(42), SSN(99))});
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(ForwardTsnChunk deserialized,
+                              ForwardTsnChunk::Parse(serialized));
+  EXPECT_EQ(*deserialized.new_cumulative_tsn(), 123u);
+  EXPECT_THAT(
+      deserialized.skipped_streams(),
+      ElementsAre(ForwardTsnChunk::SkippedStream(StreamID(1), SSN(23)),
+                  ForwardTsnChunk::SkippedStream(StreamID(42), SSN(99))));
+
+  EXPECT_EQ(deserialized.ToString(),
+            "FORWARD-TSN, new_cumulative_tsn=123, skip 1:23, skip 42:99");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_common.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_common.h
new file mode 100644
index 0000000000..37bd2aafff
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/forward_tsn_common.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_FORWARD_TSN_COMMON_H_
+#define NET_DCSCTP_PACKET_CHUNK_FORWARD_TSN_COMMON_H_
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+
+namespace dcsctp {
+
+// Base class for both ForwardTsnChunk and IForwardTsnChunk
+class AnyForwardTsnChunk : public Chunk {
+ public:
+  struct SkippedStream {
+    SkippedStream(StreamID stream_id, SSN ssn)
+        : stream_id(stream_id), ssn(ssn), unordered(false), message_id(0) {}
+    SkippedStream(IsUnordered unordered, StreamID stream_id, MID message_id)
+        : stream_id(stream_id),
+          ssn(0),
+          unordered(unordered),
+          message_id(message_id) {}
+
+    StreamID stream_id;
+
+    // Set for FORWARD_TSN
+    SSN ssn;
+
+    // Set for I-FORWARD_TSN
+    IsUnordered unordered;
+    MID message_id;
+
+    bool operator==(const SkippedStream& other) const {
+      return stream_id == other.stream_id && ssn == other.ssn &&
+             unordered == other.unordered && message_id == other.message_id;
+    }
+  };
+
+  AnyForwardTsnChunk(TSN new_cumulative_tsn,
+                     std::vector<SkippedStream> skipped_streams)
+      : new_cumulative_tsn_(new_cumulative_tsn),
+        skipped_streams_(std::move(skipped_streams)) {}
+
+  TSN new_cumulative_tsn() const { return new_cumulative_tsn_; }
+
+  rtc::ArrayView<const SkippedStream> skipped_streams() const {
+    return skipped_streams_;
+  }
+
+ private:
+  TSN new_cumulative_tsn_;
+  std::vector<SkippedStream> skipped_streams_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_FORWARD_TSN_COMMON_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc
new file mode 100644
index 0000000000..3cbcd09c75
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.6
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 5    | Chunk  Flags  |    Heartbeat Ack Length       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /            Heartbeat Information TLV (Variable-Length)        /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int HeartbeatAckChunk::kType;
+
+absl::optional<HeartbeatAckChunk> HeartbeatAckChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  absl::optional<Parameters> parameters =
+      Parameters::Parse(reader->variable_data());
+  if (!parameters.has_value()) {
+    return absl::nullopt;
+  }
+  return HeartbeatAckChunk(*std::move(parameters));
+}
+
+void HeartbeatAckChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  rtc::ArrayView<const uint8_t> parameters = parameters_.data();
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, parameters.size());
+  writer.CopyToVariableData(parameters);
+}
+
+std::string HeartbeatAckChunk::ToString() const {
+  return "HEARTBEAT-ACK";
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h
new file mode 100644
index 0000000000..a6479f78b0
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_HEARTBEAT_ACK_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_HEARTBEAT_ACK_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.6
+struct HeartbeatAckChunkConfig : ChunkConfig {
+  static constexpr int kType = 5;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class HeartbeatAckChunk : public Chunk,
+                          public TLVTrait<HeartbeatAckChunkConfig> {
+ public:
+  static constexpr int kType = HeartbeatAckChunkConfig::kType;
+
+  explicit HeartbeatAckChunk(Parameters parameters)
+      : parameters_(std::move(parameters)) {}
+
+  HeartbeatAckChunk(HeartbeatAckChunk&& other) = default;
+  HeartbeatAckChunk& operator=(HeartbeatAckChunk&& other) = default;
+
+  static absl::optional<HeartbeatAckChunk> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  const Parameters& parameters() const { return parameters_; }
+
+  absl::optional<HeartbeatInfoParameter> info() const {
+    return parameters_.get<HeartbeatInfoParameter>();
+  }
+
+ private:
+  Parameters parameters_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_HEARTBEAT_ACK_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_ack_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_ack_chunk_test.cc
new file mode 100644
index 0000000000..e4d0dd1489
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_ack_chunk_test.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(HeartbeatAckChunkTest, FromCapture) {
+  /*
+  HEARTBEAT_ACK chunk (Information: 40 bytes)
+      Chunk type: HEARTBEAT_ACK (5)
+      Chunk flags: 0x00
+      Chunk length: 44
+      Heartbeat info parameter (Information: 36 bytes)
+          Parameter type: Heartbeat info (0x0001)
+          Parameter length: 40
+          Heartbeat information: ad2436603726070000000000000000007b1000000100…
+  */
+
+  uint8_t data[] = {0x05, 0x00, 0x00, 0x2c, 0x00, 0x01, 0x00, 0x28, 0xad,
+                    0x24, 0x36, 0x60, 0x37, 0x26, 0x07, 0x00, 0x00, 0x00,
+                    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x7b, 0x10, 0x00,
+                    0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+                    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(HeartbeatAckChunk chunk,
+                              HeartbeatAckChunk::Parse(data));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(HeartbeatInfoParameter info, chunk.info());
+
+  EXPECT_THAT(
+      info.info(),
+      ElementsAre(0xad, 0x24, 0x36, 0x60, 0x37, 0x26, 0x07, 0x00, 0x00, 0x00,
+                  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x7b, 0x10, 0x00, 0x00,
+                  0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+                  0x00, 0x00, 0x00, 0x00, 0x00, 0x00));
+}
+
+TEST(HeartbeatAckChunkTest, SerializeAndDeserialize) {
+  uint8_t info_data[] = {1, 2, 3, 4};
+  Parameters parameters =
+      Parameters::Builder().Add(HeartbeatInfoParameter(info_data)).Build();
+  HeartbeatAckChunk chunk(std::move(parameters));
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(HeartbeatAckChunk deserialized,
+                              HeartbeatAckChunk::Parse(serialized));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(HeartbeatInfoParameter info, deserialized.info());
+
+  EXPECT_THAT(info.info(), ElementsAre(1, 2, 3, 4));
+
+  EXPECT_EQ(deserialized.ToString(), "HEARTBEAT-ACK");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc
new file mode 100644
index 0000000000..d759d6b16d
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.5
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 4    | Chunk  Flags  |      Heartbeat Length         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /            Heartbeat Information TLV (Variable-Length)        /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int HeartbeatRequestChunk::kType;
+
+absl::optional<HeartbeatRequestChunk> HeartbeatRequestChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  absl::optional<Parameters> parameters =
+      Parameters::Parse(reader->variable_data());
+  if (!parameters.has_value()) {
+    return absl::nullopt;
+  }
+  return HeartbeatRequestChunk(*std::move(parameters));
+}
+
+void HeartbeatRequestChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  rtc::ArrayView<const uint8_t> parameters = parameters_.data();
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, parameters.size());
+  writer.CopyToVariableData(parameters);
+}
+
+std::string HeartbeatRequestChunk::ToString() const {
+  return "HEARTBEAT";
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_request_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_request_chunk.h
new file mode 100644
index 0000000000..fe2ce19504
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_request_chunk.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_HEARTBEAT_REQUEST_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_HEARTBEAT_REQUEST_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+// https://tools.ietf.org/html/rfc4960#section-3.3.5
+struct HeartbeatRequestChunkConfig : ChunkConfig {
+  static constexpr int kType = 4;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class HeartbeatRequestChunk : public Chunk,
+                              public TLVTrait<HeartbeatRequestChunkConfig> {
+ public:
+  static constexpr int kType = HeartbeatRequestChunkConfig::kType;
+
+  explicit HeartbeatRequestChunk(Parameters parameters)
+      : parameters_(std::move(parameters)) {}
+
+  HeartbeatRequestChunk(HeartbeatRequestChunk&& other) = default;
+  HeartbeatRequestChunk& operator=(HeartbeatRequestChunk&& other) = default;
+
+  static absl::optional<HeartbeatRequestChunk> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  const Parameters& parameters() const { return parameters_; }
+  Parameters extract_parameters() && { return std::move(parameters_); }
+  absl::optional<HeartbeatInfoParameter> info() const {
+    return parameters_.get<HeartbeatInfoParameter>();
+  }
+
+ private:
+  Parameters parameters_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_HEARTBEAT_REQUEST_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_request_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_request_chunk_test.cc
new file mode 100644
index 0000000000..94911fe28b
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/heartbeat_request_chunk_test.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(HeartbeatRequestChunkTest, FromCapture) {
+  /*
+  HEARTBEAT chunk (Information: 40 bytes)
+      Chunk type: HEARTBEAT (4)
+      Chunk flags: 0x00
+      Chunk length: 44
+      Heartbeat info parameter (Information: 36 bytes)
+          Parameter type: Heartbeat info (0x0001)
+          Parameter length: 40
+          Heartbeat information: ad2436603726070000000000000000007b10000001…
+  */
+
+  uint8_t data[] = {0x04, 0x00, 0x00, 0x2c, 0x00, 0x01, 0x00, 0x28, 0xad,
+                    0x24, 0x36, 0x60, 0x37, 0x26, 0x07, 0x00, 0x00, 0x00,
+                    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x7b, 0x10, 0x00,
+                    0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+                    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(HeartbeatRequestChunk chunk,
+                              HeartbeatRequestChunk::Parse(data));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(HeartbeatInfoParameter info, chunk.info());
+
+  EXPECT_THAT(
+      info.info(),
+      ElementsAre(0xad, 0x24, 0x36, 0x60, 0x37, 0x26, 0x07, 0x00, 0x00, 0x00,
+                  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x7b, 0x10, 0x00, 0x00,
+                  0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+                  0x00, 0x00, 0x00, 0x00, 0x00, 0x00));
+}
+
+TEST(HeartbeatRequestChunkTest, SerializeAndDeserialize) {
+  uint8_t info_data[] = {1, 2, 3, 4};
+  Parameters parameters =
+      Parameters::Builder().Add(HeartbeatInfoParameter(info_data)).Build();
+  HeartbeatRequestChunk chunk(std::move(parameters));
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(HeartbeatRequestChunk deserialized,
+                              HeartbeatRequestChunk::Parse(serialized));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(HeartbeatInfoParameter info, deserialized.info());
+
+  EXPECT_THAT(info.info(), ElementsAre(1, 2, 3, 4));
+
+  EXPECT_EQ(deserialized.ToString(), "HEARTBEAT");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/idata_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/idata_chunk.cc
new file mode 100644
index 0000000000..378c527909
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/idata_chunk.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/chunk/data_common.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc8260#section-2.1
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 64   |  Res  |I|U|B|E|       Length = Variable       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                              TSN                              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |        Stream Identifier      |           Reserved            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                      Message Identifier                       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |    Payload Protocol Identifier / Fragment Sequence Number     |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /                           User Data                           /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int IDataChunk::kType;
+
+absl::optional<IDataChunk> IDataChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  uint8_t flags = reader->Load8<1>();
+  TSN tsn(reader->Load32<4>());
+  StreamID stream_identifier(reader->Load16<8>());
+  MID message_id(reader->Load32<12>());
+  uint32_t ppid_or_fsn = reader->Load32<16>();
+
+  Options options;
+  options.is_end = Data::IsEnd((flags & (1 << kFlagsBitEnd)) != 0);
+  options.is_beginning =
+      Data::IsBeginning((flags & (1 << kFlagsBitBeginning)) != 0);
+  options.is_unordered = IsUnordered((flags & (1 << kFlagsBitUnordered)) != 0);
+  options.immediate_ack =
+      ImmediateAckFlag((flags & (1 << kFlagsBitImmediateAck)) != 0);
+
+  return IDataChunk(tsn, stream_identifier, message_id,
+                    PPID(options.is_beginning ? ppid_or_fsn : 0),
+                    FSN(options.is_beginning ? 0 : ppid_or_fsn),
+                    std::vector<uint8_t>(reader->variable_data().begin(),
+                                         reader->variable_data().end()),
+                    options);
+}
+
+void IDataChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, payload().size());
+
+  writer.Store8<1>(
+      (*options().is_end ? (1 << kFlagsBitEnd) : 0) |
+      (*options().is_beginning ? (1 << kFlagsBitBeginning) : 0) |
+      (*options().is_unordered ? (1 << kFlagsBitUnordered) : 0) |
+      (*options().immediate_ack ? (1 << kFlagsBitImmediateAck) : 0));
+  writer.Store32<4>(*tsn());
+  writer.Store16<8>(*stream_id());
+  writer.Store32<12>(*message_id());
+  writer.Store32<16>(options().is_beginning ? *ppid() : *fsn());
+  writer.CopyToVariableData(payload());
+}
+
+std::string IDataChunk::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "I-DATA, type=" << (options().is_unordered ? "unordered" : "ordered")
+     << "::"
+     << (*options().is_beginning && *options().is_end
+             ? "complete"
+             : *options().is_beginning ? "first"
+                                       : *options().is_end ? "last" : "middle")
+     << ", tsn=" << *tsn() << ", stream_id=" << *stream_id()
+     << ", message_id=" << *message_id();
+
+  if (*options().is_beginning) {
+    sb << ", ppid=" << *ppid();
+  } else {
+    sb << ", fsn=" << *fsn();
+  }
+  sb << ", length=" << payload().size();
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/idata_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/idata_chunk.h
new file mode 100644
index 0000000000..8cdf2a1fc4
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/idata_chunk.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_IDATA_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_IDATA_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/chunk/data_common.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc8260#section-2.1
+struct IDataChunkConfig : ChunkConfig {
+  static constexpr int kType = 64;
+  static constexpr size_t kHeaderSize = 20;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class IDataChunk : public AnyDataChunk, public TLVTrait<IDataChunkConfig> {
+ public:
+  static constexpr int kType = IDataChunkConfig::kType;
+
+  // Exposed to allow the retransmission queue to make room for the correct
+  // header size.
+  static constexpr size_t kHeaderSize = IDataChunkConfig::kHeaderSize;
+  IDataChunk(TSN tsn,
+             StreamID stream_id,
+             MID message_id,
+             PPID ppid,
+             FSN fsn,
+             std::vector<uint8_t> payload,
+             const Options& options)
+      : AnyDataChunk(tsn,
+                     stream_id,
+                     SSN(0),
+                     message_id,
+                     fsn,
+                     ppid,
+                     std::move(payload),
+                     options) {}
+
+  explicit IDataChunk(TSN tsn, Data&& data, bool immediate_ack)
+      : AnyDataChunk(tsn, std::move(data), immediate_ack) {}
+
+  static absl::optional<IDataChunk> Parse(rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_IDATA_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/idata_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/idata_chunk_test.cc
new file mode 100644
index 0000000000..fea492d71e
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/idata_chunk_test.cc
@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(IDataChunkTest, AtBeginningFromCapture) {
+  /*
+  I_DATA chunk(ordered, first segment, TSN: 2487901653, SID: 1, MID: 0,
+    payload length: 1180 bytes)
+      Chunk type: I_DATA (64)
+      Chunk flags: 0x02
+      Chunk length: 1200
+      Transmission sequence number: 2487901653
+      Stream identifier: 0x0001
+      Reserved: 0
+      Message identifier: 0
+      Payload protocol identifier: WebRTC Binary (53)
+      Reassembled Message in frame: 39
+  */
+
+  uint8_t data[] = {0x40, 0x02, 0x00, 0x15, 0x94, 0x4a, 0x5d, 0xd5,
+                    0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+                    0x00, 0x00, 0x00, 0x35, 0x01, 0x00, 0x00, 0x00};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(IDataChunk chunk, IDataChunk::Parse(data));
+  EXPECT_EQ(*chunk.tsn(), 2487901653);
+  EXPECT_EQ(*chunk.stream_id(), 1);
+  EXPECT_EQ(*chunk.message_id(), 0u);
+  EXPECT_EQ(*chunk.ppid(), 53u);
+  EXPECT_EQ(*chunk.fsn(), 0u);  // Not provided (so set to zero)
+}
+
+TEST(IDataChunkTest, AtBeginningSerializeAndDeserialize) {
+  IDataChunk::Options options;
+  options.is_beginning = Data::IsBeginning(true);
+  IDataChunk chunk(TSN(123), StreamID(456), MID(789), PPID(53), FSN(0), {1},
+                   options);
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(IDataChunk deserialized,
+                              IDataChunk::Parse(serialized));
+  EXPECT_EQ(*deserialized.tsn(), 123u);
+  EXPECT_EQ(*deserialized.stream_id(), 456u);
+  EXPECT_EQ(*deserialized.message_id(), 789u);
+  EXPECT_EQ(*deserialized.ppid(), 53u);
+  EXPECT_EQ(*deserialized.fsn(), 0u);
+
+  EXPECT_EQ(deserialized.ToString(),
+            "I-DATA, type=ordered::first, tsn=123, stream_id=456, "
+            "message_id=789, ppid=53, length=1");
+}
+
+TEST(IDataChunkTest, InMiddleFromCapture) {
+  /*
+  I_DATA chunk(ordered, last segment, TSN: 2487901706, SID: 3, MID: 1,
+    FSN: 8, payload length: 560 bytes)
+      Chunk type: I_DATA (64)
+      Chunk flags: 0x01
+      Chunk length: 580
+      Transmission sequence number: 2487901706
+      Stream identifier: 0x0003
+      Reserved: 0
+      Message identifier: 1
+      Fragment sequence number: 8
+      Reassembled SCTP Fragments (10000 bytes, 9 fragments):
+  */
+
+  uint8_t data[] = {0x40, 0x01, 0x00, 0x15, 0x94, 0x4a, 0x5e, 0x0a,
+                    0x00, 0x03, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
+                    0x00, 0x00, 0x00, 0x08, 0x01, 0x00, 0x00, 0x00};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(IDataChunk chunk, IDataChunk::Parse(data));
+  EXPECT_EQ(*chunk.tsn(), 2487901706);
+  EXPECT_EQ(*chunk.stream_id(), 3u);
+  EXPECT_EQ(*chunk.message_id(), 1u);
+  EXPECT_EQ(*chunk.ppid(), 0u);  // Not provided (so set to zero)
+  EXPECT_EQ(*chunk.fsn(), 8u);
+}
+
+TEST(IDataChunkTest, InMiddleSerializeAndDeserialize) {
+  IDataChunk chunk(TSN(123), StreamID(456), MID(789), PPID(0), FSN(101112),
+                   {1, 2, 3}, /*options=*/{});
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(IDataChunk deserialized,
+                              IDataChunk::Parse(serialized));
+  EXPECT_EQ(*deserialized.tsn(), 123u);
+  EXPECT_EQ(*deserialized.stream_id(), 456u);
+  EXPECT_EQ(*deserialized.message_id(), 789u);
+  EXPECT_EQ(*deserialized.ppid(), 0u);
+  EXPECT_EQ(*deserialized.fsn(), 101112u);
+  EXPECT_THAT(deserialized.payload(), ElementsAre(1, 2, 3));
+
+  EXPECT_EQ(deserialized.ToString(),
+            "I-DATA, type=ordered::middle, tsn=123, stream_id=456, "
+            "message_id=789, fsn=101112, length=3");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc
new file mode 100644
index 0000000000..a647a8bf8a
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc8260#section-2.3.1
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 194  | Flags = 0x00  |      Length = Variable        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                       New Cumulative TSN                      |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |       Stream Identifier       |          Reserved           |U|
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                       Message Identifier                      |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /                                                               /
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |       Stream Identifier       |          Reserved           |U|
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                       Message Identifier                      |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int IForwardTsnChunk::kType;
+
+absl::optional<IForwardTsnChunk> IForwardTsnChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  TSN new_cumulative_tsn(reader->Load32<4>());
+
+  size_t streams_skipped =
+      reader->variable_data_size() / kSkippedStreamBufferSize;
+  std::vector<SkippedStream> skipped_streams;
+  skipped_streams.reserve(streams_skipped);
+  size_t offset = 0;
+  for (size_t i = 0; i < streams_skipped; ++i) {
+    BoundedByteReader<kSkippedStreamBufferSize> sub_reader =
+        reader->sub_reader<kSkippedStreamBufferSize>(offset);
+
+    StreamID stream_id(sub_reader.Load16<0>());
+    IsUnordered unordered(sub_reader.Load8<3>() & 0x01);
+    MID message_id(sub_reader.Load32<4>());
+    skipped_streams.emplace_back(unordered, stream_id, message_id);
+    offset += kSkippedStreamBufferSize;
+  }
+  RTC_DCHECK(offset == reader->variable_data_size());
+  return IForwardTsnChunk(new_cumulative_tsn, std::move(skipped_streams));
+}
+
+void IForwardTsnChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  rtc::ArrayView<const SkippedStream> skipped = skipped_streams();
+  size_t variable_size = skipped.size() * kSkippedStreamBufferSize;
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, variable_size);
+
+  writer.Store32<4>(*new_cumulative_tsn());
+  size_t offset = 0;
+  for (size_t i = 0; i < skipped.size(); ++i) {
+    BoundedByteWriter<kSkippedStreamBufferSize> sub_writer =
+        writer.sub_writer<kSkippedStreamBufferSize>(offset);
+
+    sub_writer.Store16<0>(*skipped[i].stream_id);
+    sub_writer.Store8<3>(skipped[i].unordered ? 1 : 0);
+    sub_writer.Store32<4>(*skipped[i].message_id);
+    offset += kSkippedStreamBufferSize;
+  }
+  RTC_DCHECK(offset == variable_size);
+}
+
+std::string IForwardTsnChunk::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "I-FORWARD-TSN, new_cumulative_tsn=" << *new_cumulative_tsn();
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/iforward_tsn_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/iforward_tsn_chunk.h
new file mode 100644
index 0000000000..54d23f7a83
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/iforward_tsn_chunk.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_IFORWARD_TSN_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_IFORWARD_TSN_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc8260#section-2.3.1
+struct IForwardTsnChunkConfig : ChunkConfig {
+  static constexpr int kType = 194;
+  static constexpr size_t kHeaderSize = 8;
+  static constexpr size_t kVariableLengthAlignment = 8;
+};
+
+class IForwardTsnChunk : public AnyForwardTsnChunk,
+                         public TLVTrait<IForwardTsnChunkConfig> {
+ public:
+  static constexpr int kType = IForwardTsnChunkConfig::kType;
+
+  IForwardTsnChunk(TSN new_cumulative_tsn,
+                   std::vector<SkippedStream> skipped_streams)
+      : AnyForwardTsnChunk(new_cumulative_tsn, std::move(skipped_streams)) {}
+
+  static absl::optional<IForwardTsnChunk> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+ private:
+  static constexpr size_t kSkippedStreamBufferSize = 8;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_IFORWARD_TSN_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/iforward_tsn_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/iforward_tsn_chunk_test.cc
new file mode 100644
index 0000000000..6a89433be1
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/iforward_tsn_chunk_test.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(IForwardTsnChunkTest, FromCapture) {
+  /*
+  I_FORWARD_TSN chunk(Cumulative TSN: 3094631148)
+      Chunk type: I_FORWARD_TSN (194)
+      Chunk flags: 0x00
+      Chunk length: 16
+      New cumulative TSN: 3094631148
+      Stream identifier: 1
+      Flags: 0x0000
+      Message identifier: 2
+  */
+
+  uint8_t data[] = {0xc2, 0x00, 0x00, 0x10, 0xb8, 0x74, 0x52, 0xec,
+                    0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(IForwardTsnChunk chunk,
+                              IForwardTsnChunk::Parse(data));
+  EXPECT_EQ(*chunk.new_cumulative_tsn(), 3094631148u);
+  EXPECT_THAT(chunk.skipped_streams(),
+              ElementsAre(IForwardTsnChunk::SkippedStream(
+                  IsUnordered(false), StreamID(1), MID(2))));
+}
+
+TEST(IForwardTsnChunkTest, SerializeAndDeserialize) {
+  IForwardTsnChunk chunk(
+      TSN(123), {IForwardTsnChunk::SkippedStream(IsUnordered(false),
+                                                 StreamID(1), MID(23)),
+                 IForwardTsnChunk::SkippedStream(IsUnordered(true),
+                                                 StreamID(42), MID(99))});
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(IForwardTsnChunk deserialized,
+                              IForwardTsnChunk::Parse(serialized));
+  EXPECT_EQ(*deserialized.new_cumulative_tsn(), 123u);
+  EXPECT_THAT(deserialized.skipped_streams(),
+              ElementsAre(IForwardTsnChunk::SkippedStream(IsUnordered(false),
+                                                          StreamID(1), MID(23)),
+                          IForwardTsnChunk::SkippedStream(
+                              IsUnordered(true), StreamID(42), MID(99))));
+
+  EXPECT_EQ(deserialized.ToString(), "I-FORWARD-TSN, new_cumulative_tsn=123");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/init_ack_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_ack_chunk.cc
new file mode 100644
index 0000000000..c7ef9da1f1
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_ack_chunk.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/init_ack_chunk.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_format.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.3
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 2    |  Chunk Flags  |      Chunk Length             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                         Initiate Tag                          |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |              Advertised Receiver Window Credit                |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  Number of Outbound Streams   |  Number of Inbound Streams    |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                          Initial TSN                          |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /              Optional/Variable-Length Parameters              /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int InitAckChunk::kType;
+
+absl::optional<InitAckChunk> InitAckChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  VerificationTag initiate_tag(reader->Load32<4>());
+  uint32_t a_rwnd = reader->Load32<8>();
+  uint16_t nbr_outbound_streams = reader->Load16<12>();
+  uint16_t nbr_inbound_streams = reader->Load16<14>();
+  TSN initial_tsn(reader->Load32<16>());
+  absl::optional<Parameters> parameters =
+      Parameters::Parse(reader->variable_data());
+  if (!parameters.has_value()) {
+    return absl::nullopt;
+  }
+  return InitAckChunk(initiate_tag, a_rwnd, nbr_outbound_streams,
+                      nbr_inbound_streams, initial_tsn, *std::move(parameters));
+}
+
+void InitAckChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  rtc::ArrayView<const uint8_t> parameters = parameters_.data();
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, parameters.size());
+
+  writer.Store32<4>(*initiate_tag_);
+  writer.Store32<8>(a_rwnd_);
+  writer.Store16<12>(nbr_outbound_streams_);
+  writer.Store16<14>(nbr_inbound_streams_);
+  writer.Store32<16>(*initial_tsn_);
+  writer.CopyToVariableData(parameters);
+}
+
+std::string InitAckChunk::ToString() const {
+  return rtc::StringFormat("INIT_ACK, initiate_tag=0x%0x, initial_tsn=%u",
+                           *initiate_tag(), *initial_tsn());
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/init_ack_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_ack_chunk.h
new file mode 100644
index 0000000000..6fcf64b2eb
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_ack_chunk.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_INIT_ACK_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_INIT_ACK_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.3
+struct InitAckChunkConfig : ChunkConfig {
+  static constexpr int kType = 2;
+  static constexpr size_t kHeaderSize = 20;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class InitAckChunk : public Chunk, public TLVTrait<InitAckChunkConfig> {
+ public:
+  static constexpr int kType = InitAckChunkConfig::kType;
+
+  InitAckChunk(VerificationTag initiate_tag,
+               uint32_t a_rwnd,
+               uint16_t nbr_outbound_streams,
+               uint16_t nbr_inbound_streams,
+               TSN initial_tsn,
+               Parameters parameters)
+      : initiate_tag_(initiate_tag),
+        a_rwnd_(a_rwnd),
+        nbr_outbound_streams_(nbr_outbound_streams),
+        nbr_inbound_streams_(nbr_inbound_streams),
+        initial_tsn_(initial_tsn),
+        parameters_(std::move(parameters)) {}
+
+  InitAckChunk(InitAckChunk&& other) = default;
+  InitAckChunk& operator=(InitAckChunk&& other) = default;
+
+  static absl::optional<InitAckChunk> Parse(rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  VerificationTag initiate_tag() const { return initiate_tag_; }
+  uint32_t a_rwnd() const { return a_rwnd_; }
+  uint16_t nbr_outbound_streams() const { return nbr_outbound_streams_; }
+  uint16_t nbr_inbound_streams() const { return nbr_inbound_streams_; }
+  TSN initial_tsn() const { return initial_tsn_; }
+  const Parameters& parameters() const { return parameters_; }
+
+ private:
+  VerificationTag initiate_tag_;
+  uint32_t a_rwnd_;
+  uint16_t nbr_outbound_streams_;
+  uint16_t nbr_inbound_streams_;
+  TSN initial_tsn_;
+  Parameters parameters_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_INIT_ACK_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/init_ack_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_ack_chunk_test.cc
new file mode 100644
index 0000000000..184ade747d
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_ack_chunk_test.cc
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/init_ack_chunk.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/parameter/state_cookie_parameter.h"
+#include "net/dcsctp/packet/parameter/supported_extensions_parameter.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(InitAckChunkTest, FromCapture) {
+  /*
+  INIT_ACK chunk (Outbound streams: 1000, inbound streams: 2048)
+      Chunk type: INIT_ACK (2)
+      Chunk flags: 0x00
+      Chunk length: 292
+      Initiate tag: 0x579c2f98
+      Advertised receiver window credit (a_rwnd): 131072
+      Number of outbound streams: 1000
+      Number of inbound streams: 2048
+      Initial TSN: 1670811335
+      Forward TSN supported parameter
+          Parameter type: Forward TSN supported (0xc000)
+          Parameter length: 4
+      Supported Extensions parameter (Supported types: FORWARD_TSN, RE_CONFIG)
+          Parameter type: Supported Extensions (0x8008)
+          Parameter length: 6
+          Supported chunk type: FORWARD_TSN (192)
+          Supported chunk type: RE_CONFIG (130)
+          Parameter padding: 0000
+      State cookie parameter (Cookie length: 256 bytes)
+          Parameter type: State cookie (0x0007)
+          Parameter length: 260
+          State cookie: 4b414d452d42534420312e310000000096b8386000000000…
+  */
+
+  uint8_t data[] = {
+      0x02, 0x00, 0x01, 0x24, 0x57, 0x9c, 0x2f, 0x98, 0x00, 0x02, 0x00, 0x00,
+      0x03, 0xe8, 0x08, 0x00, 0x63, 0x96, 0x8e, 0xc7, 0xc0, 0x00, 0x00, 0x04,
+      0x80, 0x08, 0x00, 0x06, 0xc0, 0x82, 0x00, 0x00, 0x00, 0x07, 0x01, 0x04,
+      0x4b, 0x41, 0x4d, 0x45, 0x2d, 0x42, 0x53, 0x44, 0x20, 0x31, 0x2e, 0x31,
+      0x00, 0x00, 0x00, 0x00, 0x96, 0xb8, 0x38, 0x60, 0x00, 0x00, 0x00, 0x00,
+      0x52, 0x5a, 0x0e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60, 0xea, 0x00, 0x00,
+      0xb5, 0xaa, 0x19, 0xea, 0x31, 0xef, 0xa4, 0x2b, 0x90, 0x16, 0x7a, 0xde,
+      0x57, 0x9c, 0x2f, 0x98, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x13, 0x88, 0x13, 0x88, 0x00, 0x00, 0x01, 0x00, 0x01, 0x01, 0x01, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x5a, 0xde, 0x7a, 0x16, 0x90,
+      0x00, 0x02, 0x00, 0x00, 0x03, 0xe8, 0x03, 0xe8, 0x25, 0x0d, 0x37, 0xe8,
+      0x80, 0x00, 0x00, 0x04, 0xc0, 0x00, 0x00, 0x04, 0x80, 0x08, 0x00, 0x09,
+      0xc0, 0x0f, 0xc1, 0x80, 0x82, 0x00, 0x00, 0x00, 0x80, 0x02, 0x00, 0x24,
+      0xab, 0x31, 0x44, 0x62, 0x12, 0x1a, 0x15, 0x13, 0xfd, 0x5a, 0x5f, 0x69,
+      0xef, 0xaa, 0x06, 0xe9, 0xab, 0xd7, 0x48, 0xcc, 0x3b, 0xd1, 0x4b, 0x60,
+      0xed, 0x7f, 0xa6, 0x44, 0xce, 0x4d, 0xd2, 0xad, 0x80, 0x04, 0x00, 0x06,
+      0x00, 0x01, 0x00, 0x00, 0x80, 0x03, 0x00, 0x06, 0x80, 0xc1, 0x00, 0x00,
+      0x02, 0x00, 0x01, 0x24, 0x57, 0x9c, 0x2f, 0x98, 0x00, 0x02, 0x00, 0x00,
+      0x03, 0xe8, 0x08, 0x00, 0x63, 0x96, 0x8e, 0xc7, 0xc0, 0x00, 0x00, 0x04,
+      0x80, 0x08, 0x00, 0x06, 0xc0, 0x82, 0x00, 0x00, 0x51, 0x95, 0x01, 0x88,
+      0x0d, 0x80, 0x7b, 0x19, 0xe7, 0xf9, 0xc6, 0x18, 0x5c, 0x4a, 0xbf, 0x39,
+      0x32, 0xe5, 0x63, 0x8e};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(InitAckChunk chunk, InitAckChunk::Parse(data));
+
+  EXPECT_EQ(chunk.initiate_tag(), VerificationTag(0x579c2f98u));
+  EXPECT_EQ(chunk.a_rwnd(), 131072u);
+  EXPECT_EQ(chunk.nbr_outbound_streams(), 1000u);
+  EXPECT_EQ(chunk.nbr_inbound_streams(), 2048u);
+  EXPECT_EQ(chunk.initial_tsn(), TSN(1670811335u));
+  EXPECT_TRUE(
+      chunk.parameters().get<ForwardTsnSupportedParameter>().has_value());
+  EXPECT_TRUE(
+      chunk.parameters().get<SupportedExtensionsParameter>().has_value());
+  EXPECT_TRUE(chunk.parameters().get<StateCookieParameter>().has_value());
+}
+
+TEST(InitAckChunkTest, SerializeAndDeserialize) {
+  uint8_t state_cookie[] = {1, 2, 3, 4, 5};
+  Parameters parameters =
+      Parameters::Builder().Add(StateCookieParameter(state_cookie)).Build();
+  InitAckChunk chunk(VerificationTag(123), /*a_rwnd=*/456,
+                     /*nbr_outbound_streams=*/65535,
+                     /*nbr_inbound_streams=*/65534, /*initial_tsn=*/TSN(789),
+                     /*parameters=*/std::move(parameters));
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(InitAckChunk deserialized,
+                              InitAckChunk::Parse(serialized));
+
+  EXPECT_EQ(chunk.initiate_tag(), VerificationTag(123u));
+  EXPECT_EQ(chunk.a_rwnd(), 456u);
+  EXPECT_EQ(chunk.nbr_outbound_streams(), 65535u);
+  EXPECT_EQ(chunk.nbr_inbound_streams(), 65534u);
+  EXPECT_EQ(chunk.initial_tsn(), TSN(789u));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      StateCookieParameter cookie,
+      deserialized.parameters().get<StateCookieParameter>());
+  EXPECT_THAT(cookie.data(), ElementsAre(1, 2, 3, 4, 5));
+  EXPECT_EQ(deserialized.ToString(),
+            "INIT_ACK, initiate_tag=0x7b, initial_tsn=789");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/init_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_chunk.cc
new file mode 100644
index 0000000000..8030107072
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_chunk.cc
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/init_chunk.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_format.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.2
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 1    |  Chunk Flags  |      Chunk Length             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                         Initiate Tag                          |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |           Advertised Receiver Window Credit (a_rwnd)          |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  Number of Outbound Streams   |  Number of Inbound Streams    |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                          Initial TSN                          |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /              Optional/Variable-Length Parameters              /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int InitChunk::kType;
+
+absl::optional<InitChunk> InitChunk::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  VerificationTag initiate_tag(reader->Load32<4>());
+  uint32_t a_rwnd = reader->Load32<8>();
+  uint16_t nbr_outbound_streams = reader->Load16<12>();
+  uint16_t nbr_inbound_streams = reader->Load16<14>();
+  TSN initial_tsn(reader->Load32<16>());
+
+  absl::optional<Parameters> parameters =
+      Parameters::Parse(reader->variable_data());
+  if (!parameters.has_value()) {
+    return absl::nullopt;
+  }
+  return InitChunk(initiate_tag, a_rwnd, nbr_outbound_streams,
+                   nbr_inbound_streams, initial_tsn, *std::move(parameters));
+}
+
+void InitChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  rtc::ArrayView<const uint8_t> parameters = parameters_.data();
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, parameters.size());
+
+  writer.Store32<4>(*initiate_tag_);
+  writer.Store32<8>(a_rwnd_);
+  writer.Store16<12>(nbr_outbound_streams_);
+  writer.Store16<14>(nbr_inbound_streams_);
+  writer.Store32<16>(*initial_tsn_);
+
+  writer.CopyToVariableData(parameters);
+}
+
+std::string InitChunk::ToString() const {
+  return rtc::StringFormat("INIT, initiate_tag=0x%0x, initial_tsn=%u",
+                           *initiate_tag(), *initial_tsn());
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/init_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_chunk.h
new file mode 100644
index 0000000000..38f9994caa
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_chunk.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_INIT_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_INIT_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.2
+struct InitChunkConfig : ChunkConfig {
+  static constexpr int kType = 1;
+  static constexpr size_t kHeaderSize = 20;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class InitChunk : public Chunk, public TLVTrait<InitChunkConfig> {
+ public:
+  static constexpr int kType = InitChunkConfig::kType;
+
+  InitChunk(VerificationTag initiate_tag,
+            uint32_t a_rwnd,
+            uint16_t nbr_outbound_streams,
+            uint16_t nbr_inbound_streams,
+            TSN initial_tsn,
+            Parameters parameters)
+      : initiate_tag_(initiate_tag),
+        a_rwnd_(a_rwnd),
+        nbr_outbound_streams_(nbr_outbound_streams),
+        nbr_inbound_streams_(nbr_inbound_streams),
+        initial_tsn_(initial_tsn),
+        parameters_(std::move(parameters)) {}
+
+  InitChunk(InitChunk&& other) = default;
+  InitChunk& operator=(InitChunk&& other) = default;
+
+  static absl::optional<InitChunk> Parse(rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  VerificationTag initiate_tag() const { return initiate_tag_; }
+  uint32_t a_rwnd() const { return a_rwnd_; }
+  uint16_t nbr_outbound_streams() const { return nbr_outbound_streams_; }
+  uint16_t nbr_inbound_streams() const { return nbr_inbound_streams_; }
+  TSN initial_tsn() const { return initial_tsn_; }
+  const Parameters& parameters() const { return parameters_; }
+
+ private:
+  VerificationTag initiate_tag_;
+  uint32_t a_rwnd_;
+  uint16_t nbr_outbound_streams_;
+  uint16_t nbr_inbound_streams_;
+  TSN initial_tsn_;
+  Parameters parameters_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_INIT_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/init_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_chunk_test.cc
new file mode 100644
index 0000000000..bd36d6fdf8
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/init_chunk_test.cc
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/init_chunk.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/parameter/supported_extensions_parameter.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(InitChunkTest, FromCapture) {
+  /*
+  INIT chunk (Outbound streams: 1000, inbound streams: 1000)
+      Chunk type: INIT (1)
+      Chunk flags: 0x00
+      Chunk length: 90
+      Initiate tag: 0xde7a1690
+      Advertised receiver window credit (a_rwnd): 131072
+      Number of outbound streams: 1000
+      Number of inbound streams: 1000
+      Initial TSN: 621623272
+      ECN parameter
+          Parameter type: ECN (0x8000)
+          Parameter length: 4
+      Forward TSN supported parameter
+          Parameter type: Forward TSN supported (0xc000)
+          Parameter length: 4
+      Supported Extensions parameter (Supported types: FORWARD_TSN, AUTH,
+  ASCONF, ASCONF_ACK, RE_CONFIG) Parameter type: Supported Extensions (0x8008)
+          Parameter length: 9
+          Supported chunk type: FORWARD_TSN (192)
+          Supported chunk type: AUTH (15)
+          Supported chunk type: ASCONF (193)
+          Supported chunk type: ASCONF_ACK (128)
+          Supported chunk type: RE_CONFIG (130)
+          Parameter padding: 000000
+      Random parameter
+          Parameter type: Random (0x8002)
+          Parameter length: 36
+          Random number: ab314462121a1513fd5a5f69efaa06e9abd748cc3bd14b60…
+      Requested HMAC Algorithm parameter (Supported HMACs: SHA-1)
+          Parameter type: Requested HMAC Algorithm (0x8004)
+          Parameter length: 6
+          HMAC identifier: SHA-1 (1)
+          Parameter padding: 0000
+      Authenticated Chunk list parameter (Chunk types to be authenticated:
+  ASCONF_ACK, ASCONF) Parameter type: Authenticated Chunk list (0x8003)
+          Parameter length: 6
+          Chunk type: ASCONF_ACK (128)
+          Chunk type: ASCONF (193)
+  */
+
+  uint8_t data[] = {
+      0x01, 0x00, 0x00, 0x5a, 0xde, 0x7a, 0x16, 0x90, 0x00, 0x02, 0x00, 0x00,
+      0x03, 0xe8, 0x03, 0xe8, 0x25, 0x0d, 0x37, 0xe8, 0x80, 0x00, 0x00, 0x04,
+      0xc0, 0x00, 0x00, 0x04, 0x80, 0x08, 0x00, 0x09, 0xc0, 0x0f, 0xc1, 0x80,
+      0x82, 0x00, 0x00, 0x00, 0x80, 0x02, 0x00, 0x24, 0xab, 0x31, 0x44, 0x62,
+      0x12, 0x1a, 0x15, 0x13, 0xfd, 0x5a, 0x5f, 0x69, 0xef, 0xaa, 0x06, 0xe9,
+      0xab, 0xd7, 0x48, 0xcc, 0x3b, 0xd1, 0x4b, 0x60, 0xed, 0x7f, 0xa6, 0x44,
+      0xce, 0x4d, 0xd2, 0xad, 0x80, 0x04, 0x00, 0x06, 0x00, 0x01, 0x00, 0x00,
+      0x80, 0x03, 0x00, 0x06, 0x80, 0xc1, 0x00, 0x00};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(InitChunk chunk, InitChunk::Parse(data));
+
+  EXPECT_EQ(chunk.initiate_tag(), VerificationTag(0xde7a1690));
+  EXPECT_EQ(chunk.a_rwnd(), 131072u);
+  EXPECT_EQ(chunk.nbr_outbound_streams(), 1000u);
+  EXPECT_EQ(chunk.nbr_inbound_streams(), 1000u);
+  EXPECT_EQ(chunk.initial_tsn(), TSN(621623272u));
+  EXPECT_TRUE(
+      chunk.parameters().get<ForwardTsnSupportedParameter>().has_value());
+  EXPECT_TRUE(
+      chunk.parameters().get<SupportedExtensionsParameter>().has_value());
+}
+
+TEST(InitChunkTest, SerializeAndDeserialize) {
+  InitChunk chunk(VerificationTag(123), /*a_rwnd=*/456,
+                  /*nbr_outbound_streams=*/65535,
+                  /*nbr_inbound_streams=*/65534, /*initial_tsn=*/TSN(789),
+                  /*parameters=*/Parameters::Builder().Build());
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(InitChunk deserialized,
+                              InitChunk::Parse(serialized));
+
+  EXPECT_EQ(deserialized.initiate_tag(), VerificationTag(123u));
+  EXPECT_EQ(deserialized.a_rwnd(), 456u);
+  EXPECT_EQ(deserialized.nbr_outbound_streams(), 65535u);
+  EXPECT_EQ(deserialized.nbr_inbound_streams(), 65534u);
+  EXPECT_EQ(deserialized.initial_tsn(), TSN(789u));
+  EXPECT_EQ(deserialized.ToString(),
+            "INIT, initiate_tag=0x7b, initial_tsn=789");
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/reconfig_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/reconfig_chunk.cc
new file mode 100644
index 0000000000..f39f3b619f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/reconfig_chunk.cc
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-3.1
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  | Type = 130    |  Chunk Flags  |      Chunk Length             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /                  Re-configuration Parameter                   /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /             Re-configuration Parameter (optional)             /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int ReConfigChunk::kType;
+
+absl::optional<ReConfigChunk> ReConfigChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  absl::optional<Parameters> parameters =
+      Parameters::Parse(reader->variable_data());
+  if (!parameters.has_value()) {
+    return absl::nullopt;
+  }
+
+  return ReConfigChunk(*std::move(parameters));
+}
+
+void ReConfigChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  rtc::ArrayView<const uint8_t> parameters = parameters_.data();
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, parameters.size());
+  writer.CopyToVariableData(parameters);
+}
+
+std::string ReConfigChunk::ToString() const {
+  return "RE-CONFIG";
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/reconfig_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/reconfig_chunk.h
new file mode 100644
index 0000000000..9d2539a515
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/reconfig_chunk.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_RECONFIG_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_RECONFIG_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-3.1
+struct ReConfigChunkConfig : ChunkConfig {
+  static constexpr int kType = 130;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class ReConfigChunk : public Chunk, public TLVTrait<ReConfigChunkConfig> {
+ public:
+  static constexpr int kType = ReConfigChunkConfig::kType;
+
+  explicit ReConfigChunk(Parameters parameters)
+      : parameters_(std::move(parameters)) {}
+
+  static absl::optional<ReConfigChunk> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  const Parameters& parameters() const { return parameters_; }
+  Parameters extract_parameters() { return std::move(parameters_); }
+
+ private:
+  Parameters parameters_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_RECONFIG_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/reconfig_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/reconfig_chunk_test.cc
new file mode 100644
index 0000000000..dbf40ff8c0
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/reconfig_chunk_test.cc
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+using ::testing::SizeIs;
+
+TEST(ReConfigChunkTest, FromCapture) {
+  /*
+  RE_CONFIG chunk
+      Chunk type: RE_CONFIG (130)
+      Chunk flags: 0x00
+      Chunk length: 22
+      Outgoing SSN reset request parameter
+          Parameter type: Outgoing SSN reset request (0x000d)
+          Parameter length: 18
+          Re-configuration request sequence number: 2270550051
+          Re-configuration response sequence number: 1905748638
+          Senders last assigned TSN: 2270550066
+          Stream Identifier: 6
+      Chunk padding: 0000
+  */
+
+  uint8_t data[] = {0x82, 0x00, 0x00, 0x16, 0x00, 0x0d, 0x00, 0x12,
+                    0x87, 0x55, 0xd8, 0x23, 0x71, 0x97, 0x6a, 0x9e,
+                    0x87, 0x55, 0xd8, 0x32, 0x00, 0x06, 0x00, 0x00};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(ReConfigChunk chunk, ReConfigChunk::Parse(data));
+
+  const Parameters& parameters = chunk.parameters();
+  EXPECT_THAT(parameters.descriptors(), SizeIs(1));
+  ParameterDescriptor desc = parameters.descriptors()[0];
+  ASSERT_EQ(desc.type, OutgoingSSNResetRequestParameter::kType);
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req,
+      OutgoingSSNResetRequestParameter::Parse(desc.data));
+  EXPECT_EQ(*req.request_sequence_number(), 2270550051u);
+  EXPECT_EQ(*req.response_sequence_number(), 1905748638u);
+  EXPECT_EQ(*req.sender_last_assigned_tsn(), 2270550066u);
+  EXPECT_THAT(req.stream_ids(), ElementsAre(StreamID(6)));
+}
+
+TEST(ReConfigChunkTest, SerializeAndDeserialize) {
+  Parameters::Builder params_builder =
+      Parameters::Builder().Add(OutgoingSSNResetRequestParameter(
+          ReconfigRequestSN(123), ReconfigRequestSN(456), TSN(789),
+          {StreamID(42), StreamID(43)}));
+
+  ReConfigChunk chunk(params_builder.Build());
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(ReConfigChunk deserialized,
+                              ReConfigChunk::Parse(serialized));
+
+  const Parameters& parameters = deserialized.parameters();
+  EXPECT_THAT(parameters.descriptors(), SizeIs(1));
+  ParameterDescriptor desc = parameters.descriptors()[0];
+  ASSERT_EQ(desc.type, OutgoingSSNResetRequestParameter::kType);
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req,
+      OutgoingSSNResetRequestParameter::Parse(desc.data));
+  EXPECT_EQ(*req.request_sequence_number(), 123u);
+  EXPECT_EQ(*req.response_sequence_number(), 456u);
+  EXPECT_EQ(*req.sender_last_assigned_tsn(), 789u);
+  EXPECT_THAT(req.stream_ids(), ElementsAre(StreamID(42), StreamID(43)));
+
+  EXPECT_EQ(deserialized.ToString(), "RE-CONFIG");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/sack_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/sack_chunk.cc
new file mode 100644
index 0000000000..d80e430082
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/sack_chunk.cc
@@ -0,0 +1,155 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/str_join.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.4
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 3    |Chunk  Flags   |      Chunk Length             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                      Cumulative TSN Ack                       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |          Advertised Receiver Window Credit (a_rwnd)           |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  | Number of Gap Ack Blocks = N  |  Number of Duplicate TSNs = X |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  Gap Ack Block #1 Start       |   Gap Ack Block #1 End        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                                                               /
+//  \                              ...                              \
+//  /                                                               /
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Gap Ack Block #N Start      |  Gap Ack Block #N End         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                       Duplicate TSN 1                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                                                               /
+//  \                              ...                              \
+//  /                                                               /
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                       Duplicate TSN X                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int SackChunk::kType;
+
+absl::optional<SackChunk> SackChunk::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  TSN tsn_ack(reader->Load32<4>());
+  uint32_t a_rwnd = reader->Load32<8>();
+  uint16_t nbr_of_gap_blocks = reader->Load16<12>();
+  uint16_t nbr_of_dup_tsns = reader->Load16<14>();
+
+  if (reader->variable_data_size() != nbr_of_gap_blocks * kGapAckBlockSize +
+                                          nbr_of_dup_tsns * kDupTsnBlockSize) {
+    RTC_DLOG(LS_WARNING) << "Invalid number of gap blocks or duplicate TSNs";
+    return absl::nullopt;
+  }
+
+  std::vector<GapAckBlock> gap_ack_blocks;
+  gap_ack_blocks.reserve(nbr_of_gap_blocks);
+  size_t offset = 0;
+  for (int i = 0; i < nbr_of_gap_blocks; ++i) {
+    BoundedByteReader<kGapAckBlockSize> sub_reader =
+        reader->sub_reader<kGapAckBlockSize>(offset);
+
+    uint16_t start = sub_reader.Load16<0>();
+    uint16_t end = sub_reader.Load16<2>();
+    gap_ack_blocks.emplace_back(start, end);
+    offset += kGapAckBlockSize;
+  }
+
+  std::set<TSN> duplicate_tsns;
+  for (int i = 0; i < nbr_of_dup_tsns; ++i) {
+    BoundedByteReader<kDupTsnBlockSize> sub_reader =
+        reader->sub_reader<kDupTsnBlockSize>(offset);
+
+    duplicate_tsns.insert(TSN(sub_reader.Load32<0>()));
+    offset += kDupTsnBlockSize;
+  }
+  RTC_DCHECK(offset == reader->variable_data_size());
+
+  return SackChunk(tsn_ack, a_rwnd, gap_ack_blocks, duplicate_tsns);
+}
+
+void SackChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  int nbr_of_gap_blocks = gap_ack_blocks_.size();
+  int nbr_of_dup_tsns = duplicate_tsns_.size();
+  size_t variable_size =
+      nbr_of_gap_blocks * kGapAckBlockSize + nbr_of_dup_tsns * kDupTsnBlockSize;
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, variable_size);
+
+  writer.Store32<4>(*cumulative_tsn_ack_);
+  writer.Store32<8>(a_rwnd_);
+  writer.Store16<12>(nbr_of_gap_blocks);
+  writer.Store16<14>(nbr_of_dup_tsns);
+
+  size_t offset = 0;
+  for (int i = 0; i < nbr_of_gap_blocks; ++i) {
+    BoundedByteWriter<kGapAckBlockSize> sub_writer =
+        writer.sub_writer<kGapAckBlockSize>(offset);
+
+    sub_writer.Store16<0>(gap_ack_blocks_[i].start);
+    sub_writer.Store16<2>(gap_ack_blocks_[i].end);
+    offset += kGapAckBlockSize;
+  }
+
+  for (TSN tsn : duplicate_tsns_) {
+    BoundedByteWriter<kDupTsnBlockSize> sub_writer =
+        writer.sub_writer<kDupTsnBlockSize>(offset);
+
+    sub_writer.Store32<0>(*tsn);
+    offset += kDupTsnBlockSize;
+  }
+
+  RTC_DCHECK(offset == variable_size);
+}
+
+std::string SackChunk::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "SACK, cum_ack_tsn=" << *cumulative_tsn_ack()
+     << ", a_rwnd=" << a_rwnd();
+  for (const GapAckBlock& gap : gap_ack_blocks_) {
+    uint32_t first = *cumulative_tsn_ack_ + gap.start;
+    uint32_t last = *cumulative_tsn_ack_ + gap.end;
+    sb << ", gap=" << first << "--" << last;
+  }
+  if (!duplicate_tsns_.empty()) {
+    sb << ", dup_tsns="
+       << StrJoin(duplicate_tsns(), ",",
+                  [](rtc::StringBuilder& sb, TSN tsn) { sb << *tsn; });
+  }
+
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/sack_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/sack_chunk.h
new file mode 100644
index 0000000000..e6758fa332
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/sack_chunk.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_SACK_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_SACK_CHUNK_H_
+#include <stddef.h>
+
+#include <cstdint>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.4
+struct SackChunkConfig : ChunkConfig {
+  static constexpr int kType = 3;
+  static constexpr size_t kHeaderSize = 16;
+  static constexpr size_t kVariableLengthAlignment = 4;
+};
+
+class SackChunk : public Chunk, public TLVTrait<SackChunkConfig> {
+ public:
+  static constexpr int kType = SackChunkConfig::kType;
+
+  struct GapAckBlock {
+    GapAckBlock(uint16_t start, uint16_t end) : start(start), end(end) {}
+
+    uint16_t start;
+    uint16_t end;
+
+    bool operator==(const GapAckBlock& other) const {
+      return start == other.start && end == other.end;
+    }
+  };
+
+  SackChunk(TSN cumulative_tsn_ack,
+            uint32_t a_rwnd,
+            std::vector<GapAckBlock> gap_ack_blocks,
+            std::set<TSN> duplicate_tsns)
+      : cumulative_tsn_ack_(cumulative_tsn_ack),
+        a_rwnd_(a_rwnd),
+        gap_ack_blocks_(std::move(gap_ack_blocks)),
+        duplicate_tsns_(std::move(duplicate_tsns)) {}
+  static absl::optional<SackChunk> Parse(rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  TSN cumulative_tsn_ack() const { return cumulative_tsn_ack_; }
+  uint32_t a_rwnd() const { return a_rwnd_; }
+  rtc::ArrayView<const GapAckBlock> gap_ack_blocks() const {
+    return gap_ack_blocks_;
+  }
+  const std::set<TSN>& duplicate_tsns() const { return duplicate_tsns_; }
+
+ private:
+  static constexpr size_t kGapAckBlockSize = 4;
+  static constexpr size_t kDupTsnBlockSize = 4;
+
+  const TSN cumulative_tsn_ack_;
+  const uint32_t a_rwnd_;
+  std::vector<GapAckBlock> gap_ack_blocks_;
+  std::set<TSN> duplicate_tsns_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_SACK_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/sack_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/sack_chunk_test.cc
new file mode 100644
index 0000000000..9122945308
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/sack_chunk_test.cc
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(SackChunkTest, FromCapture) {
+  /*
+  SACK chunk (Cumulative TSN: 916312075, a_rwnd: 126323,
+    gaps: 2, duplicate TSNs: 1)
+      Chunk type: SACK (3)
+      Chunk flags: 0x00
+      Chunk length: 28
+      Cumulative TSN ACK: 916312075
+      Advertised receiver window credit (a_rwnd): 126323
+      Number of gap acknowledgement blocks: 2
+      Number of duplicated TSNs: 1
+      Gap Acknowledgement for TSN 916312077 to 916312081
+      Gap Acknowledgement for TSN 916312083 to 916312083
+      [Number of TSNs in gap acknowledgement blocks: 6]
+      Duplicate TSN: 916312081
+
+  */
+
+  uint8_t data[] = {0x03, 0x00, 0x00, 0x1c, 0x36, 0x9d, 0xd0, 0x0b, 0x00, 0x01,
+                    0xed, 0x73, 0x00, 0x02, 0x00, 0x01, 0x00, 0x02, 0x00, 0x06,
+                    0x00, 0x08, 0x00, 0x08, 0x36, 0x9d, 0xd0, 0x11};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SackChunk chunk, SackChunk::Parse(data));
+
+  TSN cum_ack_tsn(916312075);
+  EXPECT_EQ(chunk.cumulative_tsn_ack(), cum_ack_tsn);
+  EXPECT_EQ(chunk.a_rwnd(), 126323u);
+  EXPECT_THAT(
+      chunk.gap_ack_blocks(),
+      ElementsAre(SackChunk::GapAckBlock(
+                      static_cast<uint16_t>(916312077 - *cum_ack_tsn),
+                      static_cast<uint16_t>(916312081 - *cum_ack_tsn)),
+                  SackChunk::GapAckBlock(
+                      static_cast<uint16_t>(916312083 - *cum_ack_tsn),
+                      static_cast<uint16_t>(916312083 - *cum_ack_tsn))));
+  EXPECT_THAT(chunk.duplicate_tsns(), ElementsAre(TSN(916312081)));
+}
+
+TEST(SackChunkTest, SerializeAndDeserialize) {
+  SackChunk chunk(TSN(123), /*a_rwnd=*/456, {SackChunk::GapAckBlock(2, 3)},
+                  {TSN(1), TSN(2), TSN(3)});
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SackChunk deserialized,
+                              SackChunk::Parse(serialized));
+
+  EXPECT_EQ(*deserialized.cumulative_tsn_ack(), 123u);
+  EXPECT_EQ(deserialized.a_rwnd(), 456u);
+  EXPECT_THAT(deserialized.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(2, 3)));
+  EXPECT_THAT(deserialized.duplicate_tsns(),
+              ElementsAre(TSN(1), TSN(2), TSN(3)));
+
+  EXPECT_EQ(deserialized.ToString(),
+            "SACK, cum_ack_tsn=123, a_rwnd=456, gap=125--126, dup_tsns=1,2,3");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc
new file mode 100644
index 0000000000..d42aceead4
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/shutdown_ack_chunk.h"
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.9
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 8    |Chunk  Flags   |      Length = 4               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int ShutdownAckChunk::kType;
+
+absl::optional<ShutdownAckChunk> ShutdownAckChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  if (!ParseTLV(data).has_value()) {
+    return absl::nullopt;
+  }
+  return ShutdownAckChunk();
+}
+
+void ShutdownAckChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  AllocateTLV(out);
+}
+
+std::string ShutdownAckChunk::ToString() const {
+  return "SHUTDOWN-ACK";
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_ack_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_ack_chunk.h
new file mode 100644
index 0000000000..29c1a98be6
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_ack_chunk.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_SHUTDOWN_ACK_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_SHUTDOWN_ACK_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.9
+struct ShutdownAckChunkConfig : ChunkConfig {
+  static constexpr int kType = 8;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class ShutdownAckChunk : public Chunk, public TLVTrait<ShutdownAckChunkConfig> {
+ public:
+  static constexpr int kType = ShutdownAckChunkConfig::kType;
+
+  ShutdownAckChunk() {}
+
+  static absl::optional<ShutdownAckChunk> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_SHUTDOWN_ACK_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_ack_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_ack_chunk_test.cc
new file mode 100644
index 0000000000..ef04ea9892
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_ack_chunk_test.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/shutdown_ack_chunk.h"
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(ShutdownAckChunkTest, FromCapture) {
+  /*
+  SHUTDOWN_ACK chunk
+      Chunk type: SHUTDOWN_ACK (8)
+      Chunk flags: 0x00
+      Chunk length: 4
+  */
+
+  uint8_t data[] = {0x08, 0x00, 0x00, 0x04};
+
+  EXPECT_TRUE(ShutdownAckChunk::Parse(data).has_value());
+}
+
+TEST(ShutdownAckChunkTest, SerializeAndDeserialize) {
+  ShutdownAckChunk chunk;
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  EXPECT_TRUE(ShutdownAckChunk::Parse(serialized).has_value());
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_chunk.cc
new file mode 100644
index 0000000000..59f806f7f7
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_chunk.cc
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/shutdown_chunk.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.8
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 7    | Chunk  Flags  |      Length = 8               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                      Cumulative TSN Ack                       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int ShutdownChunk::kType;
+
+absl::optional<ShutdownChunk> ShutdownChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  TSN cumulative_tsn_ack(reader->Load32<4>());
+  return ShutdownChunk(cumulative_tsn_ack);
+}
+
+void ShutdownChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out);
+  writer.Store32<4>(*cumulative_tsn_ack_);
+}
+
+std::string ShutdownChunk::ToString() const {
+  return "SHUTDOWN";
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_chunk.h
new file mode 100644
index 0000000000..8148cca286
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_chunk.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_SHUTDOWN_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_SHUTDOWN_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.8
+struct ShutdownChunkConfig : ChunkConfig {
+  static constexpr int kType = 7;
+  static constexpr size_t kHeaderSize = 8;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class ShutdownChunk : public Chunk, public TLVTrait<ShutdownChunkConfig> {
+ public:
+  static constexpr int kType = ShutdownChunkConfig::kType;
+
+  explicit ShutdownChunk(TSN cumulative_tsn_ack)
+      : cumulative_tsn_ack_(cumulative_tsn_ack) {}
+
+  static absl::optional<ShutdownChunk> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  TSN cumulative_tsn_ack() const { return cumulative_tsn_ack_; }
+
+ private:
+  TSN cumulative_tsn_ack_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_SHUTDOWN_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_chunk_test.cc
new file mode 100644
index 0000000000..16d147ca83
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_chunk_test.cc
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/shutdown_chunk.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+
+namespace dcsctp {
+namespace {
+TEST(ShutdownChunkTest, FromCapture) {
+  /*
+  SHUTDOWN chunk (Cumulative TSN ack: 101831101)
+      Chunk type: SHUTDOWN (7)
+      Chunk flags: 0x00
+      Chunk length: 8
+      Cumulative TSN Ack: 101831101
+  */
+
+  uint8_t data[] = {0x07, 0x00, 0x00, 0x08, 0x06, 0x11, 0xd1, 0xbd};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(ShutdownChunk chunk, ShutdownChunk::Parse(data));
+  EXPECT_EQ(chunk.cumulative_tsn_ack(), TSN(101831101u));
+}
+
+TEST(ShutdownChunkTest, SerializeAndDeserialize) {
+  ShutdownChunk chunk(TSN(12345678));
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(ShutdownChunk deserialized,
+                              ShutdownChunk::Parse(serialized));
+
+  EXPECT_EQ(deserialized.cumulative_tsn_ack(), TSN(12345678u));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc
new file mode 100644
index 0000000000..3f54857437
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/shutdown_complete_chunk.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.13
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 14   |Reserved     |T|      Length = 4               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int ShutdownCompleteChunk::kType;
+
+absl::optional<ShutdownCompleteChunk> ShutdownCompleteChunk::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  uint8_t flags = reader->Load8<1>();
+  bool tag_reflected = (flags & (1 << kFlagsBitT)) != 0;
+  return ShutdownCompleteChunk(tag_reflected);
+}
+
+void ShutdownCompleteChunk::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out);
+  writer.Store8<1>(tag_reflected_ ? (1 << kFlagsBitT) : 0);
+}
+
+std::string ShutdownCompleteChunk::ToString() const {
+  return "SHUTDOWN-COMPLETE";
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_complete_chunk.h b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_complete_chunk.h
new file mode 100644
index 0000000000..46d28e88dc
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_complete_chunk.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_SHUTDOWN_COMPLETE_CHUNK_H_
+#define NET_DCSCTP_PACKET_CHUNK_SHUTDOWN_COMPLETE_CHUNK_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.13
+struct ShutdownCompleteChunkConfig : ChunkConfig {
+  static constexpr int kType = 14;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class ShutdownCompleteChunk : public Chunk,
+                              public TLVTrait<ShutdownCompleteChunkConfig> {
+ public:
+  static constexpr int kType = ShutdownCompleteChunkConfig::kType;
+
+  explicit ShutdownCompleteChunk(bool tag_reflected)
+      : tag_reflected_(tag_reflected) {}
+
+  static absl::optional<ShutdownCompleteChunk> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+  bool tag_reflected() const { return tag_reflected_; }
+
+ private:
+  static constexpr int kFlagsBitT = 0;
+  bool tag_reflected_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_SHUTDOWN_COMPLETE_CHUNK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_complete_chunk_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_complete_chunk_test.cc
new file mode 100644
index 0000000000..253900d5cd
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk/shutdown_complete_chunk_test.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/shutdown_complete_chunk.h"
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(ShutdownCompleteChunkTest, FromCapture) {
+  /*
+  SHUTDOWN_COMPLETE chunk
+      Chunk type: SHUTDOWN_COMPLETE (14)
+      Chunk flags: 0x00
+      Chunk length: 4
+  */
+
+  uint8_t data[] = {0x0e, 0x00, 0x00, 0x04};
+
+  EXPECT_TRUE(ShutdownCompleteChunk::Parse(data).has_value());
+}
+
+TEST(ShutdownCompleteChunkTest, SerializeAndDeserialize) {
+  ShutdownCompleteChunk chunk(/*tag_reflected=*/false);
+
+  std::vector<uint8_t> serialized;
+  chunk.SerializeTo(serialized);
+
+  EXPECT_TRUE(ShutdownCompleteChunk::Parse(serialized).has_value());
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk_validators.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk_validators.cc
new file mode 100644
index 0000000000..48d351827e
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk_validators.cc
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk_validators.h"
+
+#include <algorithm>
+#include <utility>
+#include <vector>
+
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+
+SackChunk ChunkValidators::Clean(SackChunk&& sack) {
+  if (Validate(sack)) {
+    return std::move(sack);
+  }
+
+  RTC_DLOG(LS_WARNING) << "Received SACK is malformed; cleaning it";
+
+  std::vector<SackChunk::GapAckBlock> gap_ack_blocks;
+  gap_ack_blocks.reserve(sack.gap_ack_blocks().size());
+
+  // First: Only keep blocks that are sane
+  for (const SackChunk::GapAckBlock& gap_ack_block : sack.gap_ack_blocks()) {
+    if (gap_ack_block.end > gap_ack_block.start) {
+      gap_ack_blocks.emplace_back(gap_ack_block);
+    }
+  }
+
+  // Not more than at most one remaining? Exit early.
+  if (gap_ack_blocks.size() <= 1) {
+    return SackChunk(sack.cumulative_tsn_ack(), sack.a_rwnd(),
+                     std::move(gap_ack_blocks), sack.duplicate_tsns());
+  }
+
+  // Sort the intervals by their start value, to aid in the merging below.
+  absl::c_sort(gap_ack_blocks, [&](const SackChunk::GapAckBlock& a,
+                                   const SackChunk::GapAckBlock& b) {
+    return a.start < b.start;
+  });
+
+  // Merge overlapping ranges.
+  std::vector<SackChunk::GapAckBlock> merged;
+  merged.reserve(gap_ack_blocks.size());
+  merged.push_back(gap_ack_blocks[0]);
+
+  for (size_t i = 1; i < gap_ack_blocks.size(); ++i) {
+    if (merged.back().end + 1 >= gap_ack_blocks[i].start) {
+      merged.back().end = std::max(merged.back().end, gap_ack_blocks[i].end);
+    } else {
+      merged.push_back(gap_ack_blocks[i]);
+    }
+  }
+
+  return SackChunk(sack.cumulative_tsn_ack(), sack.a_rwnd(), std::move(merged),
+                   sack.duplicate_tsns());
+}
+
+bool ChunkValidators::Validate(const SackChunk& sack) {
+  if (sack.gap_ack_blocks().empty()) {
+    return true;
+  }
+
+  // Ensure that gap-ack-blocks are sorted, has an "end" that is not before
+  // "start" and are non-overlapping and non-adjacent.
+  uint16_t prev_end = 0;
+  for (const SackChunk::GapAckBlock& gap_ack_block : sack.gap_ack_blocks()) {
+    if (gap_ack_block.end < gap_ack_block.start) {
+      return false;
+    }
+    if (gap_ack_block.start <= (prev_end + 1)) {
+      return false;
+    }
+    prev_end = gap_ack_block.end;
+  }
+  return true;
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk_validators.h b/third_party/libwebrtc/net/dcsctp/packet/chunk_validators.h
new file mode 100644
index 0000000000..b11848a162
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk_validators.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CHUNK_VALIDATORS_H_
+#define NET_DCSCTP_PACKET_CHUNK_VALIDATORS_H_
+
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+
+namespace dcsctp {
+// Validates and cleans SCTP chunks.
+class ChunkValidators {
+ public:
+  // Given a SackChunk, will return `true` if it's valid, and `false` if not.
+  static bool Validate(const SackChunk& sack);
+
+  // Given a SackChunk, it will return a cleaned and validated variant of it.
+  // RFC4960 doesn't say anything about validity of SACKs or if the Gap ACK
+  // blocks must be sorted, and non-overlapping. While they always are in
+  // well-behaving implementations, this can't be relied on.
+  //
+  // This method internally calls `Validate`, which means that you can always
+  // pass a SackChunk to this method (valid or not), and use the results.
+  static SackChunk Clean(SackChunk&& sack);
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CHUNK_VALIDATORS_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/chunk_validators_test.cc b/third_party/libwebrtc/net/dcsctp/packet/chunk_validators_test.cc
new file mode 100644
index 0000000000..d59fd4ec48
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/chunk_validators_test.cc
@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk_validators.h"
+
+#include <utility>
+
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+
+TEST(ChunkValidatorsTest, NoGapAckBlocksAreValid) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456,
+                 /*gap_ack_blocks=*/{}, {});
+
+  EXPECT_TRUE(ChunkValidators::Validate(sack));
+
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+  EXPECT_THAT(clean.gap_ack_blocks(), IsEmpty());
+}
+
+TEST(ChunkValidatorsTest, OneValidAckBlock) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456, {SackChunk::GapAckBlock(2, 3)}, {});
+
+  EXPECT_TRUE(ChunkValidators::Validate(sack));
+
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+  EXPECT_THAT(clean.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(2, 3)));
+}
+
+TEST(ChunkValidatorsTest, TwoValidAckBlocks) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456,
+                 {SackChunk::GapAckBlock(2, 3), SackChunk::GapAckBlock(5, 6)},
+                 {});
+
+  EXPECT_TRUE(ChunkValidators::Validate(sack));
+
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+  EXPECT_THAT(
+      clean.gap_ack_blocks(),
+      ElementsAre(SackChunk::GapAckBlock(2, 3), SackChunk::GapAckBlock(5, 6)));
+}
+
+TEST(ChunkValidatorsTest, OneInvalidAckBlock) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456, {SackChunk::GapAckBlock(1, 2)}, {});
+
+  EXPECT_FALSE(ChunkValidators::Validate(sack));
+
+  // It's not strictly valid, but due to the renegable nature of gap ack blocks,
+  // the cum_ack_tsn can't simply be moved.
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+  EXPECT_THAT(clean.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(1, 2)));
+}
+
+TEST(ChunkValidatorsTest, RemovesInvalidGapAckBlockFromSack) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456,
+                 {SackChunk::GapAckBlock(2, 3), SackChunk::GapAckBlock(6, 4)},
+                 {});
+
+  EXPECT_FALSE(ChunkValidators::Validate(sack));
+
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+
+  EXPECT_THAT(clean.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(2, 3)));
+}
+
+TEST(ChunkValidatorsTest, SortsGapAckBlocksInOrder) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456,
+                 {SackChunk::GapAckBlock(6, 7), SackChunk::GapAckBlock(3, 4)},
+                 {});
+
+  EXPECT_FALSE(ChunkValidators::Validate(sack));
+
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+
+  EXPECT_THAT(
+      clean.gap_ack_blocks(),
+      ElementsAre(SackChunk::GapAckBlock(3, 4), SackChunk::GapAckBlock(6, 7)));
+}
+
+TEST(ChunkValidatorsTest, MergesAdjacentBlocks) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456,
+                 {SackChunk::GapAckBlock(3, 4), SackChunk::GapAckBlock(5, 6)},
+                 {});
+
+  EXPECT_FALSE(ChunkValidators::Validate(sack));
+
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+
+  EXPECT_THAT(clean.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(3, 6)));
+}
+
+TEST(ChunkValidatorsTest, MergesOverlappingByOne) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456,
+                 {SackChunk::GapAckBlock(3, 4), SackChunk::GapAckBlock(4, 5)},
+                 {});
+
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+
+  EXPECT_FALSE(ChunkValidators::Validate(sack));
+
+  EXPECT_THAT(clean.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(3, 5)));
+}
+
+TEST(ChunkValidatorsTest, MergesOverlappingByMore) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456,
+                 {SackChunk::GapAckBlock(3, 10), SackChunk::GapAckBlock(4, 5)},
+                 {});
+
+  EXPECT_FALSE(ChunkValidators::Validate(sack));
+
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+
+  EXPECT_THAT(clean.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(3, 10)));
+}
+
+TEST(ChunkValidatorsTest, MergesBlocksStartingWithSameStartOffset) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456,
+                 {SackChunk::GapAckBlock(3, 7), SackChunk::GapAckBlock(3, 5),
+                  SackChunk::GapAckBlock(3, 9)},
+                 {});
+
+  EXPECT_FALSE(ChunkValidators::Validate(sack));
+
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+
+  EXPECT_THAT(clean.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(3, 9)));
+}
+
+TEST(ChunkValidatorsTest, MergesBlocksPartiallyOverlapping) {
+  SackChunk sack(TSN(123), /*a_rwnd=*/456,
+                 {SackChunk::GapAckBlock(3, 7), SackChunk::GapAckBlock(5, 9)},
+                 {});
+
+  EXPECT_FALSE(ChunkValidators::Validate(sack));
+
+  SackChunk clean = ChunkValidators::Clean(std::move(sack));
+
+  EXPECT_THAT(clean.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(3, 9)));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/crc32c.cc b/third_party/libwebrtc/net/dcsctp/packet/crc32c.cc
new file mode 100644
index 0000000000..e3f0dc1d19
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/crc32c.cc
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/crc32c.h"
+
+#include <cstdint>
+
+#include "third_party/crc32c/src/include/crc32c/crc32c.h"
+
+namespace dcsctp {
+
+uint32_t GenerateCrc32C(rtc::ArrayView<const uint8_t> data) {
+  uint32_t crc32c = crc32c_value(data.data(), data.size());
+
+  // Byte swapping for little endian byte order:
+  uint8_t byte0 = crc32c;
+  uint8_t byte1 = crc32c >> 8;
+  uint8_t byte2 = crc32c >> 16;
+  uint8_t byte3 = crc32c >> 24;
+  crc32c = ((byte0 << 24) | (byte1 << 16) | (byte2 << 8) | byte3);
+  return crc32c;
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/crc32c.h b/third_party/libwebrtc/net/dcsctp/packet/crc32c.h
new file mode 100644
index 0000000000..a969e1b26b
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/crc32c.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_CRC32C_H_
+#define NET_DCSCTP_PACKET_CRC32C_H_
+
+#include <cstdint>
+
+#include "api/array_view.h"
+
+namespace dcsctp {
+
+// Generates the CRC32C checksum of `data`.
+uint32_t GenerateCrc32C(rtc::ArrayView<const uint8_t> data);
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_CRC32C_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/crc32c_test.cc b/third_party/libwebrtc/net/dcsctp/packet/crc32c_test.cc
new file mode 100644
index 0000000000..0821c4ef75
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/crc32c_test.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/crc32c.h"
+
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+constexpr std::array<const uint8_t, 0> kEmpty = {};
+constexpr std::array<const uint8_t, 1> kZero = {0};
+constexpr std::array<const uint8_t, 4> kManyZeros = {0, 0, 0, 0};
+constexpr std::array<const uint8_t, 4> kShort = {1, 2, 3, 4};
+constexpr std::array<const uint8_t, 8> kLong = {1, 2, 3, 4, 5, 6, 7, 8};
+// https://tools.ietf.org/html/rfc3720#appendix-B.4
+constexpr std::array<const uint8_t, 32> k32Zeros = {
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0};
+constexpr std::array<const uint8_t, 32> k32Ones = {
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF};
+constexpr std::array<const uint8_t, 32> k32Incrementing = {
+    0,  1,  2,  3,  4,  5,  6,  7,  8,  9,  10, 11, 12, 13, 14, 15,
+    16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31};
+constexpr std::array<const uint8_t, 32> k32Decrementing = {
+    31, 30, 29, 28, 27, 26, 25, 24, 23, 22, 21, 20, 19, 18, 17, 16,
+    15, 14, 13, 12, 11, 10, 9,  8,  7,  6,  5,  4,  3,  2,  1,  0};
+constexpr std::array<const uint8_t, 48> kISCSICommandPDU = {
+    0x01, 0xc0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x14, 0x00, 0x00, 0x00, 0x00, 0x00, 0x04, 0x00,
+    0x00, 0x00, 0x00, 0x14, 0x00, 0x00, 0x00, 0x18, 0x28, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+};
+
+TEST(Crc32Test, TestVectors) {
+  EXPECT_EQ(GenerateCrc32C(kEmpty), 0U);
+  EXPECT_EQ(GenerateCrc32C(kZero), 0x51537d52U);
+  EXPECT_EQ(GenerateCrc32C(kManyZeros), 0xc74b6748U);
+  EXPECT_EQ(GenerateCrc32C(kShort), 0xf48c3029U);
+  EXPECT_EQ(GenerateCrc32C(kLong), 0x811f8946U);
+  // https://tools.ietf.org/html/rfc3720#appendix-B.4
+  EXPECT_EQ(GenerateCrc32C(k32Zeros), 0xaa36918aU);
+  EXPECT_EQ(GenerateCrc32C(k32Ones), 0x43aba862U);
+  EXPECT_EQ(GenerateCrc32C(k32Incrementing), 0x4e79dd46U);
+  EXPECT_EQ(GenerateCrc32C(k32Decrementing), 0x5cdb3f11U);
+  EXPECT_EQ(GenerateCrc32C(kISCSICommandPDU), 0x563a96d9U);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/data.h b/third_party/libwebrtc/net/dcsctp/packet/data.h
new file mode 100644
index 0000000000..c1754ed59a
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/data.h
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_DATA_H_
+#define NET_DCSCTP_PACKET_DATA_H_
+
+#include <cstdint>
+#include <utility>
+#include <vector>
+
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// Represents data that is either received and extracted from a DATA/I-DATA
+// chunk, or data that is supposed to be sent, and wrapped in a DATA/I-DATA
+// chunk (depending on peer capabilities).
+//
+// The data wrapped in this structure is actually the same as the DATA/I-DATA
+// chunk (actually the union of them), but to avoid having all components be
+// aware of the implementation details of the different chunks, this abstraction
+// is used instead. A notable difference is also that it doesn't carry a
+// Transmission Sequence Number (TSN), as that is not known when a chunk is
+// created (assigned late, just when sending), and that the TSNs in DATA/I-DATA
+// are wrapped numbers, and within the library, unwrapped sequence numbers are
+// preferably used.
+struct Data {
+  // Indicates if a chunk is the first in a fragmented message and maps to the
+  // "beginning" flag in DATA/I-DATA chunk.
+  using IsBeginning = webrtc::StrongAlias<class IsBeginningTag, bool>;
+
+  // Indicates if a chunk is the last in a fragmented message  and maps to the
+  // "end" flag in DATA/I-DATA chunk.
+  using IsEnd = webrtc::StrongAlias<class IsEndTag, bool>;
+
+  Data(StreamID stream_id,
+       SSN ssn,
+       MID message_id,
+       FSN fsn,
+       PPID ppid,
+       std::vector<uint8_t> payload,
+       IsBeginning is_beginning,
+       IsEnd is_end,
+       IsUnordered is_unordered)
+      : stream_id(stream_id),
+        ssn(ssn),
+        message_id(message_id),
+        fsn(fsn),
+        ppid(ppid),
+        payload(std::move(payload)),
+        is_beginning(is_beginning),
+        is_end(is_end),
+        is_unordered(is_unordered) {}
+
+  // Move-only, to avoid accidental copies.
+  Data(Data&& other) = default;
+  Data& operator=(Data&& other) = default;
+
+  // Creates a copy of this `Data` object.
+  Data Clone() const {
+    return Data(stream_id, ssn, message_id, fsn, ppid, payload, is_beginning,
+                is_end, is_unordered);
+  }
+
+  // The size of this data, which translates to the size of its payload.
+  size_t size() const { return payload.size(); }
+
+  // Stream Identifier.
+  StreamID stream_id;
+
+  // Stream Sequence Number (SSN), per stream, for ordered chunks. Defined by
+  // RFC4960 and used only in DATA chunks (not I-DATA).
+  SSN ssn;
+
+  // Message Identifier (MID) per stream and ordered/unordered. Defined by
+  // RFC8260, and used together with options.is_unordered and stream_id to
+  // uniquely identify a message. Used only in I-DATA chunks (not DATA).
+  MID message_id;
+  // Fragment Sequence Number (FSN) per stream and ordered/unordered, as above.
+  FSN fsn;
+
+  // Payload Protocol Identifier (PPID).
+  PPID ppid;
+
+  // The actual data payload.
+  std::vector<uint8_t> payload;
+
+  // If this data represents the first, last or a middle chunk.
+  IsBeginning is_beginning;
+  IsEnd is_end;
+  // If this data is sent/received unordered.
+  IsUnordered is_unordered;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_DATA_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc
new file mode 100644
index 0000000000..ef67c2a49f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h"
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.10
+
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Cause Code=10              |      Cause Length=4          |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int CookieReceivedWhileShuttingDownCause::kType;
+
+absl::optional<CookieReceivedWhileShuttingDownCause>
+CookieReceivedWhileShuttingDownCause::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  if (!ParseTLV(data).has_value()) {
+    return absl::nullopt;
+  }
+  return CookieReceivedWhileShuttingDownCause();
+}
+
+void CookieReceivedWhileShuttingDownCause::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  AllocateTLV(out);
+}
+
+std::string CookieReceivedWhileShuttingDownCause::ToString() const {
+  return "Cookie Received While Shutting Down";
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h
new file mode 100644
index 0000000000..362f181fba
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_COOKIE_RECEIVED_WHILE_SHUTTING_DOWN_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_COOKIE_RECEIVED_WHILE_SHUTTING_DOWN_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.10
+struct CookieReceivedWhileShuttingDownCauseConfig : public ParameterConfig {
+  static constexpr int kType = 10;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class CookieReceivedWhileShuttingDownCause
+    : public Parameter,
+      public TLVTrait<CookieReceivedWhileShuttingDownCauseConfig> {
+ public:
+  static constexpr int kType =
+      CookieReceivedWhileShuttingDownCauseConfig::kType;
+
+  CookieReceivedWhileShuttingDownCause() {}
+
+  static absl::optional<CookieReceivedWhileShuttingDownCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_COOKIE_RECEIVED_WHILE_SHUTTING_DOWN_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause_test.cc
new file mode 100644
index 0000000000..afb8364c32
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause_test.cc
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(CookieReceivedWhileShuttingDownCauseTest, SerializeAndDeserialize) {
+  CookieReceivedWhileShuttingDownCause parameter;
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      CookieReceivedWhileShuttingDownCause deserialized,
+      CookieReceivedWhileShuttingDownCause::Parse(serialized));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/error_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/error_cause.cc
new file mode 100644
index 0000000000..dcd07472ed
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/error_cause.cc
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/math.h"
+#include "net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h"
+#include "net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h"
+#include "net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h"
+#include "net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h"
+#include "net/dcsctp/packet/error_cause/no_user_data_cause.h"
+#include "net/dcsctp/packet/error_cause/out_of_resource_error_cause.h"
+#include "net/dcsctp/packet/error_cause/protocol_violation_cause.h"
+#include "net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h"
+#include "net/dcsctp/packet/error_cause/stale_cookie_error_cause.h"
+#include "net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h"
+#include "net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h"
+#include "net/dcsctp/packet/error_cause/unresolvable_address_cause.h"
+#include "net/dcsctp/packet/error_cause/user_initiated_abort_cause.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+template <class ErrorCause>
+bool ParseAndPrint(ParameterDescriptor descriptor, rtc::StringBuilder& sb) {
+  if (descriptor.type == ErrorCause::kType) {
+    absl::optional<ErrorCause> p = ErrorCause::Parse(descriptor.data);
+    if (p.has_value()) {
+      sb << p->ToString();
+    } else {
+      sb << "Failed to parse error cause of type " << ErrorCause::kType;
+    }
+    return true;
+  }
+  return false;
+}
+
+std::string ErrorCausesToString(const Parameters& parameters) {
+  rtc::StringBuilder sb;
+
+  std::vector<ParameterDescriptor> descriptors = parameters.descriptors();
+  for (size_t i = 0; i < descriptors.size(); ++i) {
+    if (i > 0) {
+      sb << "\n";
+    }
+
+    const ParameterDescriptor& d = descriptors[i];
+    if (!ParseAndPrint<InvalidStreamIdentifierCause>(d, sb) &&
+        !ParseAndPrint<MissingMandatoryParameterCause>(d, sb) &&
+        !ParseAndPrint<StaleCookieErrorCause>(d, sb) &&
+        !ParseAndPrint<OutOfResourceErrorCause>(d, sb) &&
+        !ParseAndPrint<UnresolvableAddressCause>(d, sb) &&
+        !ParseAndPrint<UnrecognizedChunkTypeCause>(d, sb) &&
+        !ParseAndPrint<InvalidMandatoryParameterCause>(d, sb) &&
+        !ParseAndPrint<UnrecognizedParametersCause>(d, sb) &&
+        !ParseAndPrint<NoUserDataCause>(d, sb) &&
+        !ParseAndPrint<CookieReceivedWhileShuttingDownCause>(d, sb) &&
+        !ParseAndPrint<RestartOfAnAssociationWithNewAddressesCause>(d, sb) &&
+        !ParseAndPrint<UserInitiatedAbortCause>(d, sb) &&
+        !ParseAndPrint<ProtocolViolationCause>(d, sb)) {
+      sb << "Unhandled parameter of type: " << d.type;
+    }
+  }
+
+  return sb.Release();
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/error_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/error_cause.h
new file mode 100644
index 0000000000..fa2bf81478
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/error_cause.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_ERROR_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_ERROR_CAUSE_H_
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <iosfwd>
+#include <memory>
+#include <string>
+#include <type_traits>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// Converts the Error Causes in `parameters` to a human readable string,
+// to be used in error reporting and logging.
+std::string ErrorCausesToString(const Parameters& parameters);
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_ERROR_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc
new file mode 100644
index 0000000000..0187544226
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h"
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.7
+
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Cause Code=7              |      Cause Length=4           |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int InvalidMandatoryParameterCause::kType;
+
+absl::optional<InvalidMandatoryParameterCause>
+InvalidMandatoryParameterCause::Parse(rtc::ArrayView<const uint8_t> data) {
+  if (!ParseTLV(data).has_value()) {
+    return absl::nullopt;
+  }
+  return InvalidMandatoryParameterCause();
+}
+
+void InvalidMandatoryParameterCause::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  AllocateTLV(out);
+}
+
+std::string InvalidMandatoryParameterCause::ToString() const {
+  return "Invalid Mandatory Parameter";
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h
new file mode 100644
index 0000000000..e192b5a42f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_INVALID_MANDATORY_PARAMETER_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_INVALID_MANDATORY_PARAMETER_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.7
+struct InvalidMandatoryParameterCauseConfig : public ParameterConfig {
+  static constexpr int kType = 7;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class InvalidMandatoryParameterCause
+    : public Parameter,
+      public TLVTrait<InvalidMandatoryParameterCauseConfig> {
+ public:
+  static constexpr int kType = InvalidMandatoryParameterCauseConfig::kType;
+
+  InvalidMandatoryParameterCause() {}
+
+  static absl::optional<InvalidMandatoryParameterCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_INVALID_MANDATORY_PARAMETER_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause_test.cc
new file mode 100644
index 0000000000..3d532d09b1
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause_test.cc
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(InvalidMandatoryParameterCauseTest, SerializeAndDeserialize) {
+  InvalidMandatoryParameterCause parameter;
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      InvalidMandatoryParameterCause deserialized,
+      InvalidMandatoryParameterCause::Parse(serialized));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc
new file mode 100644
index 0000000000..b2ddd6f4ef
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.1
+
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Cause Code=1              |      Cause Length=8           |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |        Stream Identifier      |         (Reserved)            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int InvalidStreamIdentifierCause::kType;
+
+absl::optional<InvalidStreamIdentifierCause>
+InvalidStreamIdentifierCause::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  StreamID stream_id(reader->Load16<4>());
+  return InvalidStreamIdentifierCause(stream_id);
+}
+
+void InvalidStreamIdentifierCause::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out);
+
+  writer.Store16<4>(*stream_id_);
+}
+
+std::string InvalidStreamIdentifierCause::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Invalid Stream Identifier, stream_id=" << *stream_id_;
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h
new file mode 100644
index 0000000000..b7dfe177b8
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_INVALID_STREAM_IDENTIFIER_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_INVALID_STREAM_IDENTIFIER_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.1
+struct InvalidStreamIdentifierCauseConfig : public ParameterConfig {
+  static constexpr int kType = 1;
+  static constexpr size_t kHeaderSize = 8;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class InvalidStreamIdentifierCause
+    : public Parameter,
+      public TLVTrait<InvalidStreamIdentifierCauseConfig> {
+ public:
+  static constexpr int kType = InvalidStreamIdentifierCauseConfig::kType;
+
+  explicit InvalidStreamIdentifierCause(StreamID stream_id)
+      : stream_id_(stream_id) {}
+
+  static absl::optional<InvalidStreamIdentifierCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  StreamID stream_id() const { return stream_id_; }
+
+ private:
+  StreamID stream_id_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_INVALID_STREAM_IDENTIFIER_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause_test.cc
new file mode 100644
index 0000000000..a282ce5ee8
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause_test.cc
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(InvalidStreamIdentifierCauseTest, SerializeAndDeserialize) {
+  InvalidStreamIdentifierCause parameter(StreamID(1));
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(InvalidStreamIdentifierCause deserialized,
+                              InvalidStreamIdentifierCause::Parse(serialized));
+
+  EXPECT_EQ(*deserialized.stream_id(), 1);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc
new file mode 100644
index 0000000000..b89f86e43e
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/str_join.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.2
+
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Cause Code=2              |      Cause Length=8+N*2       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                   Number of missing params=N                  |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Missing Param Type #1       |   Missing Param Type #2       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Missing Param Type #N-1     |   Missing Param Type #N       |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int MissingMandatoryParameterCause::kType;
+
+absl::optional<MissingMandatoryParameterCause>
+MissingMandatoryParameterCause::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  uint32_t count = reader->Load32<4>();
+  if (reader->variable_data_size() / kMissingParameterSize != count) {
+    RTC_DLOG(LS_WARNING) << "Invalid number of missing parameters";
+    return absl::nullopt;
+  }
+
+  std::vector<uint16_t> missing_parameter_types;
+  missing_parameter_types.reserve(count);
+  for (uint32_t i = 0; i < count; ++i) {
+    BoundedByteReader<kMissingParameterSize> sub_reader =
+        reader->sub_reader<kMissingParameterSize>(i * kMissingParameterSize);
+
+    missing_parameter_types.push_back(sub_reader.Load16<0>());
+  }
+  return MissingMandatoryParameterCause(missing_parameter_types);
+}
+
+void MissingMandatoryParameterCause::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  size_t variable_size =
+      missing_parameter_types_.size() * kMissingParameterSize;
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, variable_size);
+
+  writer.Store32<4>(missing_parameter_types_.size());
+
+  for (size_t i = 0; i < missing_parameter_types_.size(); ++i) {
+    BoundedByteWriter<kMissingParameterSize> sub_writer =
+        writer.sub_writer<kMissingParameterSize>(i * kMissingParameterSize);
+
+    sub_writer.Store16<0>(missing_parameter_types_[i]);
+  }
+}
+
+std::string MissingMandatoryParameterCause::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Missing Mandatory Parameter, missing_parameter_types="
+     << StrJoin(missing_parameter_types_, ",");
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h
new file mode 100644
index 0000000000..4435424295
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_MISSING_MANDATORY_PARAMETER_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_MISSING_MANDATORY_PARAMETER_CAUSE_H_
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.2
+struct MissingMandatoryParameterCauseConfig : public ParameterConfig {
+  static constexpr int kType = 2;
+  static constexpr size_t kHeaderSize = 8;
+  static constexpr size_t kVariableLengthAlignment = 2;
+};
+
+class MissingMandatoryParameterCause
+    : public Parameter,
+      public TLVTrait<MissingMandatoryParameterCauseConfig> {
+ public:
+  static constexpr int kType = MissingMandatoryParameterCauseConfig::kType;
+
+  explicit MissingMandatoryParameterCause(
+      rtc::ArrayView<const uint16_t> missing_parameter_types)
+      : missing_parameter_types_(missing_parameter_types.begin(),
+                                 missing_parameter_types.end()) {}
+
+  static absl::optional<MissingMandatoryParameterCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  rtc::ArrayView<const uint16_t> missing_parameter_types() const {
+    return missing_parameter_types_;
+  }
+
+ private:
+  static constexpr size_t kMissingParameterSize = 2;
+  std::vector<uint16_t> missing_parameter_types_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_MISSING_MANDATORY_PARAMETER_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause_test.cc
new file mode 100644
index 0000000000..1c526ff0e2
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause_test.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+
+TEST(MissingMandatoryParameterCauseTest, SerializeAndDeserialize) {
+  uint16_t parameter_types[] = {1, 2, 3};
+  MissingMandatoryParameterCause parameter(parameter_types);
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      MissingMandatoryParameterCause deserialized,
+      MissingMandatoryParameterCause::Parse(serialized));
+
+  EXPECT_THAT(deserialized.missing_parameter_types(), ElementsAre(1, 2, 3));
+}
+
+TEST(MissingMandatoryParameterCauseTest, HandlesDeserializeZeroParameters) {
+  uint8_t serialized[] = {0, 2, 0, 8, 0, 0, 0, 0};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      MissingMandatoryParameterCause deserialized,
+      MissingMandatoryParameterCause::Parse(serialized));
+
+  EXPECT_THAT(deserialized.missing_parameter_types(), IsEmpty());
+}
+
+TEST(MissingMandatoryParameterCauseTest, HandlesOverflowParameterCount) {
+  // 0x80000004 * 2 = 2**32 + 8 -> if overflow, would validate correctly.
+  uint8_t serialized[] = {0, 2, 0, 8, 0x80, 0x00, 0x00, 0x04};
+
+  EXPECT_FALSE(MissingMandatoryParameterCause::Parse(serialized).has_value());
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/no_user_data_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/no_user_data_cause.cc
new file mode 100644
index 0000000000..2853915b0c
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/no_user_data_cause.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/no_user_data_cause.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.9
+
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Cause Code=9              |      Cause Length=8           |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                  TSN value                                    /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int NoUserDataCause::kType;
+
+absl::optional<NoUserDataCause> NoUserDataCause::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  TSN tsn(reader->Load32<4>());
+  return NoUserDataCause(tsn);
+}
+
+void NoUserDataCause::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out);
+  writer.Store32<4>(*tsn_);
+}
+
+std::string NoUserDataCause::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "No User Data, tsn=" << *tsn_;
+  return sb.Release();
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/no_user_data_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/no_user_data_cause.h
new file mode 100644
index 0000000000..1087dcc97c
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/no_user_data_cause.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_NO_USER_DATA_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_NO_USER_DATA_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.9
+struct NoUserDataCauseConfig : public ParameterConfig {
+  static constexpr int kType = 9;
+  static constexpr size_t kHeaderSize = 8;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class NoUserDataCause : public Parameter,
+                        public TLVTrait<NoUserDataCauseConfig> {
+ public:
+  static constexpr int kType = NoUserDataCauseConfig::kType;
+
+  explicit NoUserDataCause(TSN tsn) : tsn_(tsn) {}
+
+  static absl::optional<NoUserDataCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  TSN tsn() const { return tsn_; }
+
+ private:
+  TSN tsn_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_NO_USER_DATA_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/no_user_data_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/no_user_data_cause_test.cc
new file mode 100644
index 0000000000..0a535bf4fa
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/no_user_data_cause_test.cc
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/no_user_data_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(NoUserDataCauseTest, SerializeAndDeserialize) {
+  NoUserDataCause parameter(TSN(123));
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(NoUserDataCause deserialized,
+                              NoUserDataCause::Parse(serialized));
+
+  EXPECT_EQ(*deserialized.tsn(), 123u);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc
new file mode 100644
index 0000000000..e5c7c0e787
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/out_of_resource_error_cause.h"
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.4
+
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Cause Code=4              |      Cause Length=4           |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int OutOfResourceErrorCause::kType;
+
+absl::optional<OutOfResourceErrorCause> OutOfResourceErrorCause::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  if (!ParseTLV(data).has_value()) {
+    return absl::nullopt;
+  }
+  return OutOfResourceErrorCause();
+}
+
+void OutOfResourceErrorCause::SerializeTo(std::vector<uint8_t>& out) const {
+  AllocateTLV(out);
+}
+
+std::string OutOfResourceErrorCause::ToString() const {
+  return "Out Of Resource";
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h
new file mode 100644
index 0000000000..fc798ca4ac
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_OUT_OF_RESOURCE_ERROR_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_OUT_OF_RESOURCE_ERROR_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.4
+struct OutOfResourceParameterConfig : public ParameterConfig {
+  static constexpr int kType = 4;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class OutOfResourceErrorCause : public Parameter,
+                                public TLVTrait<OutOfResourceParameterConfig> {
+ public:
+  static constexpr int kType = OutOfResourceParameterConfig::kType;
+
+  OutOfResourceErrorCause() {}
+
+  static absl::optional<OutOfResourceErrorCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_OUT_OF_RESOURCE_ERROR_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/out_of_resource_error_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/out_of_resource_error_cause_test.cc
new file mode 100644
index 0000000000..501fc201cd
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/out_of_resource_error_cause_test.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/out_of_resource_error_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(OutOfResourceErrorCauseTest, SerializeAndDeserialize) {
+  OutOfResourceErrorCause parameter;
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(OutOfResourceErrorCause deserialized,
+                              OutOfResourceErrorCause::Parse(serialized));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/protocol_violation_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/protocol_violation_cause.cc
new file mode 100644
index 0000000000..1b8d423afb
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/protocol_violation_cause.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/protocol_violation_cause.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.13
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |         Cause Code=13         |      Cause Length=Variable    |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                    Additional Information                     /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int ProtocolViolationCause::kType;
+
+absl::optional<ProtocolViolationCause> ProtocolViolationCause::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  return ProtocolViolationCause(
+      std::string(reinterpret_cast<const char*>(reader->variable_data().data()),
+                  reader->variable_data().size()));
+}
+
+void ProtocolViolationCause::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer =
+      AllocateTLV(out, additional_information_.size());
+  writer.CopyToVariableData(rtc::MakeArrayView(
+      reinterpret_cast<const uint8_t*>(additional_information_.data()),
+      additional_information_.size()));
+}
+
+std::string ProtocolViolationCause::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Protocol Violation, additional_information="
+     << additional_information_;
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/protocol_violation_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/protocol_violation_cause.h
new file mode 100644
index 0000000000..3081e1f28c
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/protocol_violation_cause.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_PROTOCOL_VIOLATION_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_PROTOCOL_VIOLATION_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.13
+struct ProtocolViolationCauseConfig : public ParameterConfig {
+  static constexpr int kType = 13;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class ProtocolViolationCause : public Parameter,
+                               public TLVTrait<ProtocolViolationCauseConfig> {
+ public:
+  static constexpr int kType = ProtocolViolationCauseConfig::kType;
+
+  explicit ProtocolViolationCause(absl::string_view additional_information)
+      : additional_information_(additional_information) {}
+
+  static absl::optional<ProtocolViolationCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  absl::string_view additional_information() const {
+    return additional_information_;
+  }
+
+ private:
+  std::string additional_information_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_PROTOCOL_VIOLATION_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/protocol_violation_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/protocol_violation_cause_test.cc
new file mode 100644
index 0000000000..902d867091
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/protocol_violation_cause_test.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/protocol_violation_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::SizeIs;
+
+TEST(ProtocolViolationCauseTest, EmptyReason) {
+  Parameters causes =
+      Parameters::Builder().Add(ProtocolViolationCause("")).Build();
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(Parameters deserialized,
+                              Parameters::Parse(causes.data()));
+  ASSERT_THAT(deserialized.descriptors(), SizeIs(1));
+  EXPECT_EQ(deserialized.descriptors()[0].type, ProtocolViolationCause::kType);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      ProtocolViolationCause cause,
+      ProtocolViolationCause::Parse(deserialized.descriptors()[0].data));
+
+  EXPECT_EQ(cause.additional_information(), "");
+}
+
+TEST(ProtocolViolationCauseTest, SetReason) {
+  Parameters causes = Parameters::Builder()
+                          .Add(ProtocolViolationCause("Reason goes here"))
+                          .Build();
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(Parameters deserialized,
+                              Parameters::Parse(causes.data()));
+  ASSERT_THAT(deserialized.descriptors(), SizeIs(1));
+  EXPECT_EQ(deserialized.descriptors()[0].type, ProtocolViolationCause::kType);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      ProtocolViolationCause cause,
+      ProtocolViolationCause::Parse(deserialized.descriptors()[0].data));
+
+  EXPECT_EQ(cause.additional_information(), "Reason goes here");
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc
new file mode 100644
index 0000000000..abe5de6211
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.11
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |         Cause Code=11         |      Cause Length=Variable    |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                       New Address TLVs                        /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int RestartOfAnAssociationWithNewAddressesCause::kType;
+
+absl::optional<RestartOfAnAssociationWithNewAddressesCause>
+RestartOfAnAssociationWithNewAddressesCause::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  return RestartOfAnAssociationWithNewAddressesCause(reader->variable_data());
+}
+
+void RestartOfAnAssociationWithNewAddressesCause::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer =
+      AllocateTLV(out, new_address_tlvs_.size());
+  writer.CopyToVariableData(new_address_tlvs_);
+}
+
+std::string RestartOfAnAssociationWithNewAddressesCause::ToString() const {
+  return "Restart of an Association with New Addresses";
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h
new file mode 100644
index 0000000000..a1cccdc8a1
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_RESTART_OF_AN_ASSOCIATION_WITH_NEW_ADDRESS_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_RESTART_OF_AN_ASSOCIATION_WITH_NEW_ADDRESS_CAUSE_H_
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.11
+struct RestartOfAnAssociationWithNewAddressesCauseConfig
+    : public ParameterConfig {
+  static constexpr int kType = 11;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class RestartOfAnAssociationWithNewAddressesCause
+    : public Parameter,
+      public TLVTrait<RestartOfAnAssociationWithNewAddressesCauseConfig> {
+ public:
+  static constexpr int kType =
+      RestartOfAnAssociationWithNewAddressesCauseConfig::kType;
+
+  explicit RestartOfAnAssociationWithNewAddressesCause(
+      rtc::ArrayView<const uint8_t> new_address_tlvs)
+      : new_address_tlvs_(new_address_tlvs.begin(), new_address_tlvs.end()) {}
+
+  static absl::optional<RestartOfAnAssociationWithNewAddressesCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  rtc::ArrayView<const uint8_t> new_address_tlvs() const {
+    return new_address_tlvs_;
+  }
+
+ private:
+  std::vector<uint8_t> new_address_tlvs_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_RESTART_OF_AN_ASSOCIATION_WITH_NEW_ADDRESS_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause_test.cc
new file mode 100644
index 0000000000..b8ab8b6803
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause_test.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(RestartOfAnAssociationWithNewAddressesCauseTest, SerializeAndDeserialize) {
+  uint8_t data[] = {1, 2, 3};
+  RestartOfAnAssociationWithNewAddressesCause parameter(data);
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      RestartOfAnAssociationWithNewAddressesCause deserialized,
+      RestartOfAnAssociationWithNewAddressesCause::Parse(serialized));
+
+  EXPECT_THAT(deserialized.new_address_tlvs(), ElementsAre(1, 2, 3));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc
new file mode 100644
index 0000000000..d77d8488f1
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/stale_cookie_error_cause.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.3
+
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Cause Code=3              |       Cause Length=8          |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                 Measure of Staleness (usec.)                  |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int StaleCookieErrorCause::kType;
+
+absl::optional<StaleCookieErrorCause> StaleCookieErrorCause::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  uint32_t staleness_us = reader->Load32<4>();
+  return StaleCookieErrorCause(staleness_us);
+}
+
+void StaleCookieErrorCause::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out);
+  writer.Store32<4>(staleness_us_);
+}
+
+std::string StaleCookieErrorCause::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Stale Cookie Error, staleness_us=" << staleness_us_;
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h
new file mode 100644
index 0000000000..d8b7b5b5bd
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_STALE_COOKIE_ERROR_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_STALE_COOKIE_ERROR_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.3
+struct StaleCookieParameterConfig : public ParameterConfig {
+  static constexpr int kType = 3;
+  static constexpr size_t kHeaderSize = 8;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class StaleCookieErrorCause : public Parameter,
+                              public TLVTrait<StaleCookieParameterConfig> {
+ public:
+  static constexpr int kType = StaleCookieParameterConfig::kType;
+
+  explicit StaleCookieErrorCause(uint32_t staleness_us)
+      : staleness_us_(staleness_us) {}
+
+  static absl::optional<StaleCookieErrorCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  uint16_t staleness_us() const { return staleness_us_; }
+
+ private:
+  uint32_t staleness_us_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_STALE_COOKIE_ERROR_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/stale_cookie_error_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/stale_cookie_error_cause_test.cc
new file mode 100644
index 0000000000..c0d1ac1c58
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/stale_cookie_error_cause_test.cc
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/stale_cookie_error_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(StaleCookieErrorCauseTest, SerializeAndDeserialize) {
+  StaleCookieErrorCause parameter(123);
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(StaleCookieErrorCause deserialized,
+                              StaleCookieErrorCause::Parse(serialized));
+
+  EXPECT_EQ(deserialized.staleness_us(), 123);
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc
new file mode 100644
index 0000000000..04b960d992
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h"
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.6
+
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Cause Code=6              |      Cause Length             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                  Unrecognized Chunk                           /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int UnrecognizedChunkTypeCause::kType;
+
+absl::optional<UnrecognizedChunkTypeCause> UnrecognizedChunkTypeCause::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  std::vector<uint8_t> unrecognized_chunk(reader->variable_data().begin(),
+                                          reader->variable_data().end());
+  return UnrecognizedChunkTypeCause(std::move(unrecognized_chunk));
+}
+
+void UnrecognizedChunkTypeCause::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer =
+      AllocateTLV(out, unrecognized_chunk_.size());
+  writer.CopyToVariableData(unrecognized_chunk_);
+}
+
+std::string UnrecognizedChunkTypeCause::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Unrecognized Chunk Type, chunk_type=";
+  if (!unrecognized_chunk_.empty()) {
+    sb << static_cast<int>(unrecognized_chunk_[0]);
+  } else {
+    sb << "<missing>";
+  }
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h
new file mode 100644
index 0000000000..26d3d3b8f9
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_UNRECOGNIZED_CHUNK_TYPE_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_UNRECOGNIZED_CHUNK_TYPE_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.6
+struct UnrecognizedChunkTypeCauseConfig : public ParameterConfig {
+  static constexpr int kType = 6;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class UnrecognizedChunkTypeCause
+    : public Parameter,
+      public TLVTrait<UnrecognizedChunkTypeCauseConfig> {
+ public:
+  static constexpr int kType = UnrecognizedChunkTypeCauseConfig::kType;
+
+  explicit UnrecognizedChunkTypeCause(std::vector<uint8_t> unrecognized_chunk)
+      : unrecognized_chunk_(std::move(unrecognized_chunk)) {}
+
+  static absl::optional<UnrecognizedChunkTypeCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  rtc::ArrayView<const uint8_t> unrecognized_chunk() const {
+    return unrecognized_chunk_;
+  }
+
+ private:
+  std::vector<uint8_t> unrecognized_chunk_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_UNRECOGNIZED_CHUNK_TYPE_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause_test.cc
new file mode 100644
index 0000000000..baff852f40
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause_test.cc
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(UnrecognizedChunkTypeCauseTest, SerializeAndDeserialize) {
+  UnrecognizedChunkTypeCause parameter({1, 2, 3});
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(UnrecognizedChunkTypeCause deserialized,
+                              UnrecognizedChunkTypeCause::Parse(serialized));
+
+  EXPECT_THAT(deserialized.unrecognized_chunk(), ElementsAre(1, 2, 3));
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc
new file mode 100644
index 0000000000..80001a9eae
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.8
+
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Cause Code=8              |      Cause Length             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                  Unrecognized Parameters                      /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int UnrecognizedParametersCause::kType;
+
+absl::optional<UnrecognizedParametersCause> UnrecognizedParametersCause::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  return UnrecognizedParametersCause(reader->variable_data());
+}
+
+void UnrecognizedParametersCause::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer =
+      AllocateTLV(out, unrecognized_parameters_.size());
+  writer.CopyToVariableData(unrecognized_parameters_);
+}
+
+std::string UnrecognizedParametersCause::ToString() const {
+  return "Unrecognized Parameters";
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h
new file mode 100644
index 0000000000..ebec5ed4c3
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_UNRECOGNIZED_PARAMETER_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_UNRECOGNIZED_PARAMETER_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.8
+struct UnrecognizedParametersCauseConfig : public ParameterConfig {
+  static constexpr int kType = 8;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class UnrecognizedParametersCause
+    : public Parameter,
+      public TLVTrait<UnrecognizedParametersCauseConfig> {
+ public:
+  static constexpr int kType = UnrecognizedParametersCauseConfig::kType;
+
+  explicit UnrecognizedParametersCause(
+      rtc::ArrayView<const uint8_t> unrecognized_parameters)
+      : unrecognized_parameters_(unrecognized_parameters.begin(),
+                                 unrecognized_parameters.end()) {}
+
+  static absl::optional<UnrecognizedParametersCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  rtc::ArrayView<const uint8_t> unrecognized_parameters() const {
+    return unrecognized_parameters_;
+  }
+
+ private:
+  std::vector<uint8_t> unrecognized_parameters_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_UNRECOGNIZED_PARAMETER_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_parameter_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_parameter_cause_test.cc
new file mode 100644
index 0000000000..0449599ca6
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unrecognized_parameter_cause_test.cc
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(UnrecognizedParametersCauseTest, SerializeAndDeserialize) {
+  uint8_t unrecognized_parameters[] = {1, 2, 3};
+  UnrecognizedParametersCause parameter(unrecognized_parameters);
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(UnrecognizedParametersCause deserialized,
+                              UnrecognizedParametersCause::Parse(serialized));
+
+  EXPECT_THAT(deserialized.unrecognized_parameters(), ElementsAre(1, 2, 3));
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc
new file mode 100644
index 0000000000..8108d31aa7
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/unresolvable_address_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.5
+
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Cause Code=5              |      Cause Length             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                  Unresolvable Address                         /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int UnresolvableAddressCause::kType;
+
+absl::optional<UnresolvableAddressCause> UnresolvableAddressCause::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  return UnresolvableAddressCause(reader->variable_data());
+}
+
+void UnresolvableAddressCause::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer =
+      AllocateTLV(out, unresolvable_address_.size());
+  writer.CopyToVariableData(unresolvable_address_);
+}
+
+std::string UnresolvableAddressCause::ToString() const {
+  return "Unresolvable Address";
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/unresolvable_address_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unresolvable_address_cause.h
new file mode 100644
index 0000000000..c63b3779ef
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unresolvable_address_cause.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_UNRESOLVABLE_ADDRESS_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_UNRESOLVABLE_ADDRESS_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.5
+struct UnresolvableAddressCauseConfig : public ParameterConfig {
+  static constexpr int kType = 5;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class UnresolvableAddressCause
+    : public Parameter,
+      public TLVTrait<UnresolvableAddressCauseConfig> {
+ public:
+  static constexpr int kType = UnresolvableAddressCauseConfig::kType;
+
+  explicit UnresolvableAddressCause(
+      rtc::ArrayView<const uint8_t> unresolvable_address)
+      : unresolvable_address_(unresolvable_address.begin(),
+                              unresolvable_address.end()) {}
+
+  static absl::optional<UnresolvableAddressCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  rtc::ArrayView<const uint8_t> unresolvable_address() const {
+    return unresolvable_address_;
+  }
+
+ private:
+  std::vector<uint8_t> unresolvable_address_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_UNRESOLVABLE_ADDRESS_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/unresolvable_address_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unresolvable_address_cause_test.cc
new file mode 100644
index 0000000000..688730e6b3
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/unresolvable_address_cause_test.cc
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/unresolvable_address_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(UnresolvableAddressCauseTest, SerializeAndDeserialize) {
+  uint8_t unresolvable_address[] = {1, 2, 3};
+  UnresolvableAddressCause parameter(unresolvable_address);
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(UnresolvableAddressCause deserialized,
+                              UnresolvableAddressCause::Parse(serialized));
+
+  EXPECT_THAT(deserialized.unresolvable_address(), ElementsAre(1, 2, 3));
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc
new file mode 100644
index 0000000000..da99aacbfa
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/user_initiated_abort_cause.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.12
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |         Cause Code=12         |      Cause Length=Variable    |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                    Upper Layer Abort Reason                   /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int UserInitiatedAbortCause::kType;
+
+absl::optional<UserInitiatedAbortCause> UserInitiatedAbortCause::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  if (reader->variable_data().empty()) {
+    return UserInitiatedAbortCause("");
+  }
+  return UserInitiatedAbortCause(
+      std::string(reinterpret_cast<const char*>(reader->variable_data().data()),
+                  reader->variable_data().size()));
+}
+
+void UserInitiatedAbortCause::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer =
+      AllocateTLV(out, upper_layer_abort_reason_.size());
+  writer.CopyToVariableData(rtc::MakeArrayView(
+      reinterpret_cast<const uint8_t*>(upper_layer_abort_reason_.data()),
+      upper_layer_abort_reason_.size()));
+}
+
+std::string UserInitiatedAbortCause::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "User-Initiated Abort, reason=" << upper_layer_abort_reason_;
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h b/third_party/libwebrtc/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h
new file mode 100644
index 0000000000..9eb16657b4
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_ERROR_CAUSE_USER_INITIATED_ABORT_CAUSE_H_
+#define NET_DCSCTP_PACKET_ERROR_CAUSE_USER_INITIATED_ABORT_CAUSE_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.10.12
+struct UserInitiatedAbortCauseConfig : public ParameterConfig {
+  static constexpr int kType = 12;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class UserInitiatedAbortCause : public Parameter,
+                                public TLVTrait<UserInitiatedAbortCauseConfig> {
+ public:
+  static constexpr int kType = UserInitiatedAbortCauseConfig::kType;
+
+  explicit UserInitiatedAbortCause(absl::string_view upper_layer_abort_reason)
+      : upper_layer_abort_reason_(upper_layer_abort_reason) {}
+
+  static absl::optional<UserInitiatedAbortCause> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  absl::string_view upper_layer_abort_reason() const {
+    return upper_layer_abort_reason_;
+  }
+
+ private:
+  std::string upper_layer_abort_reason_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_ERROR_CAUSE_USER_INITIATED_ABORT_CAUSE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/error_cause/user_initiated_abort_cause_test.cc b/third_party/libwebrtc/net/dcsctp/packet/error_cause/user_initiated_abort_cause_test.cc
new file mode 100644
index 0000000000..250959e3df
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/error_cause/user_initiated_abort_cause_test.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/error_cause/user_initiated_abort_cause.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::SizeIs;
+
+TEST(UserInitiatedAbortCauseTest, EmptyReason) {
+  Parameters causes =
+      Parameters::Builder().Add(UserInitiatedAbortCause("")).Build();
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(Parameters deserialized,
+                              Parameters::Parse(causes.data()));
+  ASSERT_THAT(deserialized.descriptors(), SizeIs(1));
+  EXPECT_EQ(deserialized.descriptors()[0].type, UserInitiatedAbortCause::kType);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      UserInitiatedAbortCause cause,
+      UserInitiatedAbortCause::Parse(deserialized.descriptors()[0].data));
+
+  EXPECT_EQ(cause.upper_layer_abort_reason(), "");
+}
+
+TEST(UserInitiatedAbortCauseTest, SetReason) {
+  Parameters causes = Parameters::Builder()
+                          .Add(UserInitiatedAbortCause("User called Close"))
+                          .Build();
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(Parameters deserialized,
+                              Parameters::Parse(causes.data()));
+  ASSERT_THAT(deserialized.descriptors(), SizeIs(1));
+  EXPECT_EQ(deserialized.descriptors()[0].type, UserInitiatedAbortCause::kType);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      UserInitiatedAbortCause cause,
+      UserInitiatedAbortCause::Parse(deserialized.descriptors()[0].data));
+
+  EXPECT_EQ(cause.upper_layer_abort_reason(), "User called Close");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc
new file mode 100644
index 0000000000..c33e3e11f6
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.6
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Parameter Type = 18       |      Parameter Length = 12    |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |          Re-configuration Request Sequence Number             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |      Number of new streams    |         Reserved              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int AddIncomingStreamsRequestParameter::kType;
+
+absl::optional<AddIncomingStreamsRequestParameter>
+AddIncomingStreamsRequestParameter::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  ReconfigRequestSN request_sequence_number(reader->Load32<4>());
+  uint16_t nbr_of_new_streams = reader->Load16<8>();
+
+  return AddIncomingStreamsRequestParameter(request_sequence_number,
+                                            nbr_of_new_streams);
+}
+
+void AddIncomingStreamsRequestParameter::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out);
+  writer.Store32<4>(*request_sequence_number_);
+  writer.Store16<8>(nbr_of_new_streams_);
+}
+
+std::string AddIncomingStreamsRequestParameter::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Add Incoming Streams Request, req_seq_nbr="
+     << *request_sequence_number();
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h
new file mode 100644
index 0000000000..3859eb3f7e
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_ADD_INCOMING_STREAMS_REQUEST_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_ADD_INCOMING_STREAMS_REQUEST_PARAMETER_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.6
+struct AddIncomingStreamsRequestParameterConfig : ParameterConfig {
+  static constexpr int kType = 18;
+  static constexpr size_t kHeaderSize = 12;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class AddIncomingStreamsRequestParameter
+    : public Parameter,
+      public TLVTrait<AddIncomingStreamsRequestParameterConfig> {
+ public:
+  static constexpr int kType = AddIncomingStreamsRequestParameterConfig::kType;
+
+  explicit AddIncomingStreamsRequestParameter(
+      ReconfigRequestSN request_sequence_number,
+      uint16_t nbr_of_new_streams)
+      : request_sequence_number_(request_sequence_number),
+        nbr_of_new_streams_(nbr_of_new_streams) {}
+
+  static absl::optional<AddIncomingStreamsRequestParameter> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  ReconfigRequestSN request_sequence_number() const {
+    return request_sequence_number_;
+  }
+  uint16_t nbr_of_new_streams() const { return nbr_of_new_streams_; }
+
+ private:
+  ReconfigRequestSN request_sequence_number_;
+  uint16_t nbr_of_new_streams_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_ADD_INCOMING_STREAMS_REQUEST_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter_test.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter_test.cc
new file mode 100644
index 0000000000..a29257a8f8
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter_test.cc
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(AddIncomingStreamsRequestParameterTest, SerializeAndDeserialize) {
+  AddIncomingStreamsRequestParameter parameter(ReconfigRequestSN(1), 2);
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      AddIncomingStreamsRequestParameter deserialized,
+      AddIncomingStreamsRequestParameter::Parse(serialized));
+
+  EXPECT_EQ(*deserialized.request_sequence_number(), 1u);
+  EXPECT_EQ(deserialized.nbr_of_new_streams(), 2u);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc
new file mode 100644
index 0000000000..4787ee9718
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.5
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Parameter Type = 17       |      Parameter Length = 12    |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |          Re-configuration Request Sequence Number             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |      Number of new streams    |         Reserved              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int AddOutgoingStreamsRequestParameter::kType;
+
+absl::optional<AddOutgoingStreamsRequestParameter>
+AddOutgoingStreamsRequestParameter::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  ReconfigRequestSN request_sequence_number(reader->Load32<4>());
+  uint16_t nbr_of_new_streams = reader->Load16<8>();
+
+  return AddOutgoingStreamsRequestParameter(request_sequence_number,
+                                            nbr_of_new_streams);
+}
+
+void AddOutgoingStreamsRequestParameter::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out);
+  writer.Store32<4>(*request_sequence_number_);
+  writer.Store16<8>(nbr_of_new_streams_);
+}
+
+std::string AddOutgoingStreamsRequestParameter::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Add Outgoing Streams Request, req_seq_nbr="
+     << *request_sequence_number();
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h
new file mode 100644
index 0000000000..01e8f91cfa
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_ADD_OUTGOING_STREAMS_REQUEST_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_ADD_OUTGOING_STREAMS_REQUEST_PARAMETER_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.5
+struct AddOutgoingStreamsRequestParameterConfig : ParameterConfig {
+  static constexpr int kType = 17;
+  static constexpr size_t kHeaderSize = 12;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class AddOutgoingStreamsRequestParameter
+    : public Parameter,
+      public TLVTrait<AddOutgoingStreamsRequestParameterConfig> {
+ public:
+  static constexpr int kType = AddOutgoingStreamsRequestParameterConfig::kType;
+
+  explicit AddOutgoingStreamsRequestParameter(
+      ReconfigRequestSN request_sequence_number,
+      uint16_t nbr_of_new_streams)
+      : request_sequence_number_(request_sequence_number),
+        nbr_of_new_streams_(nbr_of_new_streams) {}
+
+  static absl::optional<AddOutgoingStreamsRequestParameter> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  ReconfigRequestSN request_sequence_number() const {
+    return request_sequence_number_;
+  }
+  uint16_t nbr_of_new_streams() const { return nbr_of_new_streams_; }
+
+ private:
+  ReconfigRequestSN request_sequence_number_;
+  uint16_t nbr_of_new_streams_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_ADD_OUTGOING_STREAMS_REQUEST_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter_test.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter_test.cc
new file mode 100644
index 0000000000..d0303b1ba8
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter_test.cc
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(AddOutgoingStreamsRequestParameterTest, SerializeAndDeserialize) {
+  AddOutgoingStreamsRequestParameter parameter(ReconfigRequestSN(1), 2);
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      AddOutgoingStreamsRequestParameter deserialized,
+      AddOutgoingStreamsRequestParameter::Parse(serialized));
+
+  EXPECT_EQ(*deserialized.request_sequence_number(), 1u);
+  EXPECT_EQ(deserialized.nbr_of_new_streams(), 2u);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc
new file mode 100644
index 0000000000..7dd8e1923f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h"
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc3758#section-3.1
+
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |    Parameter Type = 49152     |  Parameter Length = 4         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int ForwardTsnSupportedParameter::kType;
+
+absl::optional<ForwardTsnSupportedParameter>
+ForwardTsnSupportedParameter::Parse(rtc::ArrayView<const uint8_t> data) {
+  if (!ParseTLV(data).has_value()) {
+    return absl::nullopt;
+  }
+  return ForwardTsnSupportedParameter();
+}
+
+void ForwardTsnSupportedParameter::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  AllocateTLV(out);
+}
+
+std::string ForwardTsnSupportedParameter::ToString() const {
+  return "Forward TSN Supported";
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h
new file mode 100644
index 0000000000..d4cff4ac21
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_FORWARD_TSN_SUPPORTED_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_FORWARD_TSN_SUPPORTED_PARAMETER_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc3758#section-3.1
+struct ForwardTsnSupportedParameterConfig : ParameterConfig {
+  static constexpr int kType = 49152;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class ForwardTsnSupportedParameter
+    : public Parameter,
+      public TLVTrait<ForwardTsnSupportedParameterConfig> {
+ public:
+  static constexpr int kType = ForwardTsnSupportedParameterConfig::kType;
+
+  ForwardTsnSupportedParameter() {}
+
+  static absl::optional<ForwardTsnSupportedParameter> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_FORWARD_TSN_SUPPORTED_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/forward_tsn_supported_parameter_test.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/forward_tsn_supported_parameter_test.cc
new file mode 100644
index 0000000000..fb4f983fae
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/forward_tsn_supported_parameter_test.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(ForwardTsnSupportedParameterTest, SerializeAndDeserialize) {
+  ForwardTsnSupportedParameter parameter;
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(ForwardTsnSupportedParameter deserialized,
+                              ForwardTsnSupportedParameter::Parse(serialized));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc
new file mode 100644
index 0000000000..918976d305
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.5
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   Type = 4    | Chunk  Flags  |      Heartbeat Length         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  \                                                               \
+//  /            Heartbeat Information TLV (Variable-Length)        /
+//  \                                                               \
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int HeartbeatInfoParameter::kType;
+
+absl::optional<HeartbeatInfoParameter> HeartbeatInfoParameter::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  return HeartbeatInfoParameter(reader->variable_data());
+}
+
+void HeartbeatInfoParameter::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, info_.size());
+  writer.CopyToVariableData(info_);
+}
+
+std::string HeartbeatInfoParameter::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Heartbeat Info parameter (info_length=" << info_.size() << ")";
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/heartbeat_info_parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/heartbeat_info_parameter.h
new file mode 100644
index 0000000000..ec503a94b2
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/heartbeat_info_parameter.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_HEARTBEAT_INFO_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_HEARTBEAT_INFO_PARAMETER_H_
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.5
+struct HeartbeatInfoParameterConfig : ParameterConfig {
+  static constexpr int kType = 1;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class HeartbeatInfoParameter : public Parameter,
+                               public TLVTrait<HeartbeatInfoParameterConfig> {
+ public:
+  static constexpr int kType = HeartbeatInfoParameterConfig::kType;
+
+  explicit HeartbeatInfoParameter(rtc::ArrayView<const uint8_t> info)
+      : info_(info.begin(), info.end()) {}
+
+  static absl::optional<HeartbeatInfoParameter> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  rtc::ArrayView<const uint8_t> info() const { return info_; }
+
+ private:
+  std::vector<uint8_t> info_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_HEARTBEAT_INFO_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc
new file mode 100644
index 0000000000..6191adfe9d
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.2
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Parameter Type = 14       |  Parameter Length = 8 + 2 * N |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |          Re-configuration Request Sequence Number             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  Stream Number 1 (optional)   |    Stream Number 2 (optional) |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                            ......                             /
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  Stream Number N-1 (optional) |    Stream Number N (optional) |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int IncomingSSNResetRequestParameter::kType;
+
+absl::optional<IncomingSSNResetRequestParameter>
+IncomingSSNResetRequestParameter::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  ReconfigRequestSN request_sequence_number(reader->Load32<4>());
+
+  size_t stream_count = reader->variable_data_size() / kStreamIdSize;
+  std::vector<StreamID> stream_ids;
+  stream_ids.reserve(stream_count);
+  for (size_t i = 0; i < stream_count; ++i) {
+    BoundedByteReader<kStreamIdSize> sub_reader =
+        reader->sub_reader<kStreamIdSize>(i * kStreamIdSize);
+
+    stream_ids.push_back(StreamID(sub_reader.Load16<0>()));
+  }
+
+  return IncomingSSNResetRequestParameter(request_sequence_number,
+                                          std::move(stream_ids));
+}
+
+void IncomingSSNResetRequestParameter::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  size_t variable_size = stream_ids_.size() * kStreamIdSize;
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, variable_size);
+
+  writer.Store32<4>(*request_sequence_number_);
+
+  for (size_t i = 0; i < stream_ids_.size(); ++i) {
+    BoundedByteWriter<kStreamIdSize> sub_writer =
+        writer.sub_writer<kStreamIdSize>(i * kStreamIdSize);
+    sub_writer.Store16<0>(*stream_ids_[i]);
+  }
+}
+
+std::string IncomingSSNResetRequestParameter::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Incoming SSN Reset Request, req_seq_nbr="
+     << *request_sequence_number();
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h
new file mode 100644
index 0000000000..18963efafc
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_INCOMING_SSN_RESET_REQUEST_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_INCOMING_SSN_RESET_REQUEST_PARAMETER_H_
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.2
+struct IncomingSSNResetRequestParameterConfig : ParameterConfig {
+  static constexpr int kType = 14;
+  static constexpr size_t kHeaderSize = 8;
+  static constexpr size_t kVariableLengthAlignment = 2;
+};
+
+class IncomingSSNResetRequestParameter
+    : public Parameter,
+      public TLVTrait<IncomingSSNResetRequestParameterConfig> {
+ public:
+  static constexpr int kType = IncomingSSNResetRequestParameterConfig::kType;
+
+  explicit IncomingSSNResetRequestParameter(
+      ReconfigRequestSN request_sequence_number,
+      std::vector<StreamID> stream_ids)
+      : request_sequence_number_(request_sequence_number),
+        stream_ids_(std::move(stream_ids)) {}
+
+  static absl::optional<IncomingSSNResetRequestParameter> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  ReconfigRequestSN request_sequence_number() const {
+    return request_sequence_number_;
+  }
+  rtc::ArrayView<const StreamID> stream_ids() const { return stream_ids_; }
+
+ private:
+  static constexpr size_t kStreamIdSize = sizeof(uint16_t);
+
+  ReconfigRequestSN request_sequence_number_;
+  std::vector<StreamID> stream_ids_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_INCOMING_SSN_RESET_REQUEST_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter_test.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter_test.cc
new file mode 100644
index 0000000000..17793f6638
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter_test.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(IncomingSSNResetRequestParameterTest, SerializeAndDeserialize) {
+  IncomingSSNResetRequestParameter parameter(
+      ReconfigRequestSN(1), {StreamID(2), StreamID(3), StreamID(4)});
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      IncomingSSNResetRequestParameter deserialized,
+      IncomingSSNResetRequestParameter::Parse(serialized));
+
+  EXPECT_EQ(*deserialized.request_sequence_number(), 1u);
+  EXPECT_THAT(deserialized.stream_ids(),
+              ElementsAre(StreamID(2), StreamID(3), StreamID(4)));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc
new file mode 100644
index 0000000000..c25a2426be
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/public/types.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.1
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Parameter Type = 13       | Parameter Length = 16 + 2 * N |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |           Re-configuration Request Sequence Number            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |           Re-configuration Response Sequence Number           |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                Sender's Last Assigned TSN                     |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  Stream Number 1 (optional)   |    Stream Number 2 (optional) |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  /                            ......                             /
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  Stream Number N-1 (optional) |    Stream Number N (optional) |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int OutgoingSSNResetRequestParameter::kType;
+
+absl::optional<OutgoingSSNResetRequestParameter>
+OutgoingSSNResetRequestParameter::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  ReconfigRequestSN request_sequence_number(reader->Load32<4>());
+  ReconfigRequestSN response_sequence_number(reader->Load32<8>());
+  TSN sender_last_assigned_tsn(reader->Load32<12>());
+
+  size_t stream_count = reader->variable_data_size() / kStreamIdSize;
+  std::vector<StreamID> stream_ids;
+  stream_ids.reserve(stream_count);
+  for (size_t i = 0; i < stream_count; ++i) {
+    BoundedByteReader<kStreamIdSize> sub_reader =
+        reader->sub_reader<kStreamIdSize>(i * kStreamIdSize);
+
+    stream_ids.push_back(StreamID(sub_reader.Load16<0>()));
+  }
+
+  return OutgoingSSNResetRequestParameter(
+      request_sequence_number, response_sequence_number,
+      sender_last_assigned_tsn, std::move(stream_ids));
+}
+
+void OutgoingSSNResetRequestParameter::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  size_t variable_size = stream_ids_.size() * kStreamIdSize;
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, variable_size);
+
+  writer.Store32<4>(*request_sequence_number_);
+  writer.Store32<8>(*response_sequence_number_);
+  writer.Store32<12>(*sender_last_assigned_tsn_);
+
+  for (size_t i = 0; i < stream_ids_.size(); ++i) {
+    BoundedByteWriter<kStreamIdSize> sub_writer =
+        writer.sub_writer<kStreamIdSize>(i * kStreamIdSize);
+    sub_writer.Store16<0>(*stream_ids_[i]);
+  }
+}
+
+std::string OutgoingSSNResetRequestParameter::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Outgoing SSN Reset Request, req_seq_nbr=" << *request_sequence_number()
+     << ", resp_seq_nbr=" << *response_sequence_number()
+     << ", sender_last_asg_tsn=" << *sender_last_assigned_tsn();
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h
new file mode 100644
index 0000000000..6eb44e079f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h
@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_OUTGOING_SSN_RESET_REQUEST_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_OUTGOING_SSN_RESET_REQUEST_PARAMETER_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.1
+struct OutgoingSSNResetRequestParameterConfig : ParameterConfig {
+  static constexpr int kType = 13;
+  static constexpr size_t kHeaderSize = 16;
+  static constexpr size_t kVariableLengthAlignment = 2;
+};
+
+class OutgoingSSNResetRequestParameter
+    : public Parameter,
+      public TLVTrait<OutgoingSSNResetRequestParameterConfig> {
+ public:
+  static constexpr int kType = OutgoingSSNResetRequestParameterConfig::kType;
+
+  explicit OutgoingSSNResetRequestParameter(
+      ReconfigRequestSN request_sequence_number,
+      ReconfigRequestSN response_sequence_number,
+      TSN sender_last_assigned_tsn,
+      std::vector<StreamID> stream_ids)
+      : request_sequence_number_(request_sequence_number),
+        response_sequence_number_(response_sequence_number),
+        sender_last_assigned_tsn_(sender_last_assigned_tsn),
+        stream_ids_(std::move(stream_ids)) {}
+
+  static absl::optional<OutgoingSSNResetRequestParameter> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  ReconfigRequestSN request_sequence_number() const {
+    return request_sequence_number_;
+  }
+  ReconfigRequestSN response_sequence_number() const {
+    return response_sequence_number_;
+  }
+  TSN sender_last_assigned_tsn() const { return sender_last_assigned_tsn_; }
+  rtc::ArrayView<const StreamID> stream_ids() const { return stream_ids_; }
+
+ private:
+  static constexpr size_t kStreamIdSize = sizeof(uint16_t);
+
+  ReconfigRequestSN request_sequence_number_;
+  ReconfigRequestSN response_sequence_number_;
+  TSN sender_last_assigned_tsn_;
+  std::vector<StreamID> stream_ids_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_OUTGOING_SSN_RESET_REQUEST_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter_test.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter_test.cc
new file mode 100644
index 0000000000..dae73c2fba
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter_test.cc
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(OutgoingSSNResetRequestParameterTest, SerializeAndDeserialize) {
+  OutgoingSSNResetRequestParameter parameter(
+      ReconfigRequestSN(1), ReconfigRequestSN(2), TSN(3),
+      {StreamID(4), StreamID(5), StreamID(6)});
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter deserialized,
+      OutgoingSSNResetRequestParameter::Parse(serialized));
+
+  EXPECT_EQ(*deserialized.request_sequence_number(), 1u);
+  EXPECT_EQ(*deserialized.response_sequence_number(), 2u);
+  EXPECT_EQ(*deserialized.sender_last_assigned_tsn(), 3u);
+  EXPECT_THAT(deserialized.stream_ids(),
+              ElementsAre(StreamID(4), StreamID(5), StreamID(6)));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/parameter.cc
new file mode 100644
index 0000000000..b3b2bffef7
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/parameter.cc
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/parameter.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/math.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h"
+#include "net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h"
+#include "net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h"
+#include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
+#include "net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/reconfiguration_response_parameter.h"
+#include "net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/state_cookie_parameter.h"
+#include "net/dcsctp/packet/parameter/supported_extensions_parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+constexpr size_t kParameterHeaderSize = 4;
+
+Parameters::Builder& Parameters::Builder::Add(const Parameter& p) {
+  // https://tools.ietf.org/html/rfc4960#section-3.2.1
+  // "If the length of the parameter is not a multiple of 4 bytes, the sender
+  // pads the parameter at the end (i.e., after the Parameter Value field) with
+  // all zero bytes."
+  if (data_.size() % 4 != 0) {
+    data_.resize(RoundUpTo4(data_.size()));
+  }
+
+  p.SerializeTo(data_);
+  return *this;
+}
+
+std::vector<ParameterDescriptor> Parameters::descriptors() const {
+  rtc::ArrayView<const uint8_t> span(data_);
+  std::vector<ParameterDescriptor> result;
+  while (!span.empty()) {
+    BoundedByteReader<kParameterHeaderSize> header(span);
+    uint16_t type = header.Load16<0>();
+    uint16_t length = header.Load16<2>();
+    result.emplace_back(type, span.subview(0, length));
+    size_t length_with_padding = RoundUpTo4(length);
+    if (length_with_padding > span.size()) {
+      break;
+    }
+    span = span.subview(length_with_padding);
+  }
+  return result;
+}
+
+absl::optional<Parameters> Parameters::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  // Validate the parameter descriptors
+  rtc::ArrayView<const uint8_t> span(data);
+  while (!span.empty()) {
+    if (span.size() < kParameterHeaderSize) {
+      RTC_DLOG(LS_WARNING) << "Insufficient parameter length";
+      return absl::nullopt;
+    }
+    BoundedByteReader<kParameterHeaderSize> header(span);
+    uint16_t length = header.Load16<2>();
+    if (length < kParameterHeaderSize || length > span.size()) {
+      RTC_DLOG(LS_WARNING) << "Invalid parameter length field";
+      return absl::nullopt;
+    }
+    size_t length_with_padding = RoundUpTo4(length);
+    if (length_with_padding > span.size()) {
+      break;
+    }
+    span = span.subview(length_with_padding);
+  }
+  return Parameters(std::vector<uint8_t>(data.begin(), data.end()));
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/parameter.h
new file mode 100644
index 0000000000..e8fa67c8f7
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/parameter.h
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_PARAMETER_H_
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <iterator>
+#include <memory>
+#include <string>
+#include <type_traits>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+class Parameter {
+ public:
+  Parameter() {}
+  virtual ~Parameter() = default;
+
+  Parameter(const Parameter& other) = default;
+  Parameter& operator=(const Parameter& other) = default;
+
+  virtual void SerializeTo(std::vector<uint8_t>& out) const = 0;
+  virtual std::string ToString() const = 0;
+};
+
+struct ParameterDescriptor {
+  ParameterDescriptor(uint16_t type, rtc::ArrayView<const uint8_t> data)
+      : type(type), data(data) {}
+  uint16_t type;
+  rtc::ArrayView<const uint8_t> data;
+};
+
+class Parameters {
+ public:
+  class Builder {
+   public:
+    Builder() {}
+    Builder& Add(const Parameter& p);
+    Parameters Build() { return Parameters(std::move(data_)); }
+
+   private:
+    std::vector<uint8_t> data_;
+  };
+
+  static absl::optional<Parameters> Parse(rtc::ArrayView<const uint8_t> data);
+
+  Parameters() {}
+  Parameters(Parameters&& other) = default;
+  Parameters& operator=(Parameters&& other) = default;
+
+  rtc::ArrayView<const uint8_t> data() const { return data_; }
+  std::vector<ParameterDescriptor> descriptors() const;
+
+  template <typename P>
+  absl::optional<P> get() const {
+    static_assert(std::is_base_of<Parameter, P>::value,
+                  "Template parameter not derived from Parameter");
+    for (const auto& p : descriptors()) {
+      if (p.type == P::kType) {
+        return P::Parse(p.data);
+      }
+    }
+    return absl::nullopt;
+  }
+
+ private:
+  explicit Parameters(std::vector<uint8_t> data) : data_(std::move(data)) {}
+  std::vector<uint8_t> data_;
+};
+
+struct ParameterConfig {
+  static constexpr int kTypeSizeInBytes = 2;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/parameter_test.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/parameter_test.cc
new file mode 100644
index 0000000000..467e324592
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/parameter_test.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/parameter.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+using ::testing::SizeIs;
+
+TEST(ParameterTest, SerializeDeserializeParameter) {
+  Parameters parameters =
+      Parameters::Builder()
+          .Add(OutgoingSSNResetRequestParameter(ReconfigRequestSN(123),
+                                                ReconfigRequestSN(456),
+                                                TSN(789), {StreamID(42)}))
+          .Build();
+
+  rtc::ArrayView<const uint8_t> serialized = parameters.data();
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(Parameters parsed, Parameters::Parse(serialized));
+  auto descriptors = parsed.descriptors();
+  ASSERT_THAT(descriptors, SizeIs(1));
+  EXPECT_THAT(descriptors[0].type, OutgoingSSNResetRequestParameter::kType);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter parsed_param,
+      OutgoingSSNResetRequestParameter::Parse(descriptors[0].data));
+  EXPECT_EQ(*parsed_param.request_sequence_number(), 123u);
+  EXPECT_EQ(*parsed_param.response_sequence_number(), 456u);
+  EXPECT_EQ(*parsed_param.sender_last_assigned_tsn(), 789u);
+  EXPECT_THAT(parsed_param.stream_ids(), ElementsAre(StreamID(42)));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc
new file mode 100644
index 0000000000..fafb204acc
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/reconfiguration_response_parameter.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.4
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Parameter Type = 16       |      Parameter Length         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |         Re-configuration Response Sequence Number             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                            Result                             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                   Sender's Next TSN (optional)                |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  Receiver's Next TSN (optional)               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int ReconfigurationResponseParameter::kType;
+
+absl::string_view ToString(ReconfigurationResponseParameter::Result result) {
+  switch (result) {
+    case ReconfigurationResponseParameter::Result::kSuccessNothingToDo:
+      return "Success: nothing to do";
+    case ReconfigurationResponseParameter::Result::kSuccessPerformed:
+      return "Success: performed";
+    case ReconfigurationResponseParameter::Result::kDenied:
+      return "Denied";
+    case ReconfigurationResponseParameter::Result::kErrorWrongSSN:
+      return "Error: wrong ssn";
+    case ReconfigurationResponseParameter::Result::
+        kErrorRequestAlreadyInProgress:
+      return "Error: request already in progress";
+    case ReconfigurationResponseParameter::Result::kErrorBadSequenceNumber:
+      return "Error: bad sequence number";
+    case ReconfigurationResponseParameter::Result::kInProgress:
+      return "In progress";
+  }
+}
+
+absl::optional<ReconfigurationResponseParameter>
+ReconfigurationResponseParameter::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  ReconfigRequestSN response_sequence_number(reader->Load32<4>());
+  Result result;
+  uint32_t result_nbr = reader->Load32<8>();
+  switch (result_nbr) {
+    case 0:
+      result = ReconfigurationResponseParameter::Result::kSuccessNothingToDo;
+      break;
+    case 1:
+      result = ReconfigurationResponseParameter::Result::kSuccessPerformed;
+      break;
+    case 2:
+      result = ReconfigurationResponseParameter::Result::kDenied;
+      break;
+    case 3:
+      result = ReconfigurationResponseParameter::Result::kErrorWrongSSN;
+      break;
+    case 4:
+      result = ReconfigurationResponseParameter::Result::
+          kErrorRequestAlreadyInProgress;
+      break;
+    case 5:
+      result =
+          ReconfigurationResponseParameter::Result::kErrorBadSequenceNumber;
+      break;
+    case 6:
+      result = ReconfigurationResponseParameter::Result::kInProgress;
+      break;
+    default:
+      RTC_DLOG(LS_WARNING) << "Invalid reconfig response result: "
+                           << result_nbr;
+      return absl::nullopt;
+  }
+
+  if (reader->variable_data().empty()) {
+    return ReconfigurationResponseParameter(response_sequence_number, result);
+  } else if (reader->variable_data_size() != kNextTsnHeaderSize) {
+    RTC_DLOG(LS_WARNING) << "Invalid parameter size";
+    return absl::nullopt;
+  }
+
+  BoundedByteReader<kNextTsnHeaderSize> sub_reader =
+      reader->sub_reader<kNextTsnHeaderSize>(0);
+
+  TSN sender_next_tsn(sub_reader.Load32<0>());
+  TSN receiver_next_tsn(sub_reader.Load32<4>());
+
+  return ReconfigurationResponseParameter(response_sequence_number, result,
+                                          sender_next_tsn, receiver_next_tsn);
+}
+
+void ReconfigurationResponseParameter::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  size_t variable_size =
+      (sender_next_tsn().has_value() ? kNextTsnHeaderSize : 0);
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, variable_size);
+
+  writer.Store32<4>(*response_sequence_number_);
+  uint32_t result_nbr =
+      static_cast<std::underlying_type<Result>::type>(result_);
+  writer.Store32<8>(result_nbr);
+
+  if (sender_next_tsn().has_value()) {
+    BoundedByteWriter<kNextTsnHeaderSize> sub_writer =
+        writer.sub_writer<kNextTsnHeaderSize>(0);
+
+    sub_writer.Store32<0>(sender_next_tsn_.has_value() ? **sender_next_tsn_
+                                                       : 0);
+    sub_writer.Store32<4>(receiver_next_tsn_.has_value() ? **receiver_next_tsn_
+                                                         : 0);
+  }
+}
+
+std::string ReconfigurationResponseParameter::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Re-configuration Response, resp_seq_nbr="
+     << *response_sequence_number();
+  return sb.Release();
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h
new file mode 100644
index 0000000000..c5a68acb33
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_RECONFIGURATION_RESPONSE_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_RECONFIGURATION_RESPONSE_PARAMETER_H_
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.4
+struct ReconfigurationResponseParameterConfig : ParameterConfig {
+  static constexpr int kType = 16;
+  static constexpr size_t kHeaderSize = 12;
+  static constexpr size_t kVariableLengthAlignment = 4;
+};
+
+class ReconfigurationResponseParameter
+    : public Parameter,
+      public TLVTrait<ReconfigurationResponseParameterConfig> {
+ public:
+  static constexpr int kType = ReconfigurationResponseParameterConfig::kType;
+
+  enum class Result {
+    kSuccessNothingToDo = 0,
+    kSuccessPerformed = 1,
+    kDenied = 2,
+    kErrorWrongSSN = 3,
+    kErrorRequestAlreadyInProgress = 4,
+    kErrorBadSequenceNumber = 5,
+    kInProgress = 6,
+  };
+
+  ReconfigurationResponseParameter(ReconfigRequestSN response_sequence_number,
+                                   Result result)
+      : response_sequence_number_(response_sequence_number),
+        result_(result),
+        sender_next_tsn_(absl::nullopt),
+        receiver_next_tsn_(absl::nullopt) {}
+
+  explicit ReconfigurationResponseParameter(
+      ReconfigRequestSN response_sequence_number,
+      Result result,
+      TSN sender_next_tsn,
+      TSN receiver_next_tsn)
+      : response_sequence_number_(response_sequence_number),
+        result_(result),
+        sender_next_tsn_(sender_next_tsn),
+        receiver_next_tsn_(receiver_next_tsn) {}
+
+  static absl::optional<ReconfigurationResponseParameter> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  ReconfigRequestSN response_sequence_number() const {
+    return response_sequence_number_;
+  }
+  Result result() const { return result_; }
+  absl::optional<TSN> sender_next_tsn() const { return sender_next_tsn_; }
+  absl::optional<TSN> receiver_next_tsn() const { return receiver_next_tsn_; }
+
+ private:
+  static constexpr size_t kNextTsnHeaderSize = 8;
+  ReconfigRequestSN response_sequence_number_;
+  Result result_;
+  absl::optional<TSN> sender_next_tsn_;
+  absl::optional<TSN> receiver_next_tsn_;
+};
+
+absl::string_view ToString(ReconfigurationResponseParameter::Result result);
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_RECONFIGURATION_RESPONSE_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/reconfiguration_response_parameter_test.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/reconfiguration_response_parameter_test.cc
new file mode 100644
index 0000000000..8125d93cd0
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/reconfiguration_response_parameter_test.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/reconfiguration_response_parameter.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(ReconfigurationResponseParameterTest, SerializeAndDeserializeFirstForm) {
+  ReconfigurationResponseParameter parameter(
+      ReconfigRequestSN(1),
+      ReconfigurationResponseParameter::Result::kSuccessPerformed);
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      ReconfigurationResponseParameter deserialized,
+      ReconfigurationResponseParameter::Parse(serialized));
+
+  EXPECT_EQ(*deserialized.response_sequence_number(), 1u);
+  EXPECT_EQ(deserialized.result(),
+            ReconfigurationResponseParameter::Result::kSuccessPerformed);
+  EXPECT_EQ(deserialized.sender_next_tsn(), absl::nullopt);
+  EXPECT_EQ(deserialized.receiver_next_tsn(), absl::nullopt);
+}
+
+TEST(ReconfigurationResponseParameterTest,
+     SerializeAndDeserializeFirstFormSecondForm) {
+  ReconfigurationResponseParameter parameter(
+      ReconfigRequestSN(1),
+      ReconfigurationResponseParameter::Result::kSuccessPerformed, TSN(2),
+      TSN(3));
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      ReconfigurationResponseParameter deserialized,
+      ReconfigurationResponseParameter::Parse(serialized));
+
+  EXPECT_EQ(*deserialized.response_sequence_number(), 1u);
+  EXPECT_EQ(deserialized.result(),
+            ReconfigurationResponseParameter::Result::kSuccessPerformed);
+  EXPECT_TRUE(deserialized.sender_next_tsn().has_value());
+  EXPECT_EQ(**deserialized.sender_next_tsn(), 2u);
+  EXPECT_TRUE(deserialized.receiver_next_tsn().has_value());
+  EXPECT_EQ(**deserialized.receiver_next_tsn(), 3u);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc
new file mode 100644
index 0000000000..d656e0db8f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.3
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Parameter Type = 15       |      Parameter Length = 8     |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |         Re-configuration Request Sequence Number              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int SSNTSNResetRequestParameter::kType;
+
+absl::optional<SSNTSNResetRequestParameter> SSNTSNResetRequestParameter::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  ReconfigRequestSN request_sequence_number(reader->Load32<4>());
+
+  return SSNTSNResetRequestParameter(request_sequence_number);
+}
+
+void SSNTSNResetRequestParameter::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out);
+  writer.Store32<4>(*request_sequence_number_);
+}
+
+std::string SSNTSNResetRequestParameter::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "SSN/TSN Reset Request, req_seq_nbr=" << *request_sequence_number();
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h
new file mode 100644
index 0000000000..e31d7ebe8f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_SSN_TSN_RESET_REQUEST_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_SSN_TSN_RESET_REQUEST_PARAMETER_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc6525#section-4.3
+struct SSNTSNResetRequestParameterConfig : ParameterConfig {
+  static constexpr int kType = 15;
+  static constexpr size_t kHeaderSize = 8;
+  static constexpr size_t kVariableLengthAlignment = 0;
+};
+
+class SSNTSNResetRequestParameter
+    : public Parameter,
+      public TLVTrait<SSNTSNResetRequestParameterConfig> {
+ public:
+  static constexpr int kType = SSNTSNResetRequestParameterConfig::kType;
+
+  explicit SSNTSNResetRequestParameter(
+      ReconfigRequestSN request_sequence_number)
+      : request_sequence_number_(request_sequence_number) {}
+
+  static absl::optional<SSNTSNResetRequestParameter> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  ReconfigRequestSN request_sequence_number() const {
+    return request_sequence_number_;
+  }
+
+ private:
+  ReconfigRequestSN request_sequence_number_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_SSN_TSN_RESET_REQUEST_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter_test.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter_test.cc
new file mode 100644
index 0000000000..eeb973cbcb
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter_test.cc
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(SSNTSNResetRequestParameterTest, SerializeAndDeserialize) {
+  SSNTSNResetRequestParameter parameter(ReconfigRequestSN(1));
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SSNTSNResetRequestParameter deserialized,
+                              SSNTSNResetRequestParameter::Parse(serialized));
+
+  EXPECT_EQ(*deserialized.request_sequence_number(), 1u);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/state_cookie_parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/state_cookie_parameter.cc
new file mode 100644
index 0000000000..9777aa6667
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/state_cookie_parameter.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/state_cookie_parameter.h"
+
+#include <stdint.h>
+
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.3.1
+
+constexpr int StateCookieParameter::kType;
+
+absl::optional<StateCookieParameter> StateCookieParameter::Parse(
+    rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+  return StateCookieParameter(reader->variable_data());
+}
+
+void StateCookieParameter::SerializeTo(std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, data_.size());
+  writer.CopyToVariableData(data_);
+}
+
+std::string StateCookieParameter::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "State Cookie parameter (cookie_length=" << data_.size() << ")";
+  return sb.Release();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/state_cookie_parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/state_cookie_parameter.h
new file mode 100644
index 0000000000..f4355495e2
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/state_cookie_parameter.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_STATE_COOKIE_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_STATE_COOKIE_PARAMETER_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.3.1
+struct StateCookieParameterConfig : ParameterConfig {
+  static constexpr int kType = 7;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class StateCookieParameter : public Parameter,
+                             public TLVTrait<StateCookieParameterConfig> {
+ public:
+  static constexpr int kType = StateCookieParameterConfig::kType;
+
+  explicit StateCookieParameter(rtc::ArrayView<const uint8_t> data)
+      : data_(data.begin(), data.end()) {}
+
+  static absl::optional<StateCookieParameter> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  rtc::ArrayView<const uint8_t> data() const { return data_; }
+
+ private:
+  std::vector<uint8_t> data_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_STATE_COOKIE_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/state_cookie_parameter_test.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/state_cookie_parameter_test.cc
new file mode 100644
index 0000000000..bcca38b586
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/state_cookie_parameter_test.cc
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/state_cookie_parameter.h"
+
+#include <stdint.h>
+
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(StateCookieParameterTest, SerializeAndDeserialize) {
+  uint8_t cookie[] = {1, 2, 3};
+  StateCookieParameter parameter(cookie);
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(StateCookieParameter deserialized,
+                              StateCookieParameter::Parse(serialized));
+
+  EXPECT_THAT(deserialized.data(), ElementsAre(1, 2, 3));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/supported_extensions_parameter.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/supported_extensions_parameter.cc
new file mode 100644
index 0000000000..6a8fb214de
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/supported_extensions_parameter.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/supported_extensions_parameter.h"
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/str_join.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc5061#section-4.2.7
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     Parameter Type = 0x8008   |      Parameter Length         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  | CHUNK TYPE 1  |  CHUNK TYPE 2 |  CHUNK TYPE 3 |  CHUNK TYPE 4 |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                             ....                              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  | CHUNK TYPE N  |      PAD      |      PAD      |      PAD      |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr int SupportedExtensionsParameter::kType;
+
+absl::optional<SupportedExtensionsParameter>
+SupportedExtensionsParameter::Parse(rtc::ArrayView<const uint8_t> data) {
+  absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+  if (!reader.has_value()) {
+    return absl::nullopt;
+  }
+
+  std::vector<uint8_t> chunk_types(reader->variable_data().begin(),
+                                   reader->variable_data().end());
+  return SupportedExtensionsParameter(std::move(chunk_types));
+}
+
+void SupportedExtensionsParameter::SerializeTo(
+    std::vector<uint8_t>& out) const {
+  BoundedByteWriter<kHeaderSize> writer = AllocateTLV(out, chunk_types_.size());
+  writer.CopyToVariableData(chunk_types_);
+}
+
+std::string SupportedExtensionsParameter::ToString() const {
+  rtc::StringBuilder sb;
+  sb << "Supported Extensions (" << StrJoin(chunk_types_, ", ") << ")";
+  return sb.Release();
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/supported_extensions_parameter.h b/third_party/libwebrtc/net/dcsctp/packet/parameter/supported_extensions_parameter.h
new file mode 100644
index 0000000000..5689fd8035
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/supported_extensions_parameter.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_PARAMETER_SUPPORTED_EXTENSIONS_PARAMETER_H_
+#define NET_DCSCTP_PACKET_PARAMETER_SUPPORTED_EXTENSIONS_PARAMETER_H_
+#include <stddef.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <iterator>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+
+namespace dcsctp {
+
+// https://tools.ietf.org/html/rfc5061#section-4.2.7
+struct SupportedExtensionsParameterConfig : ParameterConfig {
+  static constexpr int kType = 0x8008;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr size_t kVariableLengthAlignment = 1;
+};
+
+class SupportedExtensionsParameter
+    : public Parameter,
+      public TLVTrait<SupportedExtensionsParameterConfig> {
+ public:
+  static constexpr int kType = SupportedExtensionsParameterConfig::kType;
+
+  explicit SupportedExtensionsParameter(std::vector<uint8_t> chunk_types)
+      : chunk_types_(std::move(chunk_types)) {}
+
+  static absl::optional<SupportedExtensionsParameter> Parse(
+      rtc::ArrayView<const uint8_t> data);
+
+  void SerializeTo(std::vector<uint8_t>& out) const override;
+  std::string ToString() const override;
+
+  bool supports(uint8_t chunk_type) const {
+    return std::find(chunk_types_.begin(), chunk_types_.end(), chunk_type) !=
+           chunk_types_.end();
+  }
+
+  rtc::ArrayView<const uint8_t> chunk_types() const { return chunk_types_; }
+
+ private:
+  std::vector<uint8_t> chunk_types_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_PARAMETER_SUPPORTED_EXTENSIONS_PARAMETER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/parameter/supported_extensions_parameter_test.cc b/third_party/libwebrtc/net/dcsctp/packet/parameter/supported_extensions_parameter_test.cc
new file mode 100644
index 0000000000..c870af2e70
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/parameter/supported_extensions_parameter_test.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/parameter/supported_extensions_parameter.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+
+TEST(SupportedExtensionsParameterTest, SerializeAndDeserialize) {
+  SupportedExtensionsParameter parameter({1, 2, 3});
+
+  std::vector<uint8_t> serialized;
+  parameter.SerializeTo(serialized);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SupportedExtensionsParameter deserialized,
+                              SupportedExtensionsParameter::Parse(serialized));
+
+  EXPECT_THAT(deserialized.chunk_types(), ElementsAre(1, 2, 3));
+  EXPECT_TRUE(deserialized.supports(1));
+  EXPECT_TRUE(deserialized.supports(2));
+  EXPECT_TRUE(deserialized.supports(3));
+  EXPECT_FALSE(deserialized.supports(4));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/sctp_packet.cc b/third_party/libwebrtc/net/dcsctp/packet/sctp_packet.cc
new file mode 100644
index 0000000000..cc66235122
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/sctp_packet.cc
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/sctp_packet.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/math.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/crc32c.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_format.h"
+
+namespace dcsctp {
+namespace {
+constexpr size_t kMaxUdpPacketSize = 65535;
+constexpr size_t kChunkTlvHeaderSize = 4;
+constexpr size_t kExpectedDescriptorCount = 4;
+}  // namespace
+
+/*
+  0                   1                   2                   3
+  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  |     Source Port Number        |     Destination Port Number   |
+  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  |                      Verification Tag                         |
+  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  |                           Checksum                            |
+  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+*/
+
+SctpPacket::Builder::Builder(VerificationTag verification_tag,
+                             const DcSctpOptions& options)
+    : verification_tag_(verification_tag),
+      source_port_(options.local_port),
+      dest_port_(options.remote_port),
+      max_packet_size_(RoundDownTo4(options.mtu)) {}
+
+SctpPacket::Builder& SctpPacket::Builder::Add(const Chunk& chunk) {
+  if (out_.empty()) {
+    out_.reserve(max_packet_size_);
+    out_.resize(SctpPacket::kHeaderSize);
+    BoundedByteWriter<kHeaderSize> buffer(out_);
+    buffer.Store16<0>(source_port_);
+    buffer.Store16<2>(dest_port_);
+    buffer.Store32<4>(*verification_tag_);
+    // Checksum is at offset 8 - written when calling Build();
+  }
+  RTC_DCHECK(IsDivisibleBy4(out_.size()));
+
+  chunk.SerializeTo(out_);
+  if (out_.size() % 4 != 0) {
+    out_.resize(RoundUpTo4(out_.size()));
+  }
+
+  RTC_DCHECK(out_.size() <= max_packet_size_)
+      << "Exceeded max size, data=" << out_.size()
+      << ", max_size=" << max_packet_size_;
+  return *this;
+}
+
+size_t SctpPacket::Builder::bytes_remaining() const {
+  if (out_.empty()) {
+    // The packet header (CommonHeader) hasn't been written yet:
+    return max_packet_size_ - kHeaderSize;
+  } else if (out_.size() > max_packet_size_) {
+    RTC_DCHECK_NOTREACHED() << "Exceeded max size, data=" << out_.size()
+                            << ", max_size=" << max_packet_size_;
+    return 0;
+  }
+  return max_packet_size_ - out_.size();
+}
+
+std::vector<uint8_t> SctpPacket::Builder::Build() {
+  std::vector<uint8_t> out;
+  out_.swap(out);
+
+  if (!out.empty()) {
+    uint32_t crc = GenerateCrc32C(out);
+    BoundedByteWriter<kHeaderSize>(out).Store32<8>(crc);
+  }
+
+  RTC_DCHECK(out.size() <= max_packet_size_)
+      << "Exceeded max size, data=" << out.size()
+      << ", max_size=" << max_packet_size_;
+
+  return out;
+}
+
+absl::optional<SctpPacket> SctpPacket::Parse(
+    rtc::ArrayView<const uint8_t> data,
+    bool disable_checksum_verification) {
+  if (data.size() < kHeaderSize + kChunkTlvHeaderSize ||
+      data.size() > kMaxUdpPacketSize) {
+    RTC_DLOG(LS_WARNING) << "Invalid packet size";
+    return absl::nullopt;
+  }
+
+  BoundedByteReader<kHeaderSize> reader(data);
+
+  CommonHeader common_header;
+  common_header.source_port = reader.Load16<0>();
+  common_header.destination_port = reader.Load16<2>();
+  common_header.verification_tag = VerificationTag(reader.Load32<4>());
+  common_header.checksum = reader.Load32<8>();
+
+  // Create a copy of the packet, which will be held by this object.
+  std::vector<uint8_t> data_copy =
+      std::vector<uint8_t>(data.begin(), data.end());
+
+  // Verify the checksum. The checksum field must be zero when that's done.
+  BoundedByteWriter<kHeaderSize>(data_copy).Store32<8>(0);
+  uint32_t calculated_checksum = GenerateCrc32C(data_copy);
+  if (!disable_checksum_verification &&
+      calculated_checksum != common_header.checksum) {
+    RTC_DLOG(LS_WARNING) << rtc::StringFormat(
+        "Invalid packet checksum, packet_checksum=0x%08x, "
+        "calculated_checksum=0x%08x",
+        common_header.checksum, calculated_checksum);
+    return absl::nullopt;
+  }
+  // Restore the checksum in the header.
+  BoundedByteWriter<kHeaderSize>(data_copy).Store32<8>(common_header.checksum);
+
+  // Validate and parse the chunk headers in the message.
+  /*
+    0                   1                   2                   3
+    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    |   Chunk Type  | Chunk  Flags  |        Chunk Length           |
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  */
+
+  std::vector<ChunkDescriptor> descriptors;
+  descriptors.reserve(kExpectedDescriptorCount);
+  rtc::ArrayView<const uint8_t> descriptor_data =
+      rtc::ArrayView<const uint8_t>(data_copy).subview(kHeaderSize);
+  while (!descriptor_data.empty()) {
+    if (descriptor_data.size() < kChunkTlvHeaderSize) {
+      RTC_DLOG(LS_WARNING) << "Too small chunk";
+      return absl::nullopt;
+    }
+    BoundedByteReader<kChunkTlvHeaderSize> chunk_header(descriptor_data);
+    uint8_t type = chunk_header.Load8<0>();
+    uint8_t flags = chunk_header.Load8<1>();
+    uint16_t length = chunk_header.Load16<2>();
+    uint16_t padded_length = RoundUpTo4(length);
+    if (padded_length > descriptor_data.size()) {
+      RTC_DLOG(LS_WARNING) << "Too large chunk. length=" << length
+                           << ", remaining=" << descriptor_data.size();
+      return absl::nullopt;
+    } else if (padded_length < kChunkTlvHeaderSize) {
+      RTC_DLOG(LS_WARNING) << "Too small chunk. length=" << length;
+      return absl::nullopt;
+    }
+    descriptors.emplace_back(type, flags,
+                             descriptor_data.subview(0, padded_length));
+    descriptor_data = descriptor_data.subview(padded_length);
+  }
+
+  // Note that iterators (and pointer) are guaranteed to be stable when moving a
+  // std::vector, and `descriptors` have pointers to within `data_copy`.
+  return SctpPacket(common_header, std::move(data_copy),
+                    std::move(descriptors));
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/sctp_packet.h b/third_party/libwebrtc/net/dcsctp/packet/sctp_packet.h
new file mode 100644
index 0000000000..4c6234e0c9
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/sctp_packet.h
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_SCTP_PACKET_H_
+#define NET_DCSCTP_PACKET_SCTP_PACKET_H_
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <functional>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+
+namespace dcsctp {
+
+// The "Common Header", which every SCTP packet starts with, and is described in
+// https://tools.ietf.org/html/rfc4960#section-3.1.
+struct CommonHeader {
+  uint16_t source_port;
+  uint16_t destination_port;
+  VerificationTag verification_tag;
+  uint32_t checksum;
+};
+
+// Represents an immutable (received or to-be-sent) SCTP packet.
+class SctpPacket {
+ public:
+  static constexpr size_t kHeaderSize = 12;
+
+  struct ChunkDescriptor {
+    ChunkDescriptor(uint8_t type,
+                    uint8_t flags,
+                    rtc::ArrayView<const uint8_t> data)
+        : type(type), flags(flags), data(data) {}
+    uint8_t type;
+    uint8_t flags;
+    rtc::ArrayView<const uint8_t> data;
+  };
+
+  SctpPacket(SctpPacket&& other) = default;
+  SctpPacket& operator=(SctpPacket&& other) = default;
+  SctpPacket(const SctpPacket&) = delete;
+  SctpPacket& operator=(const SctpPacket&) = delete;
+
+  // Used for building SctpPacket, as those are immutable.
+  class Builder {
+   public:
+    Builder(VerificationTag verification_tag, const DcSctpOptions& options);
+
+    Builder(Builder&& other) = default;
+    Builder& operator=(Builder&& other) = default;
+
+    // Adds a chunk to the to-be-built SCTP packet.
+    Builder& Add(const Chunk& chunk);
+
+    // The number of bytes remaining in the packet for chunk storage until the
+    // packet reaches its maximum size.
+    size_t bytes_remaining() const;
+
+    // Indicates if any packets have been added to the builder.
+    bool empty() const { return out_.empty(); }
+
+    // Returns the payload of the build SCTP packet. The Builder will be cleared
+    // after having called this function, and can be used to build a new packet.
+    std::vector<uint8_t> Build();
+
+   private:
+    VerificationTag verification_tag_;
+    uint16_t source_port_;
+    uint16_t dest_port_;
+    // The maximum packet size is always even divisible by four, as chunks are
+    // always padded to a size even divisible by four.
+    size_t max_packet_size_;
+    std::vector<uint8_t> out_;
+  };
+
+  // Parses `data` as an SCTP packet and returns it if it validates.
+  static absl::optional<SctpPacket> Parse(
+      rtc::ArrayView<const uint8_t> data,
+      bool disable_checksum_verification = false);
+
+  // Returns the SCTP common header.
+  const CommonHeader& common_header() const { return common_header_; }
+
+  // Returns the chunks (types and offsets) within the packet.
+  rtc::ArrayView<const ChunkDescriptor> descriptors() const {
+    return descriptors_;
+  }
+
+ private:
+  SctpPacket(const CommonHeader& common_header,
+             std::vector<uint8_t> data,
+             std::vector<ChunkDescriptor> descriptors)
+      : common_header_(common_header),
+        data_(std::move(data)),
+        descriptors_(std::move(descriptors)) {}
+
+  CommonHeader common_header_;
+
+  // As the `descriptors_` refer to offset within data, and since SctpPacket is
+  // movable, `data` needs to be pointer stable, which it is according to
+  // http://www.open-std.org/JTC1/SC22/WG21/docs/lwg-active.html#2321
+  std::vector<uint8_t> data_;
+  // The chunks and their offsets within `data_ `.
+  std::vector<ChunkDescriptor> descriptors_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_SCTP_PACKET_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/sctp_packet_test.cc b/third_party/libwebrtc/net/dcsctp/packet/sctp_packet_test.cc
new file mode 100644
index 0000000000..7438315eec
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/sctp_packet_test.cc
@@ -0,0 +1,342 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/sctp_packet.h"
+
+#include <cstdint>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/common/math.h"
+#include "net/dcsctp/packet/chunk/abort_chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/init_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/error_cause/user_initiated_abort_cause.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::SizeIs;
+
+constexpr VerificationTag kVerificationTag = VerificationTag(0x12345678);
+
+TEST(SctpPacketTest, DeserializeSimplePacketFromCapture) {
+  /*
+  Stream Control Transmission Protocol, Src Port: 5000 (5000), Dst Port: 5000
+    (5000) Source port: 5000 Destination port: 5000 Verification tag: 0x00000000
+      [Association index: 1]
+      Checksum: 0xaa019d33 [unverified]
+      [Checksum Status: Unverified]
+      INIT chunk (Outbound streams: 1000, inbound streams: 1000)
+          Chunk type: INIT (1)
+          Chunk flags: 0x00
+          Chunk length: 90
+          Initiate tag: 0x0eddca08
+          Advertised receiver window credit (a_rwnd): 131072
+          Number of outbound streams: 1000
+          Number of inbound streams: 1000
+          Initial TSN: 1426601527
+          ECN parameter
+              Parameter type: ECN (0x8000)
+              Parameter length: 4
+          Forward TSN supported parameter
+              Parameter type: Forward TSN supported (0xc000)
+              Parameter length: 4
+          Supported Extensions parameter (Supported types: FORWARD_TSN, AUTH,
+            ASCONF, ASCONF_ACK, RE_CONFIG) Parameter type: Supported Extensions
+  (0x8008) Parameter length: 9 Supported chunk type: FORWARD_TSN (192) Supported
+  chunk type: AUTH (15) Supported chunk type: ASCONF (193) Supported chunk type:
+  ASCONF_ACK (128) Supported chunk type: RE_CONFIG (130) Parameter padding:
+  000000 Random parameter Parameter type: Random (0x8002) Parameter length: 36
+              Random number: c5a86155090e6f420050634cc8d6b908dfd53e17c99cb143…
+          Requested HMAC Algorithm parameter (Supported HMACs: SHA-1)
+              Parameter type: Requested HMAC Algorithm (0x8004)
+              Parameter length: 6
+              HMAC identifier: SHA-1 (1)
+              Parameter padding: 0000
+          Authenticated Chunk list parameter (Chunk types to be authenticated:
+            ASCONF_ACK, ASCONF) Parameter type: Authenticated Chunk list
+  (0x8003) Parameter length: 6 Chunk type: ASCONF_ACK (128) Chunk type: ASCONF
+  (193) Chunk padding: 0000
+  */
+
+  uint8_t data[] = {
+      0x13, 0x88, 0x13, 0x88, 0x00, 0x00, 0x00, 0x00, 0xaa, 0x01, 0x9d, 0x33,
+      0x01, 0x00, 0x00, 0x5a, 0x0e, 0xdd, 0xca, 0x08, 0x00, 0x02, 0x00, 0x00,
+      0x03, 0xe8, 0x03, 0xe8, 0x55, 0x08, 0x36, 0x37, 0x80, 0x00, 0x00, 0x04,
+      0xc0, 0x00, 0x00, 0x04, 0x80, 0x08, 0x00, 0x09, 0xc0, 0x0f, 0xc1, 0x80,
+      0x82, 0x00, 0x00, 0x00, 0x80, 0x02, 0x00, 0x24, 0xc5, 0xa8, 0x61, 0x55,
+      0x09, 0x0e, 0x6f, 0x42, 0x00, 0x50, 0x63, 0x4c, 0xc8, 0xd6, 0xb9, 0x08,
+      0xdf, 0xd5, 0x3e, 0x17, 0xc9, 0x9c, 0xb1, 0x43, 0x28, 0x4e, 0xaf, 0x64,
+      0x68, 0x2a, 0xc2, 0x97, 0x80, 0x04, 0x00, 0x06, 0x00, 0x01, 0x00, 0x00,
+      0x80, 0x03, 0x00, 0x06, 0x80, 0xc1, 0x00, 0x00};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet, SctpPacket::Parse(data));
+  EXPECT_EQ(packet.common_header().source_port, 5000);
+  EXPECT_EQ(packet.common_header().destination_port, 5000);
+  EXPECT_EQ(packet.common_header().verification_tag, VerificationTag(0));
+  EXPECT_EQ(packet.common_header().checksum, 0xaa019d33);
+
+  EXPECT_THAT(packet.descriptors(), SizeIs(1));
+  EXPECT_EQ(packet.descriptors()[0].type, InitChunk::kType);
+  ASSERT_HAS_VALUE_AND_ASSIGN(InitChunk init,
+                              InitChunk::Parse(packet.descriptors()[0].data));
+  EXPECT_EQ(init.initial_tsn(), TSN(1426601527));
+}
+
+TEST(SctpPacketTest, DeserializePacketWithTwoChunks) {
+  /*
+  Stream Control Transmission Protocol, Src Port: 1234 (1234),
+    Dst Port: 4321 (4321)
+      Source port: 1234
+      Destination port: 4321
+      Verification tag: 0x697e3a4e
+      [Association index: 3]
+      Checksum: 0xc06e8b36 [unverified]
+      [Checksum Status: Unverified]
+      COOKIE_ACK chunk
+          Chunk type: COOKIE_ACK (11)
+          Chunk flags: 0x00
+          Chunk length: 4
+      SACK chunk (Cumulative TSN: 2930332242, a_rwnd: 131072,
+        gaps: 0, duplicate TSNs: 0)
+          Chunk type: SACK (3)
+          Chunk flags: 0x00
+          Chunk length: 16
+          Cumulative TSN ACK: 2930332242
+          Advertised receiver window credit (a_rwnd): 131072
+          Number of gap acknowledgement blocks: 0
+          Number of duplicated TSNs: 0
+  */
+
+  uint8_t data[] = {0x04, 0xd2, 0x10, 0xe1, 0x69, 0x7e, 0x3a, 0x4e,
+                    0xc0, 0x6e, 0x8b, 0x36, 0x0b, 0x00, 0x00, 0x04,
+                    0x03, 0x00, 0x00, 0x10, 0xae, 0xa9, 0x52, 0x52,
+                    0x00, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet, SctpPacket::Parse(data));
+  EXPECT_EQ(packet.common_header().source_port, 1234);
+  EXPECT_EQ(packet.common_header().destination_port, 4321);
+  EXPECT_EQ(packet.common_header().verification_tag,
+            VerificationTag(0x697e3a4eu));
+  EXPECT_EQ(packet.common_header().checksum, 0xc06e8b36u);
+
+  EXPECT_THAT(packet.descriptors(), SizeIs(2));
+  EXPECT_EQ(packet.descriptors()[0].type, CookieAckChunk::kType);
+  EXPECT_EQ(packet.descriptors()[1].type, SackChunk::kType);
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      CookieAckChunk cookie_ack,
+      CookieAckChunk::Parse(packet.descriptors()[0].data));
+  ASSERT_HAS_VALUE_AND_ASSIGN(SackChunk sack,
+                              SackChunk::Parse(packet.descriptors()[1].data));
+}
+
+TEST(SctpPacketTest, DeserializePacketWithWrongChecksum) {
+  /*
+  Stream Control Transmission Protocol, Src Port: 5000 (5000),
+    Dst Port: 5000 (5000)
+      Source port: 5000
+      Destination port: 5000
+      Verification tag: 0x0eddca08
+      [Association index: 1]
+      Checksum: 0x2a81f531 [unverified]
+      [Checksum Status: Unverified]
+      SACK chunk (Cumulative TSN: 1426601536, a_rwnd: 131072,
+        gaps: 0, duplicate TSNs: 0)
+          Chunk type: SACK (3)
+          Chunk flags: 0x00
+          Chunk length: 16
+          Cumulative TSN ACK: 1426601536
+          Advertised receiver window credit (a_rwnd): 131072
+          Number of gap acknowledgement blocks: 0
+          Number of duplicated TSNs: 0
+  */
+
+  uint8_t data[] = {0x13, 0x88, 0x13, 0x88, 0x0e, 0xdd, 0xca, 0x08, 0x2a, 0x81,
+                    0xf5, 0x31, 0x03, 0x00, 0x00, 0x10, 0x55, 0x08, 0x36, 0x40,
+                    0x00, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
+
+  EXPECT_FALSE(SctpPacket::Parse(data).has_value());
+}
+
+TEST(SctpPacketTest, DeserializePacketDontValidateChecksum) {
+  /*
+  Stream Control Transmission Protocol, Src Port: 5000 (5000),
+    Dst Port: 5000 (5000)
+      Source port: 5000
+      Destination port: 5000
+      Verification tag: 0x0eddca08
+      [Association index: 1]
+      Checksum: 0x2a81f531 [unverified]
+      [Checksum Status: Unverified]
+      SACK chunk (Cumulative TSN: 1426601536, a_rwnd: 131072,
+        gaps: 0, duplicate TSNs: 0)
+          Chunk type: SACK (3)
+          Chunk flags: 0x00
+          Chunk length: 16
+          Cumulative TSN ACK: 1426601536
+          Advertised receiver window credit (a_rwnd): 131072
+          Number of gap acknowledgement blocks: 0
+          Number of duplicated TSNs: 0
+  */
+
+  uint8_t data[] = {0x13, 0x88, 0x13, 0x88, 0x0e, 0xdd, 0xca, 0x08, 0x2a, 0x81,
+                    0xf5, 0x31, 0x03, 0x00, 0x00, 0x10, 0x55, 0x08, 0x36, 0x40,
+                    0x00, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      SctpPacket packet,
+      SctpPacket::Parse(data, /*disable_checksum_verification=*/true));
+  EXPECT_EQ(packet.common_header().source_port, 5000);
+  EXPECT_EQ(packet.common_header().destination_port, 5000);
+  EXPECT_EQ(packet.common_header().verification_tag,
+            VerificationTag(0x0eddca08u));
+  EXPECT_EQ(packet.common_header().checksum, 0x2a81f531u);
+}
+
+TEST(SctpPacketTest, SerializeAndDeserializeSingleChunk) {
+  SctpPacket::Builder b(kVerificationTag, {});
+  InitChunk init(/*initiate_tag=*/VerificationTag(123), /*a_rwnd=*/456,
+                 /*nbr_outbound_streams=*/65535,
+                 /*nbr_inbound_streams=*/65534, /*initial_tsn=*/TSN(789),
+                 /*parameters=*/Parameters());
+
+  b.Add(init);
+  std::vector<uint8_t> serialized = b.Build();
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet, SctpPacket::Parse(serialized));
+
+  EXPECT_EQ(packet.common_header().verification_tag, kVerificationTag);
+
+  ASSERT_THAT(packet.descriptors(), SizeIs(1));
+  EXPECT_EQ(packet.descriptors()[0].type, InitChunk::kType);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(InitChunk deserialized,
+                              InitChunk::Parse(packet.descriptors()[0].data));
+  EXPECT_EQ(deserialized.initiate_tag(), VerificationTag(123));
+  EXPECT_EQ(deserialized.a_rwnd(), 456u);
+  EXPECT_EQ(deserialized.nbr_outbound_streams(), 65535u);
+  EXPECT_EQ(deserialized.nbr_inbound_streams(), 65534u);
+  EXPECT_EQ(deserialized.initial_tsn(), TSN(789));
+}
+
+TEST(SctpPacketTest, SerializeAndDeserializeThreeChunks) {
+  SctpPacket::Builder b(kVerificationTag, {});
+  b.Add(SackChunk(/*cumulative_tsn_ack=*/TSN(999), /*a_rwnd=*/456,
+                  {SackChunk::GapAckBlock(2, 3)},
+                  /*duplicate_tsns=*/{TSN(1), TSN(2), TSN(3)}));
+  b.Add(DataChunk(TSN(123), StreamID(456), SSN(789), PPID(9090),
+                  /*payload=*/{1, 2, 3, 4, 5},
+                  /*options=*/{}));
+  b.Add(DataChunk(TSN(124), StreamID(654), SSN(987), PPID(909),
+                  /*payload=*/{5, 4, 3, 3, 1},
+                  /*options=*/{}));
+
+  std::vector<uint8_t> serialized = b.Build();
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet, SctpPacket::Parse(serialized));
+
+  EXPECT_EQ(packet.common_header().verification_tag, kVerificationTag);
+
+  ASSERT_THAT(packet.descriptors(), SizeIs(3));
+  EXPECT_EQ(packet.descriptors()[0].type, SackChunk::kType);
+  EXPECT_EQ(packet.descriptors()[1].type, DataChunk::kType);
+  EXPECT_EQ(packet.descriptors()[2].type, DataChunk::kType);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SackChunk sack,
+                              SackChunk::Parse(packet.descriptors()[0].data));
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(999));
+  EXPECT_EQ(sack.a_rwnd(), 456u);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(DataChunk data1,
+                              DataChunk::Parse(packet.descriptors()[1].data));
+  EXPECT_EQ(data1.tsn(), TSN(123));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(DataChunk data2,
+                              DataChunk::Parse(packet.descriptors()[2].data));
+  EXPECT_EQ(data2.tsn(), TSN(124));
+}
+
+TEST(SctpPacketTest, ParseAbortWithEmptyCause) {
+  SctpPacket::Builder b(kVerificationTag, {});
+  b.Add(AbortChunk(
+      /*filled_in_verification_tag=*/true,
+      Parameters::Builder().Add(UserInitiatedAbortCause("")).Build()));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet, SctpPacket::Parse(b.Build()));
+
+  EXPECT_EQ(packet.common_header().verification_tag, kVerificationTag);
+
+  ASSERT_THAT(packet.descriptors(), SizeIs(1));
+  EXPECT_EQ(packet.descriptors()[0].type, AbortChunk::kType);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(AbortChunk abort,
+                              AbortChunk::Parse(packet.descriptors()[0].data));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      UserInitiatedAbortCause cause,
+      abort.error_causes().get<UserInitiatedAbortCause>());
+  EXPECT_EQ(cause.upper_layer_abort_reason(), "");
+}
+
+TEST(SctpPacketTest, DetectPacketWithZeroSizeChunk) {
+  uint8_t data[] = {0xff, 0xff, 0xff, 0xff, 0xff, 0x0a, 0x0a, 0x0a, 0x5c,
+                    0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x00, 0x00, 0x00};
+
+  EXPECT_FALSE(SctpPacket::Parse(data, true).has_value());
+}
+
+TEST(SctpPacketTest, ReturnsCorrectSpaceAvailableToStayWithinMTU) {
+  DcSctpOptions options;
+  options.mtu = 1191;
+
+  SctpPacket::Builder builder(VerificationTag(123), options);
+
+  // Chunks will be padded to an even 4 bytes, so the maximum packet size should
+  // be rounded down.
+  const size_t kMaxPacketSize = RoundDownTo4(options.mtu);
+  EXPECT_EQ(kMaxPacketSize, 1188u);
+
+  const size_t kSctpHeaderSize = 12;
+  EXPECT_EQ(builder.bytes_remaining(), kMaxPacketSize - kSctpHeaderSize);
+  EXPECT_EQ(builder.bytes_remaining(), 1176u);
+
+  // Add a smaller packet first.
+  DataChunk::Options data_options;
+
+  std::vector<uint8_t> payload1(183);
+  builder.Add(
+      DataChunk(TSN(1), StreamID(1), SSN(0), PPID(53), payload1, data_options));
+
+  size_t chunk1_size = RoundUpTo4(DataChunk::kHeaderSize + payload1.size());
+  EXPECT_EQ(builder.bytes_remaining(),
+            kMaxPacketSize - kSctpHeaderSize - chunk1_size);
+  EXPECT_EQ(builder.bytes_remaining(), 976u);  // Hand-calculated.
+
+  std::vector<uint8_t> payload2(957);
+  builder.Add(
+      DataChunk(TSN(1), StreamID(1), SSN(0), PPID(53), payload2, data_options));
+
+  size_t chunk2_size = RoundUpTo4(DataChunk::kHeaderSize + payload2.size());
+  EXPECT_EQ(builder.bytes_remaining(),
+            kMaxPacketSize - kSctpHeaderSize - chunk1_size - chunk2_size);
+  EXPECT_EQ(builder.bytes_remaining(), 0u);  // Hand-calculated.
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/tlv_trait.cc b/third_party/libwebrtc/net/dcsctp/packet/tlv_trait.cc
new file mode 100644
index 0000000000..493b6a4613
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/tlv_trait.cc
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/tlv_trait.h"
+
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+namespace tlv_trait_impl {
+void ReportInvalidSize(size_t actual_size, size_t expected_size) {
+  RTC_DLOG(LS_WARNING) << "Invalid size (" << actual_size
+                       << ", expected minimum " << expected_size << " bytes)";
+}
+
+void ReportInvalidType(int actual_type, int expected_type) {
+  RTC_DLOG(LS_WARNING) << "Invalid type (" << actual_type << ", expected "
+                       << expected_type << ")";
+}
+
+void ReportInvalidFixedLengthField(size_t value, size_t expected) {
+  RTC_DLOG(LS_WARNING) << "Invalid length field (" << value << ", expected "
+                       << expected << " bytes)";
+}
+
+void ReportInvalidVariableLengthField(size_t value, size_t available) {
+  RTC_DLOG(LS_WARNING) << "Invalid length field (" << value << ", available "
+                       << available << " bytes)";
+}
+
+void ReportInvalidPadding(size_t padding_bytes) {
+  RTC_DLOG(LS_WARNING) << "Invalid padding (" << padding_bytes << " bytes)";
+}
+
+void ReportInvalidLengthMultiple(size_t length, size_t alignment) {
+  RTC_DLOG(LS_WARNING) << "Invalid length field (" << length
+                       << ", expected an even multiple of " << alignment
+                       << " bytes)";
+}
+}  // namespace tlv_trait_impl
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/packet/tlv_trait.h b/third_party/libwebrtc/net/dcsctp/packet/tlv_trait.h
new file mode 100644
index 0000000000..a3c728efd7
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/tlv_trait.h
@@ -0,0 +1,165 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PACKET_TLV_TRAIT_H_
+#define NET_DCSCTP_PACKET_TLV_TRAIT_H_
+
+#include <stdint.h>
+#include <string.h>
+
+#include <algorithm>
+#include <cstddef>
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+
+namespace dcsctp {
+namespace tlv_trait_impl {
+// Logging functions, only to be used by TLVTrait, which is a templated class.
+void ReportInvalidSize(size_t actual_size, size_t expected_size);
+void ReportInvalidType(int actual_type, int expected_type);
+void ReportInvalidFixedLengthField(size_t value, size_t expected);
+void ReportInvalidVariableLengthField(size_t value, size_t available);
+void ReportInvalidPadding(size_t padding_bytes);
+void ReportInvalidLengthMultiple(size_t length, size_t alignment);
+}  // namespace tlv_trait_impl
+
+// Various entities in SCTP are padded data blocks, with a type and length
+// field at fixed offsets, all stored in a 4-byte header.
+//
+// See e.g. https://tools.ietf.org/html/rfc4960#section-3.2 and
+// https://tools.ietf.org/html/rfc4960#section-3.2.1
+//
+// These are helper classes for writing and parsing that data, which in SCTP is
+// called Type-Length-Value, or TLV.
+//
+// This templated class is configurable - a struct passed in as template
+// parameter with the following expected members:
+//   * kType                    - The type field's value
+//   * kTypeSizeInBytes         - The type field's width in bytes.
+//                                Either 1 or 2.
+//   * kHeaderSize              - The fixed size header
+//   * kVariableLengthAlignment - The size alignment on the variable data. Set
+//                                to zero (0) if no variable data is used.
+//
+// This class is to be used as a trait
+// (https://en.wikipedia.org/wiki/Trait_(computer_programming)) that adds a few
+// public and protected members and which a class inherits from when it
+// represents a type-length-value object.
+template <typename Config>
+class TLVTrait {
+ private:
+  static constexpr size_t kTlvHeaderSize = 4;
+
+ protected:
+  static constexpr size_t kHeaderSize = Config::kHeaderSize;
+
+  static_assert(Config::kTypeSizeInBytes == 1 || Config::kTypeSizeInBytes == 2,
+                "kTypeSizeInBytes must be 1 or 2");
+  static_assert(Config::kHeaderSize >= kTlvHeaderSize,
+                "HeaderSize must be >= 4 bytes");
+  static_assert((Config::kHeaderSize % 4 == 0),
+                "kHeaderSize must be an even multiple of 4 bytes");
+  static_assert((Config::kVariableLengthAlignment == 0 ||
+                 Config::kVariableLengthAlignment == 1 ||
+                 Config::kVariableLengthAlignment == 2 ||
+                 Config::kVariableLengthAlignment == 4 ||
+                 Config::kVariableLengthAlignment == 8),
+                "kVariableLengthAlignment must be an allowed value");
+
+  // Validates the data with regards to size, alignment and type.
+  // If valid, returns a bounded buffer.
+  static absl::optional<BoundedByteReader<Config::kHeaderSize>> ParseTLV(
+      rtc::ArrayView<const uint8_t> data) {
+    if (data.size() < Config::kHeaderSize) {
+      tlv_trait_impl::ReportInvalidSize(data.size(), Config::kHeaderSize);
+      return absl::nullopt;
+    }
+    BoundedByteReader<kTlvHeaderSize> tlv_header(data);
+
+    const int type = (Config::kTypeSizeInBytes == 1)
+                         ? tlv_header.template Load8<0>()
+                         : tlv_header.template Load16<0>();
+
+    if (type != Config::kType) {
+      tlv_trait_impl::ReportInvalidType(type, Config::kType);
+      return absl::nullopt;
+    }
+    const uint16_t length = tlv_header.template Load16<2>();
+    if (Config::kVariableLengthAlignment == 0) {
+      // Don't expect any variable length data at all.
+      if (length != Config::kHeaderSize || data.size() != Config::kHeaderSize) {
+        tlv_trait_impl::ReportInvalidFixedLengthField(length,
+                                                      Config::kHeaderSize);
+        return absl::nullopt;
+      }
+    } else {
+      // Expect variable length data - verify its size alignment.
+      if (length > data.size() || length < Config::kHeaderSize) {
+        tlv_trait_impl::ReportInvalidVariableLengthField(length, data.size());
+        return absl::nullopt;
+      }
+      const size_t padding = data.size() - length;
+      if (padding > 3) {
+        // https://tools.ietf.org/html/rfc4960#section-3.2
+        // "This padding MUST NOT be more than 3 bytes in total"
+        tlv_trait_impl::ReportInvalidPadding(padding);
+        return absl::nullopt;
+      }
+      if (!ValidateLengthAlignment(length, Config::kVariableLengthAlignment)) {
+        tlv_trait_impl::ReportInvalidLengthMultiple(
+            length, Config::kVariableLengthAlignment);
+        return absl::nullopt;
+      }
+    }
+    return BoundedByteReader<Config::kHeaderSize>(data.subview(0, length));
+  }
+
+  // Allocates space for data with a static header size, as defined by
+  // `Config::kHeaderSize` and a variable footer, as defined by `variable_size`
+  // (which may be 0) and writes the type and length in the header.
+  static BoundedByteWriter<Config::kHeaderSize> AllocateTLV(
+      std::vector<uint8_t>& out,
+      size_t variable_size = 0) {
+    const size_t offset = out.size();
+    const size_t size = Config::kHeaderSize + variable_size;
+    out.resize(offset + size);
+
+    BoundedByteWriter<kTlvHeaderSize> tlv_header(
+        rtc::ArrayView<uint8_t>(out.data() + offset, kTlvHeaderSize));
+    if (Config::kTypeSizeInBytes == 1) {
+      tlv_header.template Store8<0>(static_cast<uint8_t>(Config::kType));
+    } else {
+      tlv_header.template Store16<0>(Config::kType);
+    }
+    tlv_header.template Store16<2>(size);
+
+    return BoundedByteWriter<Config::kHeaderSize>(
+        rtc::ArrayView<uint8_t>(out.data() + offset, size));
+  }
+
+ private:
+  static bool ValidateLengthAlignment(uint16_t length, size_t alignment) {
+    // This is to avoid MSVC believing there could be a "mod by zero", when it
+    // certainly can't.
+    if (alignment == 0) {
+      return true;
+    }
+    return (length % alignment) == 0;
+  }
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PACKET_TLV_TRAIT_H_
diff --git a/third_party/libwebrtc/net/dcsctp/packet/tlv_trait_test.cc b/third_party/libwebrtc/net/dcsctp/packet/tlv_trait_test.cc
new file mode 100644
index 0000000000..a0dd1a1136
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/packet/tlv_trait_test.cc
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/tlv_trait.h"
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+using ::testing::SizeIs;
+
+struct OneByteTypeConfig {
+  static constexpr int kTypeSizeInBytes = 1;
+  static constexpr int kType = 0x49;
+  static constexpr size_t kHeaderSize = 12;
+  static constexpr int kVariableLengthAlignment = 4;
+};
+
+class OneByteChunk : public TLVTrait<OneByteTypeConfig> {
+ public:
+  static constexpr size_t kVariableSize = 4;
+
+  void SerializeTo(std::vector<uint8_t>& out) {
+    BoundedByteWriter<OneByteTypeConfig::kHeaderSize> writer =
+        AllocateTLV(out, kVariableSize);
+    writer.Store32<4>(0x01020304);
+    writer.Store16<8>(0x0506);
+    writer.Store16<10>(0x0708);
+
+    uint8_t variable_data[kVariableSize] = {0xDE, 0xAD, 0xBE, 0xEF};
+    writer.CopyToVariableData(rtc::ArrayView<const uint8_t>(variable_data));
+  }
+
+  static absl::optional<BoundedByteReader<OneByteTypeConfig::kHeaderSize>>
+  Parse(rtc::ArrayView<const uint8_t> data) {
+    return ParseTLV(data);
+  }
+};
+
+TEST(TlvDataTest, CanWriteOneByteTypeTlvs) {
+  std::vector<uint8_t> out;
+  OneByteChunk().SerializeTo(out);
+
+  EXPECT_THAT(out, SizeIs(OneByteTypeConfig::kHeaderSize +
+                          OneByteChunk::kVariableSize));
+  EXPECT_THAT(out, ElementsAre(0x49, 0x00, 0x00, 0x10, 0x01, 0x02, 0x03, 0x04,
+                               0x05, 0x06, 0x07, 0x08, 0xDE, 0xAD, 0xBE, 0xEF));
+}
+
+TEST(TlvDataTest, CanReadOneByteTypeTlvs) {
+  uint8_t data[] = {0x49, 0x00, 0x00, 0x10, 0x01, 0x02, 0x03, 0x04,
+                    0x05, 0x06, 0x07, 0x08, 0xDE, 0xAD, 0xBE, 0xEF};
+
+  absl::optional<BoundedByteReader<OneByteTypeConfig::kHeaderSize>> reader =
+      OneByteChunk::Parse(data);
+  ASSERT_TRUE(reader.has_value());
+  EXPECT_EQ(reader->Load32<4>(), 0x01020304U);
+  EXPECT_EQ(reader->Load16<8>(), 0x0506U);
+  EXPECT_EQ(reader->Load16<10>(), 0x0708U);
+  EXPECT_THAT(reader->variable_data(), ElementsAre(0xDE, 0xAD, 0xBE, 0xEF));
+}
+
+struct TwoByteTypeConfig {
+  static constexpr int kTypeSizeInBytes = 2;
+  static constexpr int kType = 31337;
+  static constexpr size_t kHeaderSize = 8;
+  static constexpr int kVariableLengthAlignment = 2;
+};
+
+class TwoByteChunk : public TLVTrait<TwoByteTypeConfig> {
+ public:
+  static constexpr size_t kVariableSize = 8;
+
+  void SerializeTo(std::vector<uint8_t>& out) {
+    BoundedByteWriter<TwoByteTypeConfig::kHeaderSize> writer =
+        AllocateTLV(out, kVariableSize);
+    writer.Store32<4>(0x01020304U);
+
+    uint8_t variable_data[] = {0x05, 0x06, 0x07, 0x08, 0xDE, 0xAD, 0xBE, 0xEF};
+    writer.CopyToVariableData(rtc::ArrayView<const uint8_t>(variable_data));
+  }
+
+  static absl::optional<BoundedByteReader<TwoByteTypeConfig::kHeaderSize>>
+  Parse(rtc::ArrayView<const uint8_t> data) {
+    return ParseTLV(data);
+  }
+};
+
+TEST(TlvDataTest, CanWriteTwoByteTypeTlvs) {
+  std::vector<uint8_t> out;
+
+  TwoByteChunk().SerializeTo(out);
+
+  EXPECT_THAT(out, SizeIs(TwoByteTypeConfig::kHeaderSize +
+                          TwoByteChunk::kVariableSize));
+  EXPECT_THAT(out, ElementsAre(0x7A, 0x69, 0x00, 0x10, 0x01, 0x02, 0x03, 0x04,
+                               0x05, 0x06, 0x07, 0x08, 0xDE, 0xAD, 0xBE, 0xEF));
+}
+
+TEST(TlvDataTest, CanReadTwoByteTypeTlvs) {
+  uint8_t data[] = {0x7A, 0x69, 0x00, 0x10, 0x01, 0x02, 0x03, 0x04,
+                    0x05, 0x06, 0x07, 0x08, 0xDE, 0xAD, 0xBE, 0xEF};
+
+  absl::optional<BoundedByteReader<TwoByteTypeConfig::kHeaderSize>> reader =
+      TwoByteChunk::Parse(data);
+  EXPECT_TRUE(reader.has_value());
+  EXPECT_EQ(reader->Load32<4>(), 0x01020304U);
+  EXPECT_THAT(reader->variable_data(),
+              ElementsAre(0x05, 0x06, 0x07, 0x08, 0xDE, 0xAD, 0xBE, 0xEF));
+}
+
+TEST(TlvDataTest, CanHandleInvalidLengthSmallerThanFixedSize) {
+  // Has 'length=6', which is below the kHeaderSize of 8.
+  uint8_t data[] = {0x7A, 0x69, 0x00, 0x06, 0x01, 0x02, 0x03, 0x04};
+
+  EXPECT_FALSE(TwoByteChunk::Parse(data).has_value());
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/public/BUILD.gn b/third_party/libwebrtc/net/dcsctp/public/BUILD.gn
new file mode 100644
index 0000000000..6cb289bf5b
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/BUILD.gn
@@ -0,0 +1,103 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("types") {
+  deps = [
+    "../../../api:array_view",
+    "../../../rtc_base:strong_alias",
+  ]
+  sources = [
+    "dcsctp_message.h",
+    "dcsctp_options.h",
+    "types.h",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_source_set("socket") {
+  deps = [
+    ":types",
+    "../../../api:array_view",
+    "../../../api/task_queue:task_queue",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:strong_alias",
+  ]
+  sources = [
+    "dcsctp_handover_state.cc",
+    "dcsctp_handover_state.h",
+    "dcsctp_socket.h",
+    "packet_observer.h",
+    "timeout.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_source_set("factory") {
+  deps = [
+    ":socket",
+    ":types",
+    "../socket:dcsctp_socket",
+  ]
+  sources = [
+    "dcsctp_socket_factory.cc",
+    "dcsctp_socket_factory.h",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_source_set("mocks") {
+  testonly = true
+  sources = [
+    "mock_dcsctp_socket.h",
+    "mock_dcsctp_socket_factory.h",
+  ]
+  deps = [
+    ":factory",
+    ":socket",
+    "../../../test:test_support",
+  ]
+}
+
+rtc_source_set("utils") {
+  deps = [
+    ":socket",
+    ":types",
+    "../../../api:array_view",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:stringutils",
+    "../socket:dcsctp_socket",
+  ]
+  sources = [
+    "text_pcap_packet_observer.cc",
+    "text_pcap_packet_observer.h",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("dcsctp_public_unittests") {
+    testonly = true
+
+    deps = [
+      ":mocks",
+      ":types",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:gunit_helpers",
+      "../../../test:test_support",
+    ]
+    sources = [
+      "mock_dcsctp_socket_test.cc",
+      "types_test.cc",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/net/dcsctp/public/dcsctp_handover_state.cc b/third_party/libwebrtc/net/dcsctp/public/dcsctp_handover_state.cc
new file mode 100644
index 0000000000..6a1bd06eba
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/dcsctp_handover_state.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/public/dcsctp_handover_state.h"
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace dcsctp {
+namespace {
+constexpr absl::string_view HandoverUnreadinessReasonToString(
+    HandoverUnreadinessReason reason) {
+  switch (reason) {
+    case HandoverUnreadinessReason::kWrongConnectionState:
+      return "WRONG_CONNECTION_STATE";
+    case HandoverUnreadinessReason::kSendQueueNotEmpty:
+      return "SEND_QUEUE_NOT_EMPTY";
+    case HandoverUnreadinessReason::kDataTrackerTsnBlocksPending:
+      return "DATA_TRACKER_TSN_BLOCKS_PENDING";
+    case HandoverUnreadinessReason::kReassemblyQueueDeliveredTSNsGap:
+      return "REASSEMBLY_QUEUE_DELIVERED_TSN_GAP";
+    case HandoverUnreadinessReason::kStreamResetDeferred:
+      return "STREAM_RESET_DEFERRED";
+    case HandoverUnreadinessReason::kOrderedStreamHasUnassembledChunks:
+      return "ORDERED_STREAM_HAS_UNASSEMBLED_CHUNKS";
+    case HandoverUnreadinessReason::kUnorderedStreamHasUnassembledChunks:
+      return "UNORDERED_STREAM_HAS_UNASSEMBLED_CHUNKS";
+    case HandoverUnreadinessReason::kRetransmissionQueueOutstandingData:
+      return "RETRANSMISSION_QUEUE_OUTSTANDING_DATA";
+    case HandoverUnreadinessReason::kRetransmissionQueueFastRecovery:
+      return "RETRANSMISSION_QUEUE_FAST_RECOVERY";
+    case HandoverUnreadinessReason::kRetransmissionQueueNotEmpty:
+      return "RETRANSMISSION_QUEUE_NOT_EMPTY";
+    case HandoverUnreadinessReason::kPendingStreamReset:
+      return "PENDING_STREAM_RESET";
+    case HandoverUnreadinessReason::kPendingStreamResetRequest:
+      return "PENDING_STREAM_RESET_REQUEST";
+  }
+}
+}  // namespace
+
+std::string HandoverReadinessStatus::ToString() const {
+  std::string result;
+  for (uint32_t bit = 1;
+       bit <= static_cast<uint32_t>(HandoverUnreadinessReason::kMax);
+       bit *= 2) {
+    auto flag = static_cast<HandoverUnreadinessReason>(bit);
+    if (Contains(flag)) {
+      if (!result.empty()) {
+        result.append(",");
+      }
+      absl::string_view s = HandoverUnreadinessReasonToString(flag);
+      result.append(s.data(), s.size());
+    }
+  }
+  if (result.empty()) {
+    result = "READY";
+  }
+  return result;
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/public/dcsctp_handover_state.h b/third_party/libwebrtc/net/dcsctp/public/dcsctp_handover_state.h
new file mode 100644
index 0000000000..253f4da939
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/dcsctp_handover_state.h
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_DCSCTP_HANDOVER_STATE_H_
+#define NET_DCSCTP_PUBLIC_DCSCTP_HANDOVER_STATE_H_
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "rtc_base/strong_alias.h"
+
+namespace dcsctp {
+
+// Stores state snapshot of a dcSCTP socket. The snapshot can be used to
+// recreate the socket - possibly in another process. This state should be
+// treaded as opaque - the calling client should not inspect or alter it except
+// for serialization. Serialization is not provided by dcSCTP. If needed it has
+// to be implemented in the calling client.
+struct DcSctpSocketHandoverState {
+  enum class SocketState {
+    kClosed,
+    kConnected,
+  };
+  SocketState socket_state = SocketState::kClosed;
+
+  uint32_t my_verification_tag = 0;
+  uint32_t my_initial_tsn = 0;
+  uint32_t peer_verification_tag = 0;
+  uint32_t peer_initial_tsn = 0;
+  uint64_t tie_tag = 0;
+
+  struct Capabilities {
+    bool partial_reliability = false;
+    bool message_interleaving = false;
+    bool reconfig = false;
+    uint16_t negotiated_maximum_incoming_streams = 0;
+    uint16_t negotiated_maximum_outgoing_streams = 0;
+  };
+  Capabilities capabilities;
+
+  struct OutgoingStream {
+    uint32_t id = 0;
+    uint32_t next_ssn = 0;
+    uint32_t next_unordered_mid = 0;
+    uint32_t next_ordered_mid = 0;
+    uint16_t priority = 0;
+  };
+  struct Transmission {
+    uint32_t next_tsn = 0;
+    uint32_t next_reset_req_sn = 0;
+    uint32_t cwnd = 0;
+    uint32_t rwnd = 0;
+    uint32_t ssthresh = 0;
+    uint32_t partial_bytes_acked = 0;
+    std::vector<OutgoingStream> streams;
+  };
+  Transmission tx;
+
+  struct OrderedStream {
+    uint32_t id = 0;
+    uint32_t next_ssn = 0;
+  };
+  struct UnorderedStream {
+    uint32_t id = 0;
+  };
+  struct Receive {
+    bool seen_packet = false;
+    uint32_t last_cumulative_acked_tsn = 0;
+    uint32_t last_assembled_tsn = 0;
+    uint32_t last_completed_deferred_reset_req_sn = 0;
+    uint32_t last_completed_reset_req_sn = 0;
+    std::vector<OrderedStream> ordered_streams;
+    std::vector<UnorderedStream> unordered_streams;
+  };
+  Receive rx;
+};
+
+// A list of possible reasons for a socket to be not ready for handover.
+enum class HandoverUnreadinessReason : uint32_t {
+  kWrongConnectionState = 1,
+  kSendQueueNotEmpty = 2,
+  kPendingStreamResetRequest = 4,
+  kDataTrackerTsnBlocksPending = 8,
+  kPendingStreamReset = 16,
+  kReassemblyQueueDeliveredTSNsGap = 32,
+  kStreamResetDeferred = 64,
+  kOrderedStreamHasUnassembledChunks = 128,
+  kUnorderedStreamHasUnassembledChunks = 256,
+  kRetransmissionQueueOutstandingData = 512,
+  kRetransmissionQueueFastRecovery = 1024,
+  kRetransmissionQueueNotEmpty = 2048,
+  kMax = kRetransmissionQueueNotEmpty,
+};
+
+// Return value of `DcSctpSocketInterface::GetHandoverReadiness`. Set of
+// `HandoverUnreadinessReason` bits. When no bit is set, the socket is in the
+// state in which a snapshot of the state can be made by
+// `GetHandoverStateAndClose()`.
+class HandoverReadinessStatus
+    : public webrtc::StrongAlias<class HandoverReadinessStatusTag, uint32_t> {
+ public:
+  // Constructs an empty `HandoverReadinessStatus` which represents ready state.
+  constexpr HandoverReadinessStatus()
+      : webrtc::StrongAlias<class HandoverReadinessStatusTag, uint32_t>(0) {}
+  // Constructs status object that contains a single reason for not being
+  // handover ready.
+  constexpr explicit HandoverReadinessStatus(HandoverUnreadinessReason reason)
+      : webrtc::StrongAlias<class HandoverReadinessStatusTag, uint32_t>(
+            static_cast<uint32_t>(reason)) {}
+
+  // Convenience methods
+  constexpr bool IsReady() const { return value() == 0; }
+  constexpr bool Contains(HandoverUnreadinessReason reason) const {
+    return value() & static_cast<uint32_t>(reason);
+  }
+  HandoverReadinessStatus& Add(HandoverUnreadinessReason reason) {
+    return Add(HandoverReadinessStatus(reason));
+  }
+  HandoverReadinessStatus& Add(HandoverReadinessStatus status) {
+    value() |= status.value();
+    return *this;
+  }
+  std::string ToString() const;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PUBLIC_DCSCTP_HANDOVER_STATE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/dcsctp_message.h b/third_party/libwebrtc/net/dcsctp/public/dcsctp_message.h
new file mode 100644
index 0000000000..38e6763916
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/dcsctp_message.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_DCSCTP_MESSAGE_H_
+#define NET_DCSCTP_PUBLIC_DCSCTP_MESSAGE_H_
+
+#include <cstdint>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// An SCTP message is a group of bytes sent and received as a whole on a
+// specified stream identifier (`stream_id`), and with a payload protocol
+// identifier (`ppid`).
+class DcSctpMessage {
+ public:
+  DcSctpMessage(StreamID stream_id, PPID ppid, std::vector<uint8_t> payload)
+      : stream_id_(stream_id), ppid_(ppid), payload_(std::move(payload)) {}
+
+  DcSctpMessage(DcSctpMessage&& other) = default;
+  DcSctpMessage& operator=(DcSctpMessage&& other) = default;
+  DcSctpMessage(const DcSctpMessage&) = delete;
+  DcSctpMessage& operator=(const DcSctpMessage&) = delete;
+
+  // The stream identifier to which the message is sent.
+  StreamID stream_id() const { return stream_id_; }
+
+  // The payload protocol identifier (ppid) associated with the message.
+  PPID ppid() const { return ppid_; }
+
+  // The payload of the message.
+  rtc::ArrayView<const uint8_t> payload() const { return payload_; }
+
+  // When destructing the message, extracts the payload.
+  std::vector<uint8_t> ReleasePayload() && { return std::move(payload_); }
+
+ private:
+  StreamID stream_id_;
+  PPID ppid_;
+  std::vector<uint8_t> payload_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PUBLIC_DCSCTP_MESSAGE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/dcsctp_options.h b/third_party/libwebrtc/net/dcsctp/public/dcsctp_options.h
new file mode 100644
index 0000000000..4511bed4a4
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/dcsctp_options.h
@@ -0,0 +1,201 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_DCSCTP_OPTIONS_H_
+#define NET_DCSCTP_PUBLIC_DCSCTP_OPTIONS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+struct DcSctpOptions {
+  // The largest safe SCTP packet. Starting from the minimum guaranteed MTU
+  // value of 1280 for IPv6 (which may not support fragmentation), take off 85
+  // bytes for DTLS/TURN/TCP/IP and ciphertext overhead.
+  //
+  // Additionally, it's possible that TURN adds an additional 4 bytes of
+  // overhead after a channel has been established, so an additional 4 bytes is
+  // subtracted
+  //
+  // 1280 IPV6 MTU
+  //  -40 IPV6 header
+  //   -8 UDP
+  //  -24 GCM Cipher
+  //  -13 DTLS record header
+  //   -4 TURN ChannelData
+  // = 1191 bytes.
+  static constexpr size_t kMaxSafeMTUSize = 1191;
+
+  // The local port for which the socket is supposed to be bound to. Incoming
+  // packets will be verified that they are sent to this port number and all
+  // outgoing packets will have this port number as source port.
+  int local_port = 5000;
+
+  // The remote port to send packets to. All outgoing packets will have this
+  // port number as destination port.
+  int remote_port = 5000;
+
+  // The announced maximum number of incoming streams. Note that this value is
+  // constant and can't be currently increased in run-time as "Add Incoming
+  // Streams Request" in RFC6525 isn't supported.
+  //
+  // The socket implementation doesn't have any per-stream fixed costs, which is
+  // why the default value is set to be the maximum value.
+  uint16_t announced_maximum_incoming_streams = 65535;
+
+  // The announced maximum number of outgoing streams. Note that this value is
+  // constant and can't be currently increased in run-time as "Add Outgoing
+  // Streams Request" in RFC6525 isn't supported.
+  //
+  // The socket implementation doesn't have any per-stream fixed costs, which is
+  // why the default value is set to be the maximum value.
+  uint16_t announced_maximum_outgoing_streams = 65535;
+
+  // Maximum SCTP packet size. The library will limit the size of generated
+  // packets to be less than or equal to this number. This does not include any
+  // overhead of DTLS, TURN, UDP or IP headers.
+  size_t mtu = kMaxSafeMTUSize;
+
+  // The largest allowed message payload to be sent. Messages will be rejected
+  // if their payload is larger than this value. Note that this doesn't affect
+  // incoming messages, which may larger than this value (but smaller than
+  // `max_receiver_window_buffer_size`).
+  size_t max_message_size = 256 * 1024;
+
+  // The default stream priority, if not overridden by
+  // `SctpSocket::SetStreamPriority`. The default value is selected to be
+  // compatible with https://www.w3.org/TR/webrtc-priority/, section 4.2-4.3.
+  StreamPriority default_stream_priority = StreamPriority(256);
+
+  // Maximum received window buffer size. This should be a bit larger than the
+  // largest sized message you want to be able to receive. This essentially
+  // limits the memory usage on the receive side. Note that memory is allocated
+  // dynamically, and this represents the maximum amount of buffered data. The
+  // actual memory usage of the library will be smaller in normal operation, and
+  // will be larger than this due to other allocations and overhead if the
+  // buffer is fully utilized.
+  size_t max_receiver_window_buffer_size = 5 * 1024 * 1024;
+
+  // Maximum send buffer size. It will not be possible to queue more data than
+  // this before sending it.
+  size_t max_send_buffer_size = 2'000'000;
+
+  // A threshold that, when the amount of data in the send buffer goes below
+  // this value, will trigger `DcSctpCallbacks::OnTotalBufferedAmountLow`.
+  size_t total_buffered_amount_low_threshold = 1'800'000;
+
+  // Max allowed RTT value. When the RTT is measured and it's found to be larger
+  // than this value, it will be discarded and not used for e.g. any RTO
+  // calculation. The default value is an extreme maximum but can be adapted
+  // to better match the environment.
+  DurationMs rtt_max = DurationMs(60'000);
+
+  // Initial RTO value.
+  DurationMs rto_initial = DurationMs(500);
+
+  // Maximum RTO value.
+  DurationMs rto_max = DurationMs(60'000);
+
+  // Minimum RTO value. This must be larger than an expected peer delayed ack
+  // timeout.
+  DurationMs rto_min = DurationMs(400);
+
+  // T1-init timeout.
+  DurationMs t1_init_timeout = DurationMs(1000);
+
+  // T1-cookie timeout.
+  DurationMs t1_cookie_timeout = DurationMs(1000);
+
+  // T2-shutdown timeout.
+  DurationMs t2_shutdown_timeout = DurationMs(1000);
+
+  // For t1-init, t1-cookie, t2-shutdown, t3-rtx, this value - if set - will be
+  // the upper bound on how large the exponentially backed off timeout can
+  // become. The lower the duration, the faster the connection can recover on
+  // transient network issues. Setting this value may require changing
+  // `max_retransmissions` and `max_init_retransmits` to ensure that the
+  // connection is not closed too quickly.
+  absl::optional<DurationMs> max_timer_backoff_duration = absl::nullopt;
+
+  // Hearbeat interval (on idle connections only). Set to zero to disable.
+  DurationMs heartbeat_interval = DurationMs(30000);
+
+  // The maximum time when a SACK will be sent from the arrival of an
+  // unacknowledged packet. Whatever is smallest of RTO/2 and this will be used.
+  DurationMs delayed_ack_max_timeout = DurationMs(200);
+
+  // The minimum limit for the measured RTT variance
+  //
+  // Setting this below the expected delayed ack timeout (+ margin) of the peer
+  // might result in unnecessary retransmissions, as the maximum time it takes
+  // to ACK a DATA chunk is typically RTT + ATO (delayed ack timeout), and when
+  // the SCTP channel is quite idle, and heartbeats dominate the source of RTT
+  // measurement, the RTO would converge with the smoothed RTT (SRTT). The
+  // default ATO is 200ms in usrsctp, and a 20ms (10%) margin would include the
+  // processing time of received packets and the clock granularity when setting
+  // the delayed ack timer on the peer.
+  //
+  // This is described for TCP in
+  // https://datatracker.ietf.org/doc/html/rfc6298#section-4.
+  DurationMs min_rtt_variance = DurationMs(220);
+
+  // The initial congestion window size, in number of MTUs.
+  // See https://tools.ietf.org/html/rfc4960#section-7.2.1 which defaults at ~3
+  // and https://research.google/pubs/pub36640/ which argues for at least ten
+  // segments.
+  size_t cwnd_mtus_initial = 10;
+
+  // The minimum congestion window size, in number of MTUs, upon detection of
+  // packet loss by SACK. Note that if the retransmission timer expires, the
+  // congestion window will be as small as one MTU. See
+  // https://tools.ietf.org/html/rfc4960#section-7.2.3.
+  size_t cwnd_mtus_min = 4;
+
+  // When the congestion window is at or above this number of MTUs, the
+  // congestion control algorithm will avoid filling the congestion window
+  // fully, if that results in fragmenting large messages into quite small
+  // packets. When the congestion window is smaller than this option, it will
+  // aim to fill the congestion window as much as it can, even if it results in
+  // creating small fragmented packets.
+  size_t avoid_fragmentation_cwnd_mtus = 6;
+
+  // The number of packets that may be sent at once. This is limited to avoid
+  // bursts that too quickly fill the send buffer. Typically in a a socket in
+  // its "slow start" phase (when it sends as much as it can), it will send
+  // up to three packets for every SACK received, so the default limit is set
+  // just above that, and then mostly applicable for (but not limited to) fast
+  // retransmission scenarios.
+  int max_burst = 4;
+
+  // Maximum Data Retransmit Attempts (per DATA chunk). Set to absl::nullopt for
+  // no limit.
+  absl::optional<int> max_retransmissions = 10;
+
+  // Max.Init.Retransmits (https://tools.ietf.org/html/rfc4960#section-15). Set
+  // to absl::nullopt for no limit.
+  absl::optional<int> max_init_retransmits = 8;
+
+  // RFC3758 Partial Reliability Extension
+  bool enable_partial_reliability = true;
+
+  // RFC8260 Stream Schedulers and User Message Interleaving
+  bool enable_message_interleaving = false;
+
+  // If RTO should be added to heartbeat_interval
+  bool heartbeat_interval_include_rtt = true;
+
+  // Disables SCTP packet crc32 verification. Useful when running with fuzzers.
+  bool disable_checksum_verification = false;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PUBLIC_DCSCTP_OPTIONS_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/dcsctp_socket.h b/third_party/libwebrtc/net/dcsctp/public/dcsctp_socket.h
new file mode 100644
index 0000000000..2df6a2c009
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/dcsctp_socket.h
@@ -0,0 +1,617 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_DCSCTP_SOCKET_H_
+#define NET_DCSCTP_PUBLIC_DCSCTP_SOCKET_H_
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/public/dcsctp_handover_state.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/packet_observer.h"
+#include "net/dcsctp/public/timeout.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// The socket/association state
+enum class SocketState {
+  // The socket is closed.
+  kClosed,
+  // The socket has initiated a connection, which is not yet established. Note
+  // that for incoming connections and for reconnections when the socket is
+  // already connected, the socket will not transition to this state.
+  kConnecting,
+  // The socket is connected, and the connection is established.
+  kConnected,
+  // The socket is shutting down, and the connection is not yet closed.
+  kShuttingDown,
+};
+
+// Send options for sending messages
+struct SendOptions {
+  // If the message should be sent with unordered message delivery.
+  IsUnordered unordered = IsUnordered(false);
+
+  // If set, will discard messages that haven't been correctly sent and
+  // received before the lifetime has expired. This is only available if the
+  // peer supports Partial Reliability Extension (RFC3758).
+  absl::optional<DurationMs> lifetime = absl::nullopt;
+
+  // If set, limits the number of retransmissions. This is only available
+  // if the peer supports Partial Reliability Extension (RFC3758).
+  absl::optional<size_t> max_retransmissions = absl::nullopt;
+
+  // If set, will generate lifecycle events for this message. See e.g.
+  // `DcSctpSocketCallbacks::OnLifecycleMessageFullySent`. This value is decided
+  // by the client and the library will provide it to all lifecycle callbacks.
+  LifecycleId lifecycle_id = LifecycleId::NotSet();
+};
+
+enum class ErrorKind {
+  // Indicates that no error has occurred. This will never be the case when
+  // `OnError` or `OnAborted` is called.
+  kNoError,
+  // There have been too many retries or timeouts, and the library has given up.
+  kTooManyRetries,
+  // A command was received that is only possible to execute when the socket is
+  // connected, which it is not.
+  kNotConnected,
+  // Parsing of the command or its parameters failed.
+  kParseFailed,
+  // Commands are received in the wrong sequence, which indicates a
+  // synchronisation mismatch between the peers.
+  kWrongSequence,
+  // The peer has reported an issue using ERROR or ABORT command.
+  kPeerReported,
+  // The peer has performed a protocol violation.
+  kProtocolViolation,
+  // The receive or send buffers have been exhausted.
+  kResourceExhaustion,
+  // The client has performed an invalid operation.
+  kUnsupportedOperation,
+};
+
+inline constexpr absl::string_view ToString(ErrorKind error) {
+  switch (error) {
+    case ErrorKind::kNoError:
+      return "NO_ERROR";
+    case ErrorKind::kTooManyRetries:
+      return "TOO_MANY_RETRIES";
+    case ErrorKind::kNotConnected:
+      return "NOT_CONNECTED";
+    case ErrorKind::kParseFailed:
+      return "PARSE_FAILED";
+    case ErrorKind::kWrongSequence:
+      return "WRONG_SEQUENCE";
+    case ErrorKind::kPeerReported:
+      return "PEER_REPORTED";
+    case ErrorKind::kProtocolViolation:
+      return "PROTOCOL_VIOLATION";
+    case ErrorKind::kResourceExhaustion:
+      return "RESOURCE_EXHAUSTION";
+    case ErrorKind::kUnsupportedOperation:
+      return "UNSUPPORTED_OPERATION";
+  }
+}
+
+enum class SendStatus {
+  // The message was enqueued successfully. As sending the message is done
+  // asynchronously, this is no guarantee that the message has been actually
+  // sent.
+  kSuccess,
+  // The message was rejected as the payload was empty (which is not allowed in
+  // SCTP).
+  kErrorMessageEmpty,
+  // The message was rejected as the payload was larger than what has been set
+  // as `DcSctpOptions.max_message_size`.
+  kErrorMessageTooLarge,
+  // The message could not be enqueued as the socket is out of resources. This
+  // mainly indicates that the send queue is full.
+  kErrorResourceExhaustion,
+  // The message could not be sent as the socket is shutting down.
+  kErrorShuttingDown,
+};
+
+inline constexpr absl::string_view ToString(SendStatus error) {
+  switch (error) {
+    case SendStatus::kSuccess:
+      return "SUCCESS";
+    case SendStatus::kErrorMessageEmpty:
+      return "ERROR_MESSAGE_EMPTY";
+    case SendStatus::kErrorMessageTooLarge:
+      return "ERROR_MESSAGE_TOO_LARGE";
+    case SendStatus::kErrorResourceExhaustion:
+      return "ERROR_RESOURCE_EXHAUSTION";
+    case SendStatus::kErrorShuttingDown:
+      return "ERROR_SHUTTING_DOWN";
+  }
+}
+
+// Return value of ResetStreams.
+enum class ResetStreamsStatus {
+  // If the connection is not yet established, this will be returned.
+  kNotConnected,
+  // Indicates that ResetStreams operation has been successfully initiated.
+  kPerformed,
+  // Indicates that ResetStreams has failed as it's not supported by the peer.
+  kNotSupported,
+};
+
+inline constexpr absl::string_view ToString(ResetStreamsStatus error) {
+  switch (error) {
+    case ResetStreamsStatus::kNotConnected:
+      return "NOT_CONNECTED";
+    case ResetStreamsStatus::kPerformed:
+      return "PERFORMED";
+    case ResetStreamsStatus::kNotSupported:
+      return "NOT_SUPPORTED";
+  }
+}
+
+// Return value of DcSctpSocketCallbacks::SendPacketWithStatus.
+enum class SendPacketStatus {
+  // Indicates that the packet was successfully sent. As sending is unreliable,
+  // there are no guarantees that the packet was actually delivered.
+  kSuccess,
+  // The packet was not sent due to a temporary failure, such as the local send
+  // buffer becoming exhausted. This return value indicates that the socket will
+  // recover and sending that packet can be retried at a later time.
+  kTemporaryFailure,
+  // The packet was not sent due to other reasons.
+  kError,
+};
+
+// Represent known SCTP implementations.
+enum class SctpImplementation {
+  // There is not enough information toto determine any SCTP implementation.
+  kUnknown,
+  // This implementation.
+  kDcsctp,
+  // https://github.com/sctplab/usrsctp.
+  kUsrSctp,
+  // Any other implementation.
+  kOther,
+};
+
+inline constexpr absl::string_view ToString(SctpImplementation implementation) {
+  switch (implementation) {
+    case SctpImplementation::kUnknown:
+      return "unknown";
+    case SctpImplementation::kDcsctp:
+      return "dcsctp";
+    case SctpImplementation::kUsrSctp:
+      return "usrsctp";
+    case SctpImplementation::kOther:
+      return "other";
+  }
+}
+
+// Tracked metrics, which is the return value of GetMetrics. Optional members
+// will be unset when they are not yet known.
+struct Metrics {
+  // Transmission stats and metrics.
+
+  // Number of packets sent.
+  size_t tx_packets_count = 0;
+
+  // Number of messages requested to be sent.
+  size_t tx_messages_count = 0;
+
+  // The current congestion window (cwnd) in bytes, corresponding to spinfo_cwnd
+  // defined in RFC6458.
+  size_t cwnd_bytes = 0;
+
+  // Smoothed round trip time, corresponding to spinfo_srtt defined in RFC6458.
+  int srtt_ms = 0;
+
+  // Number of data items in the retransmission queue that haven’t been
+  // acked/nacked yet and are in-flight. Corresponding to sstat_unackdata
+  // defined in RFC6458. This may be an approximation when there are messages in
+  // the send queue that haven't been fragmented/packetized yet.
+  size_t unack_data_count = 0;
+
+  // Receive stats and metrics.
+
+  // Number of packets received.
+  size_t rx_packets_count = 0;
+
+  // Number of messages received.
+  size_t rx_messages_count = 0;
+
+  // The peer’s last announced receiver window size, corresponding to
+  // sstat_rwnd defined in RFC6458.
+  uint32_t peer_rwnd_bytes = 0;
+
+  // Returns the detected SCTP implementation of the peer. As this is not
+  // explicitly signalled during the connection establishment, heuristics is
+  // used to analyze e.g. the state cookie in the INIT-ACK chunk.
+  SctpImplementation peer_implementation = SctpImplementation::kUnknown;
+
+  // Indicates if RFC8260 User Message Interleaving has been negotiated by both
+  // peers.
+  bool uses_message_interleaving = false;
+
+  // The number of negotiated incoming and outgoing streams, which is configured
+  // locally as `DcSctpOptions::announced_maximum_incoming_streams` and
+  // `DcSctpOptions::announced_maximum_outgoing_streams`, and which will be
+  // signaled by the peer during connection.
+  uint16_t negotiated_maximum_incoming_streams = 0;
+  uint16_t negotiated_maximum_outgoing_streams = 0;
+};
+
+// Callbacks that the DcSctpSocket will call synchronously to the owning
+// client. It is allowed to call back into the library from callbacks that start
+// with "On". It has been explicitly documented when it's not allowed to call
+// back into this library from within a callback.
+//
+// Theses callbacks are only synchronously triggered as a result of the client
+// calling a public method in `DcSctpSocketInterface`.
+class DcSctpSocketCallbacks {
+ public:
+  virtual ~DcSctpSocketCallbacks() = default;
+
+  // Called when the library wants the packet serialized as `data` to be sent.
+  //
+  // TODO(bugs.webrtc.org/12943): This method is deprecated, see
+  // `SendPacketWithStatus`.
+  //
+  // Note that it's NOT ALLOWED to call into this library from within this
+  // callback.
+  virtual void SendPacket(rtc::ArrayView<const uint8_t> data) {}
+
+  // Called when the library wants the packet serialized as `data` to be sent.
+  //
+  // Note that it's NOT ALLOWED to call into this library from within this
+  // callback.
+  virtual SendPacketStatus SendPacketWithStatus(
+      rtc::ArrayView<const uint8_t> data) {
+    SendPacket(data);
+    return SendPacketStatus::kSuccess;
+  }
+
+  // Called when the library wants to create a Timeout. The callback must return
+  // an object that implements that interface.
+  //
+  // Low precision tasks are scheduled more efficiently by using leeway to
+  // reduce Idle Wake Ups and is the preferred precision whenever possible. High
+  // precision timeouts do not have this leeway, but is still limited by OS
+  // timer precision. At the time of writing, kLow's additional leeway may be up
+  // to 17 ms, but please see webrtc::TaskQueueBase::DelayPrecision for
+  // up-to-date information.
+  //
+  // Note that it's NOT ALLOWED to call into this library from within this
+  // callback.
+  virtual std::unique_ptr<Timeout> CreateTimeout(
+      webrtc::TaskQueueBase::DelayPrecision precision) {
+    // TODO(hbos): When dependencies have migrated to this new signature, make
+    // this pure virtual and delete the other version.
+    return CreateTimeout();
+  }
+  // TODO(hbos): When dependencies have migrated to the other signature, delete
+  // this version.
+  virtual std::unique_ptr<Timeout> CreateTimeout() {
+    return CreateTimeout(webrtc::TaskQueueBase::DelayPrecision::kLow);
+  }
+
+  // Returns the current time in milliseconds (from any epoch).
+  //
+  // Note that it's NOT ALLOWED to call into this library from within this
+  // callback.
+  virtual TimeMs TimeMillis() = 0;
+
+  // Called when the library needs a random number uniformly distributed between
+  // `low` (inclusive) and `high` (exclusive). The random numbers used by the
+  // library are not used for cryptographic purposes. There are no requirements
+  // that the random number generator must be secure.
+  //
+  // Note that it's NOT ALLOWED to call into this library from within this
+  // callback.
+  virtual uint32_t GetRandomInt(uint32_t low, uint32_t high) = 0;
+
+  // Triggered when the outgoing message buffer is empty, meaning that there are
+  // no more queued messages, but there can still be packets in-flight or to be
+  // retransmitted. (in contrast to SCTP_SENDER_DRY_EVENT).
+  //
+  // Note that it's NOT ALLOWED to call into this library from within this
+  // callback.
+  ABSL_DEPRECATED("Use OnTotalBufferedAmountLow instead")
+  virtual void NotifyOutgoingMessageBufferEmpty() {}
+
+  // Called when the library has received an SCTP message in full and delivers
+  // it to the upper layer.
+  //
+  // It is allowed to call into this library from within this callback.
+  virtual void OnMessageReceived(DcSctpMessage message) = 0;
+
+  // Triggered when an non-fatal error is reported by either this library or
+  // from the other peer (by sending an ERROR command). These should be logged,
+  // but no other action need to be taken as the association is still viable.
+  //
+  // It is allowed to call into this library from within this callback.
+  virtual void OnError(ErrorKind error, absl::string_view message) = 0;
+
+  // Triggered when the socket has aborted - either as decided by this socket
+  // due to e.g. too many retransmission attempts, or by the peer when
+  // receiving an ABORT command. No other callbacks will be done after this
+  // callback, unless reconnecting.
+  //
+  // It is allowed to call into this library from within this callback.
+  virtual void OnAborted(ErrorKind error, absl::string_view message) = 0;
+
+  // Called when calling `Connect` succeeds, but also for incoming successful
+  // connection attempts.
+  //
+  // It is allowed to call into this library from within this callback.
+  virtual void OnConnected() = 0;
+
+  // Called when the socket is closed in a controlled way. No other
+  // callbacks will be done after this callback, unless reconnecting.
+  //
+  // It is allowed to call into this library from within this callback.
+  virtual void OnClosed() = 0;
+
+  // On connection restarted (by peer). This is just a notification, and the
+  // association is expected to work fine after this call, but there could have
+  // been packet loss as a result of restarting the association.
+  //
+  // It is allowed to call into this library from within this callback.
+  virtual void OnConnectionRestarted() = 0;
+
+  // Indicates that a stream reset request has failed.
+  //
+  // It is allowed to call into this library from within this callback.
+  virtual void OnStreamsResetFailed(
+      rtc::ArrayView<const StreamID> outgoing_streams,
+      absl::string_view reason) = 0;
+
+  // Indicates that a stream reset request has been performed.
+  //
+  // It is allowed to call into this library from within this callback.
+  virtual void OnStreamsResetPerformed(
+      rtc::ArrayView<const StreamID> outgoing_streams) = 0;
+
+  // When a peer has reset some of its outgoing streams, this will be called. An
+  // empty list indicates that all streams have been reset.
+  //
+  // It is allowed to call into this library from within this callback.
+  virtual void OnIncomingStreamsReset(
+      rtc::ArrayView<const StreamID> incoming_streams) = 0;
+
+  // Will be called when the amount of data buffered to be sent falls to or
+  // below the threshold set when calling `SetBufferedAmountLowThreshold`.
+  //
+  // It is allowed to call into this library from within this callback.
+  virtual void OnBufferedAmountLow(StreamID stream_id) {}
+
+  // Will be called when the total amount of data buffered (in the entire send
+  // buffer, for all streams) falls to or below the threshold specified in
+  // `DcSctpOptions::total_buffered_amount_low_threshold`.
+  virtual void OnTotalBufferedAmountLow() {}
+
+  // == Lifecycle Events ==
+  //
+  // If a `lifecycle_id` is provided as `SendOptions`, lifecycle callbacks will
+  // be triggered as the message is processed by the library.
+  //
+  // The possible transitions are shown in the graph below:
+  //
+  //        DcSctpSocket::Send  ────────────────────────┐
+  //                │                                   │
+  //                │                                   │
+  //                v                                   v
+  //    OnLifecycleMessageFullySent ───────> OnLifecycleMessageExpired
+  //                │                                   │
+  //                │                                   │
+  //                v                                   v
+  //    OnLifeCycleMessageDelivered ────────────> OnLifecycleEnd
+
+  // OnLifecycleMessageFullySent will be called when a message has been fully
+  // sent, meaning that the last fragment has been produced from the send queue
+  // and sent on the network. Note that this will trigger at most once per
+  // message even if the message was retransmitted due to packet loss.
+  //
+  // This is a lifecycle event.
+  //
+  // Note that it's NOT ALLOWED to call into this library from within this
+  // callback.
+  virtual void OnLifecycleMessageFullySent(LifecycleId lifecycle_id) {}
+
+  // OnLifecycleMessageExpired will be called when a message has expired. If it
+  // was expired with data remaining in the send queue that had not been sent
+  // ever, `maybe_delivered` will be set to false. If `maybe_delivered` is true,
+  // the message has at least once been sent and may have been correctly
+  // received by the peer, but it has expired before the receiver managed to
+  // acknowledge it. This means that if `maybe_delivered` is true, it's unknown
+  // if the message was lost or was delivered, and if `maybe_delivered` is
+  // false, it's guaranteed to not be delivered.
+  //
+  // It's guaranteed that `OnLifecycleMessageDelivered` is not called if this
+  // callback has triggered.
+  //
+  // This is a lifecycle event.
+  //
+  // Note that it's NOT ALLOWED to call into this library from within this
+  // callback.
+  virtual void OnLifecycleMessageExpired(LifecycleId lifecycle_id,
+                                         bool maybe_delivered) {}
+
+  // OnLifecycleMessageDelivered will be called when a non-expired message has
+  // been acknowledged by the peer as delivered.
+  //
+  // Note that this will trigger only when the peer moves its cumulative TSN ack
+  // beyond this message, and will not fire for messages acked using
+  // gap-ack-blocks as those are renegable. This means that this may fire a bit
+  // later than the message was actually first "acked" by the peer, as -
+  // according to the protocol - those acks may be unacked later by the client.
+  //
+  // It's guaranteed that `OnLifecycleMessageExpired` is not called if this
+  // callback has triggered.
+  //
+  // This is a lifecycle event.
+  //
+  // Note that it's NOT ALLOWED to call into this library from within this
+  // callback.
+  virtual void OnLifecycleMessageDelivered(LifecycleId lifecycle_id) {}
+
+  // OnLifecycleEnd will be called when a lifecycle event has reached its end.
+  // It will be called when processing of a message is complete, no matter how
+  // it completed. It will be called after all other lifecycle events, if any.
+  //
+  // Note that it's possible that this callback triggers without any other
+  // lifecycle callbacks having been called before in case of errors, such as
+  // attempting to send an empty message or failing to enqueue a message if the
+  // send queue is full.
+  //
+  // NOTE: When the socket is deallocated, there will be no `OnLifecycleEnd`
+  // callbacks sent for messages that were enqueued. But as long as the socket
+  // is alive, `OnLifecycleEnd` callbacks are guaranteed to be sent as messages
+  // are either expired or successfully acknowledged.
+  //
+  // This is a lifecycle event.
+  //
+  // Note that it's NOT ALLOWED to call into this library from within this
+  // callback.
+  virtual void OnLifecycleEnd(LifecycleId lifecycle_id) {}
+};
+
+// The DcSctpSocket implementation implements the following interface.
+// This class is thread-compatible.
+class DcSctpSocketInterface {
+ public:
+  virtual ~DcSctpSocketInterface() = default;
+
+  // To be called when an incoming SCTP packet is to be processed.
+  virtual void ReceivePacket(rtc::ArrayView<const uint8_t> data) = 0;
+
+  // To be called when a timeout has expired. The `timeout_id` is provided
+  // when the timeout was initiated.
+  virtual void HandleTimeout(TimeoutID timeout_id) = 0;
+
+  // Connects the socket. This is an asynchronous operation, and
+  // `DcSctpSocketCallbacks::OnConnected` will be called on success.
+  virtual void Connect() = 0;
+
+  // Puts this socket to the state in which the original socket was when its
+  // `DcSctpSocketHandoverState` was captured by `GetHandoverStateAndClose`.
+  // `RestoreFromState` is allowed only on the closed socket.
+  // `DcSctpSocketCallbacks::OnConnected` will be called if a connected socket
+  // state is restored.
+  // `DcSctpSocketCallbacks::OnError` will be called on error.
+  virtual void RestoreFromState(const DcSctpSocketHandoverState& state) = 0;
+
+  // Gracefully shutdowns the socket and sends all outstanding data. This is an
+  // asynchronous operation and `DcSctpSocketCallbacks::OnClosed` will be called
+  // on success.
+  virtual void Shutdown() = 0;
+
+  // Closes the connection non-gracefully. Will send ABORT if the connection is
+  // not already closed. No callbacks will be made after Close() has returned.
+  virtual void Close() = 0;
+
+  // The socket state.
+  virtual SocketState state() const = 0;
+
+  // The options it was created with.
+  virtual const DcSctpOptions& options() const = 0;
+
+  // Update the options max_message_size.
+  virtual void SetMaxMessageSize(size_t max_message_size) = 0;
+
+  // Sets the priority of an outgoing stream. The initial value, when not set,
+  // is `DcSctpOptions::default_stream_priority`.
+  virtual void SetStreamPriority(StreamID stream_id,
+                                 StreamPriority priority) = 0;
+
+  // Returns the currently set priority for an outgoing stream. The initial
+  // value, when not set, is `DcSctpOptions::default_stream_priority`.
+  virtual StreamPriority GetStreamPriority(StreamID stream_id) const = 0;
+
+  // Sends the message `message` using the provided send options.
+  // Sending a message is an asynchronous operation, and the `OnError` callback
+  // may be invoked to indicate any errors in sending the message.
+  //
+  // The association does not have to be established before calling this method.
+  // If it's called before there is an established association, the message will
+  // be queued.
+  virtual SendStatus Send(DcSctpMessage message,
+                          const SendOptions& send_options) = 0;
+
+  // Resetting streams is an asynchronous operation and the results will
+  // be notified using `DcSctpSocketCallbacks::OnStreamsResetDone()` on success
+  // and `DcSctpSocketCallbacks::OnStreamsResetFailed()` on failure. Note that
+  // only outgoing streams can be reset.
+  //
+  // When it's known that the peer has reset its own outgoing streams,
+  // `DcSctpSocketCallbacks::OnIncomingStreamReset` is called.
+  //
+  // Note that resetting a stream will also remove all queued messages on those
+  // streams, but will ensure that the currently sent message (if any) is fully
+  // sent before closing the stream.
+  //
+  // Resetting streams can only be done on an established association that
+  // supports stream resetting. Calling this method on e.g. a closed association
+  // or streams that don't support resetting will not perform any operation.
+  virtual ResetStreamsStatus ResetStreams(
+      rtc::ArrayView<const StreamID> outgoing_streams) = 0;
+
+  // Returns the number of bytes of data currently queued to be sent on a given
+  // stream.
+  virtual size_t buffered_amount(StreamID stream_id) const = 0;
+
+  // Returns the number of buffered outgoing bytes that is considered "low" for
+  // a given stream. See `SetBufferedAmountLowThreshold`.
+  virtual size_t buffered_amount_low_threshold(StreamID stream_id) const = 0;
+
+  // Used to specify the number of bytes of buffered outgoing data that is
+  // considered "low" for a given stream, which will trigger an
+  // OnBufferedAmountLow event. The default value is zero (0).
+  virtual void SetBufferedAmountLowThreshold(StreamID stream_id,
+                                             size_t bytes) = 0;
+
+  // Retrieves the latest metrics. If the socket is not fully connected,
+  // `absl::nullopt` will be returned.
+  virtual absl::optional<Metrics> GetMetrics() const = 0;
+
+  // Returns empty bitmask if the socket is in the state in which a snapshot of
+  // the state can be made by `GetHandoverStateAndClose()`. Return value is
+  // invalidated by a call to any non-const method.
+  virtual HandoverReadinessStatus GetHandoverReadiness() const = 0;
+
+  // Collects a snapshot of the socket state that can be used to reconstruct
+  // this socket in another process. On success this socket object is closed
+  // synchronously and no callbacks will be made after the method has returned.
+  // The method fails if the socket is not in a state ready for handover.
+  // nullopt indicates the failure. `DcSctpSocketCallbacks::OnClosed` will be
+  // called on success.
+  virtual absl::optional<DcSctpSocketHandoverState>
+  GetHandoverStateAndClose() = 0;
+
+  // Returns the detected SCTP implementation of the peer. As this is not
+  // explicitly signalled during the connection establishment, heuristics is
+  // used to analyze e.g. the state cookie in the INIT-ACK chunk.
+  //
+  // If this method is called too early (before
+  // `DcSctpSocketCallbacks::OnConnected` has triggered), this will likely
+  // return `SctpImplementation::kUnknown`.
+  ABSL_DEPRECATED("See Metrics::peer_implementation instead")
+  virtual SctpImplementation peer_implementation() const {
+    return SctpImplementation::kUnknown;
+  }
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PUBLIC_DCSCTP_SOCKET_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/dcsctp_socket_factory.cc b/third_party/libwebrtc/net/dcsctp/public/dcsctp_socket_factory.cc
new file mode 100644
index 0000000000..ebcb5553e3
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/dcsctp_socket_factory.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "net/dcsctp/public/dcsctp_socket_factory.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/packet_observer.h"
+#include "net/dcsctp/socket/dcsctp_socket.h"
+
+namespace dcsctp {
+
+DcSctpSocketFactory::~DcSctpSocketFactory() = default;
+
+std::unique_ptr<DcSctpSocketInterface> DcSctpSocketFactory::Create(
+    absl::string_view log_prefix,
+    DcSctpSocketCallbacks& callbacks,
+    std::unique_ptr<PacketObserver> packet_observer,
+    const DcSctpOptions& options) {
+  return std::make_unique<DcSctpSocket>(log_prefix, callbacks,
+                                        std::move(packet_observer), options);
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/public/dcsctp_socket_factory.h b/third_party/libwebrtc/net/dcsctp/public/dcsctp_socket_factory.h
new file mode 100644
index 0000000000..ca429d3275
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/dcsctp_socket_factory.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_DCSCTP_SOCKET_FACTORY_H_
+#define NET_DCSCTP_PUBLIC_DCSCTP_SOCKET_FACTORY_H_
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/packet_observer.h"
+
+namespace dcsctp {
+class DcSctpSocketFactory {
+ public:
+  virtual ~DcSctpSocketFactory();
+  virtual std::unique_ptr<DcSctpSocketInterface> Create(
+      absl::string_view log_prefix,
+      DcSctpSocketCallbacks& callbacks,
+      std::unique_ptr<PacketObserver> packet_observer,
+      const DcSctpOptions& options);
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PUBLIC_DCSCTP_SOCKET_FACTORY_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/mock_dcsctp_socket.h b/third_party/libwebrtc/net/dcsctp/public/mock_dcsctp_socket.h
new file mode 100644
index 0000000000..0fd572bd94
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/mock_dcsctp_socket.h
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_MOCK_DCSCTP_SOCKET_H_
+#define NET_DCSCTP_PUBLIC_MOCK_DCSCTP_SOCKET_H_
+
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+
+class MockDcSctpSocket : public DcSctpSocketInterface {
+ public:
+  MOCK_METHOD(void,
+              ReceivePacket,
+              (rtc::ArrayView<const uint8_t> data),
+              (override));
+
+  MOCK_METHOD(void, HandleTimeout, (TimeoutID timeout_id), (override));
+
+  MOCK_METHOD(void, Connect, (), (override));
+
+  MOCK_METHOD(void,
+              RestoreFromState,
+              (const DcSctpSocketHandoverState&),
+              (override));
+
+  MOCK_METHOD(void, Shutdown, (), (override));
+
+  MOCK_METHOD(void, Close, (), (override));
+
+  MOCK_METHOD(SocketState, state, (), (const, override));
+
+  MOCK_METHOD(const DcSctpOptions&, options, (), (const, override));
+
+  MOCK_METHOD(void, SetMaxMessageSize, (size_t max_message_size), (override));
+
+  MOCK_METHOD(void,
+              SetStreamPriority,
+              (StreamID stream_id, StreamPriority priority),
+              (override));
+
+  MOCK_METHOD(StreamPriority,
+              GetStreamPriority,
+              (StreamID stream_id),
+              (const, override));
+
+  MOCK_METHOD(SendStatus,
+              Send,
+              (DcSctpMessage message, const SendOptions& send_options),
+              (override));
+
+  MOCK_METHOD(ResetStreamsStatus,
+              ResetStreams,
+              (rtc::ArrayView<const StreamID> outgoing_streams),
+              (override));
+
+  MOCK_METHOD(size_t, buffered_amount, (StreamID stream_id), (const, override));
+
+  MOCK_METHOD(size_t,
+              buffered_amount_low_threshold,
+              (StreamID stream_id),
+              (const, override));
+
+  MOCK_METHOD(void,
+              SetBufferedAmountLowThreshold,
+              (StreamID stream_id, size_t bytes),
+              (override));
+
+  MOCK_METHOD(absl::optional<Metrics>, GetMetrics, (), (const, override));
+
+  MOCK_METHOD(HandoverReadinessStatus,
+              GetHandoverReadiness,
+              (),
+              (const, override));
+  MOCK_METHOD(absl::optional<DcSctpSocketHandoverState>,
+              GetHandoverStateAndClose,
+              (),
+              (override));
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PUBLIC_MOCK_DCSCTP_SOCKET_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/mock_dcsctp_socket_factory.h b/third_party/libwebrtc/net/dcsctp/public/mock_dcsctp_socket_factory.h
new file mode 100644
index 0000000000..61f05577f2
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/mock_dcsctp_socket_factory.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_MOCK_DCSCTP_SOCKET_FACTORY_H_
+#define NET_DCSCTP_PUBLIC_MOCK_DCSCTP_SOCKET_FACTORY_H_
+
+#include <memory>
+
+#include "net/dcsctp/public/dcsctp_socket_factory.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+
+class MockDcSctpSocketFactory : public DcSctpSocketFactory {
+ public:
+  MOCK_METHOD(std::unique_ptr<DcSctpSocketInterface>,
+              Create,
+              (absl::string_view log_prefix,
+               DcSctpSocketCallbacks& callbacks,
+               std::unique_ptr<PacketObserver> packet_observer,
+               const DcSctpOptions& options),
+              (override));
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PUBLIC_MOCK_DCSCTP_SOCKET_FACTORY_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/mock_dcsctp_socket_test.cc b/third_party/libwebrtc/net/dcsctp/public/mock_dcsctp_socket_test.cc
new file mode 100644
index 0000000000..57013e4ce2
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/mock_dcsctp_socket_test.cc
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/public/mock_dcsctp_socket.h"
+
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+
+// This test exists to ensure that all methods are mocked correctly, and to
+// generate compiler errors if they are not.
+TEST(MockDcSctpSocketTest, CanInstantiateAndConnect) {
+  testing::StrictMock<MockDcSctpSocket> socket;
+
+  EXPECT_CALL(socket, Connect);
+
+  socket.Connect();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/public/packet_observer.h b/third_party/libwebrtc/net/dcsctp/public/packet_observer.h
new file mode 100644
index 0000000000..fe7567824f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/packet_observer.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_PACKET_OBSERVER_H_
+#define NET_DCSCTP_PUBLIC_PACKET_OBSERVER_H_
+
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// A PacketObserver can be attached to a socket and will be called for
+// all sent and received packets.
+class PacketObserver {
+ public:
+  virtual ~PacketObserver() = default;
+  // Called when a packet is sent, with the current time (in milliseconds) as
+  // `now`, and the packet payload as `payload`.
+  virtual void OnSentPacket(TimeMs now,
+                            rtc::ArrayView<const uint8_t> payload) = 0;
+
+  // Called when a packet is received, with the current time (in milliseconds)
+  // as `now`, and the packet payload as `payload`.
+  virtual void OnReceivedPacket(TimeMs now,
+                                rtc::ArrayView<const uint8_t> payload) = 0;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PUBLIC_PACKET_OBSERVER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/text_pcap_packet_observer.cc b/third_party/libwebrtc/net/dcsctp/public/text_pcap_packet_observer.cc
new file mode 100644
index 0000000000..2b13060190
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/text_pcap_packet_observer.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/public/text_pcap_packet_observer.h"
+
+#include "api/array_view.h"
+#include "net/dcsctp/public/types.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+void TextPcapPacketObserver::OnSentPacket(
+    dcsctp::TimeMs now,
+    rtc::ArrayView<const uint8_t> payload) {
+  PrintPacket("O ", name_, now, payload);
+}
+
+void TextPcapPacketObserver::OnReceivedPacket(
+    dcsctp::TimeMs now,
+    rtc::ArrayView<const uint8_t> payload) {
+  PrintPacket("I ", name_, now, payload);
+}
+
+void TextPcapPacketObserver::PrintPacket(
+    absl::string_view prefix,
+    absl::string_view socket_name,
+    dcsctp::TimeMs now,
+    rtc::ArrayView<const uint8_t> payload) {
+  rtc::StringBuilder s;
+  s << "\n" << prefix;
+  int64_t remaining = *now % (24 * 60 * 60 * 1000);
+  int hours = remaining / (60 * 60 * 1000);
+  remaining = remaining % (60 * 60 * 1000);
+  int minutes = remaining / (60 * 1000);
+  remaining = remaining % (60 * 1000);
+  int seconds = remaining / 1000;
+  int ms = remaining % 1000;
+  s.AppendFormat("%02d:%02d:%02d.%03d", hours, minutes, seconds, ms);
+  s << " 0000";
+  for (uint8_t byte : payload) {
+    s.AppendFormat(" %02x", byte);
+  }
+  s << " # SCTP_PACKET " << socket_name;
+  RTC_LOG(LS_VERBOSE) << s.str();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/public/text_pcap_packet_observer.h b/third_party/libwebrtc/net/dcsctp/public/text_pcap_packet_observer.h
new file mode 100644
index 0000000000..0685771ccf
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/text_pcap_packet_observer.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_TEXT_PCAP_PACKET_OBSERVER_H_
+#define NET_DCSCTP_PUBLIC_TEXT_PCAP_PACKET_OBSERVER_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/public/packet_observer.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// Print outs all sent and received packets to the logs, at LS_VERBOSE severity.
+class TextPcapPacketObserver : public dcsctp::PacketObserver {
+ public:
+  explicit TextPcapPacketObserver(absl::string_view name) : name_(name) {}
+
+  // Implementation of `dcsctp::PacketObserver`.
+  void OnSentPacket(dcsctp::TimeMs now,
+                    rtc::ArrayView<const uint8_t> payload) override;
+
+  void OnReceivedPacket(dcsctp::TimeMs now,
+                        rtc::ArrayView<const uint8_t> payload) override;
+
+  // Prints a packet to the log. Exposed to allow it to be used in compatibility
+  // tests suites that don't use PacketObserver.
+  static void PrintPacket(absl::string_view prefix,
+                          absl::string_view socket_name,
+                          dcsctp::TimeMs now,
+                          rtc::ArrayView<const uint8_t> payload);
+
+ private:
+  const std::string name_;
+};
+
+}  // namespace dcsctp
+#endif  // NET_DCSCTP_PUBLIC_TEXT_PCAP_PACKET_OBSERVER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/timeout.h b/third_party/libwebrtc/net/dcsctp/public/timeout.h
new file mode 100644
index 0000000000..64ba351093
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/timeout.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_TIMEOUT_H_
+#define NET_DCSCTP_PUBLIC_TIMEOUT_H_
+
+#include <cstdint>
+
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// A very simple timeout that can be started and stopped. When started,
+// it will be given a unique `timeout_id` which should be provided to
+// `DcSctpSocket::HandleTimeout` when it expires.
+class Timeout {
+ public:
+  virtual ~Timeout() = default;
+
+  // Called to start time timeout, with the duration in milliseconds as
+  // `duration` and with the timeout identifier as `timeout_id`, which - if
+  // the timeout expires - shall be provided to `DcSctpSocket::HandleTimeout`.
+  //
+  // `Start` and `Stop` will always be called in pairs. In other words will
+  // ´Start` never be called twice, without a call to `Stop` in between.
+  virtual void Start(DurationMs duration, TimeoutID timeout_id) = 0;
+
+  // Called to stop the running timeout.
+  //
+  // `Start` and `Stop` will always be called in pairs. In other words will
+  // ´Start` never be called twice, without a call to `Stop` in between.
+  //
+  // `Stop` will always be called prior to releasing this object.
+  virtual void Stop() = 0;
+
+  // Called to restart an already running timeout, with the `duration` and
+  // `timeout_id` parameters as described in `Start`. This can be overridden by
+  // the implementation to restart it more efficiently.
+  virtual void Restart(DurationMs duration, TimeoutID timeout_id) {
+    Stop();
+    Start(duration, timeout_id);
+  }
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PUBLIC_TIMEOUT_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/types.h b/third_party/libwebrtc/net/dcsctp/public/types.h
new file mode 100644
index 0000000000..d0725620d8
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/types.h
@@ -0,0 +1,143 @@
+/*
+ *  Copyright 2019 The Chromium Authors. All rights reserved.
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_PUBLIC_TYPES_H_
+#define NET_DCSCTP_PUBLIC_TYPES_H_
+
+#include <cstdint>
+#include <limits>
+
+#include "rtc_base/strong_alias.h"
+
+namespace dcsctp {
+
+// Stream Identifier
+using StreamID = webrtc::StrongAlias<class StreamIDTag, uint16_t>;
+
+// Payload Protocol Identifier (PPID)
+using PPID = webrtc::StrongAlias<class PPIDTag, uint32_t>;
+
+// Timeout Identifier
+using TimeoutID = webrtc::StrongAlias<class TimeoutTag, uint64_t>;
+
+// Indicates if a message is allowed to be received out-of-order compared to
+// other messages on the same stream.
+using IsUnordered = webrtc::StrongAlias<class IsUnorderedTag, bool>;
+
+// Stream priority, where higher values indicate higher priority. The meaning of
+// this value and how it's used depends on the stream scheduler.
+using StreamPriority = webrtc::StrongAlias<class StreamPriorityTag, uint16_t>;
+
+// Duration, as milliseconds. Overflows after 24 days.
+class DurationMs : public webrtc::StrongAlias<class DurationMsTag, int32_t> {
+ public:
+  constexpr explicit DurationMs(const UnderlyingType& v)
+      : webrtc::StrongAlias<class DurationMsTag, int32_t>(v) {}
+
+  // Convenience methods for working with time.
+  constexpr DurationMs& operator+=(DurationMs d) {
+    value_ += d.value_;
+    return *this;
+  }
+  constexpr DurationMs& operator-=(DurationMs d) {
+    value_ -= d.value_;
+    return *this;
+  }
+  template <typename T>
+  constexpr DurationMs& operator*=(T factor) {
+    value_ *= factor;
+    return *this;
+  }
+};
+
+constexpr inline DurationMs operator+(DurationMs lhs, DurationMs rhs) {
+  return lhs += rhs;
+}
+constexpr inline DurationMs operator-(DurationMs lhs, DurationMs rhs) {
+  return lhs -= rhs;
+}
+template <typename T>
+constexpr inline DurationMs operator*(DurationMs lhs, T rhs) {
+  return lhs *= rhs;
+}
+template <typename T>
+constexpr inline DurationMs operator*(T lhs, DurationMs rhs) {
+  return rhs *= lhs;
+}
+constexpr inline int32_t operator/(DurationMs lhs, DurationMs rhs) {
+  return lhs.value() / rhs.value();
+}
+
+// Represents time, in milliseconds since a client-defined epoch.
+class TimeMs : public webrtc::StrongAlias<class TimeMsTag, int64_t> {
+ public:
+  constexpr explicit TimeMs(const UnderlyingType& v)
+      : webrtc::StrongAlias<class TimeMsTag, int64_t>(v) {}
+
+  // Convenience methods for working with time.
+  constexpr TimeMs& operator+=(DurationMs d) {
+    value_ += *d;
+    return *this;
+  }
+  constexpr TimeMs& operator-=(DurationMs d) {
+    value_ -= *d;
+    return *this;
+  }
+
+  static constexpr TimeMs InfiniteFuture() {
+    return TimeMs(std::numeric_limits<int64_t>::max());
+  }
+};
+
+constexpr inline TimeMs operator+(TimeMs lhs, DurationMs rhs) {
+  return lhs += rhs;
+}
+constexpr inline TimeMs operator+(DurationMs lhs, TimeMs rhs) {
+  return rhs += lhs;
+}
+constexpr inline TimeMs operator-(TimeMs lhs, DurationMs rhs) {
+  return lhs -= rhs;
+}
+constexpr inline DurationMs operator-(TimeMs lhs, TimeMs rhs) {
+  return DurationMs(*lhs - *rhs);
+}
+
+// The maximum number of times the socket should attempt to retransmit a
+// message which fails the first time in unreliable mode.
+class MaxRetransmits
+    : public webrtc::StrongAlias<class MaxRetransmitsTag, uint16_t> {
+ public:
+  constexpr explicit MaxRetransmits(const UnderlyingType& v)
+      : webrtc::StrongAlias<class MaxRetransmitsTag, uint16_t>(v) {}
+
+  // There should be no limit - the message should be sent reliably.
+  static constexpr MaxRetransmits NoLimit() {
+    return MaxRetransmits(std::numeric_limits<uint16_t>::max());
+  }
+};
+
+// An identifier that can be set on sent messages, and picked by the sending
+// client. If different from `::NotSet()`, lifecycle events will be generated,
+// and eventually `DcSctpSocketCallbacks::OnLifecycleEnd` will be called to
+// indicate that the lifecycle isn't tracked any longer. The value zero (0) is
+// not a valid lifecycle identifier, and will be interpreted as not having it
+// set.
+class LifecycleId : public webrtc::StrongAlias<class LifecycleIdTag, uint64_t> {
+ public:
+  constexpr explicit LifecycleId(const UnderlyingType& v)
+      : webrtc::StrongAlias<class LifecycleIdTag, uint64_t>(v) {}
+
+  constexpr bool IsSet() const { return value_ != 0; }
+
+  static constexpr LifecycleId NotSet() { return LifecycleId(0); }
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_PUBLIC_TYPES_H_
diff --git a/third_party/libwebrtc/net/dcsctp/public/types_test.cc b/third_party/libwebrtc/net/dcsctp/public/types_test.cc
new file mode 100644
index 0000000000..d3d1240751
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/public/types_test.cc
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/public/types.h"
+
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(TypesTest, DurationOperators) {
+  DurationMs d1(10);
+  DurationMs d2(25);
+  EXPECT_EQ(d1 + d2, DurationMs(35));
+  EXPECT_EQ(d2 - d1, DurationMs(15));
+
+  d1 += d2;
+  EXPECT_EQ(d1, DurationMs(35));
+
+  d1 -= DurationMs(5);
+  EXPECT_EQ(d1, DurationMs(30));
+
+  d1 *= 1.5;
+  EXPECT_EQ(d1, DurationMs(45));
+
+  EXPECT_EQ(DurationMs(10) * 2, DurationMs(20));
+}
+
+TEST(TypesTest, TimeOperators) {
+  EXPECT_EQ(TimeMs(250) + DurationMs(100), TimeMs(350));
+  EXPECT_EQ(DurationMs(250) + TimeMs(100), TimeMs(350));
+  EXPECT_EQ(TimeMs(250) - DurationMs(100), TimeMs(150));
+  EXPECT_EQ(TimeMs(250) - TimeMs(100), DurationMs(150));
+
+  TimeMs t1(150);
+  t1 -= DurationMs(50);
+  EXPECT_EQ(t1, TimeMs(100));
+  t1 += DurationMs(200);
+  EXPECT_EQ(t1, TimeMs(300));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/rx/BUILD.gn b/third_party/libwebrtc/net/dcsctp/rx/BUILD.gn
new file mode 100644
index 0000000000..d66fd6ba72
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/BUILD.gn
@@ -0,0 +1,148 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("data_tracker") {
+  deps = [
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:stringutils",
+    "../common:sequence_numbers",
+    "../packet:chunk",
+    "../packet:data",
+    "../public:socket",
+    "../timer",
+  ]
+  sources = [
+    "data_tracker.cc",
+    "data_tracker.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_source_set("reassembly_streams") {
+  deps = [
+    "../../../api:array_view",
+    "../common:sequence_numbers",
+    "../packet:chunk",
+    "../packet:data",
+    "../public:socket",
+    "../public:types",
+  ]
+  sources = [ "reassembly_streams.h" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_library("interleaved_reassembly_streams") {
+  deps = [
+    ":reassembly_streams",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../common:sequence_numbers",
+    "../packet:chunk",
+    "../packet:data",
+    "../public:types",
+  ]
+  sources = [
+    "interleaved_reassembly_streams.cc",
+    "interleaved_reassembly_streams.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+rtc_library("traditional_reassembly_streams") {
+  deps = [
+    ":reassembly_streams",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../common:sequence_numbers",
+    "../packet:chunk",
+    "../packet:data",
+    "../public:types",
+  ]
+  sources = [
+    "traditional_reassembly_streams.cc",
+    "traditional_reassembly_streams.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("reassembly_queue") {
+  deps = [
+    ":interleaved_reassembly_streams",
+    ":reassembly_streams",
+    ":traditional_reassembly_streams",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../common:internal_types",
+    "../common:sequence_numbers",
+    "../common:str_join",
+    "../packet:chunk",
+    "../packet:data",
+    "../packet:parameter",
+    "../public:socket",
+    "../public:types",
+  ]
+  sources = [
+    "reassembly_queue.cc",
+    "reassembly_queue.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("dcsctp_rx_unittests") {
+    testonly = true
+
+    deps = [
+      ":data_tracker",
+      ":interleaved_reassembly_streams",
+      ":reassembly_queue",
+      ":reassembly_streams",
+      ":traditional_reassembly_streams",
+      "../../../api:array_view",
+      "../../../api/task_queue:task_queue",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:gunit_helpers",
+      "../../../test:test_support",
+      "../common:handover_testing",
+      "../common:sequence_numbers",
+      "../packet:chunk",
+      "../packet:data",
+      "../public:types",
+      "../testing:data_generator",
+      "../timer",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+    sources = [
+      "data_tracker_test.cc",
+      "interleaved_reassembly_streams_test.cc",
+      "reassembly_queue_test.cc",
+      "traditional_reassembly_streams_test.cc",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/net/dcsctp/rx/data_tracker.cc b/third_party/libwebrtc/net/dcsctp/rx/data_tracker.cc
new file mode 100644
index 0000000000..1f2e43f7f5
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/data_tracker.cc
@@ -0,0 +1,386 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/rx/data_tracker.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <iterator>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/timer/timer.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+constexpr size_t DataTracker::kMaxDuplicateTsnReported;
+constexpr size_t DataTracker::kMaxGapAckBlocksReported;
+
+bool DataTracker::AdditionalTsnBlocks::Add(UnwrappedTSN tsn) {
+  // Find any block to expand. It will look for any block that includes (also
+  // when expanded) the provided `tsn`. It will return the block that is greater
+  // than, or equal to `tsn`.
+  auto it = absl::c_lower_bound(
+      blocks_, tsn, [&](const TsnRange& elem, const UnwrappedTSN& t) {
+        return elem.last.next_value() < t;
+      });
+
+  if (it == blocks_.end()) {
+    // No matching block found. There is no greater than, or equal block - which
+    // means that this TSN is greater than any block. It can then be inserted at
+    // the end.
+    blocks_.emplace_back(tsn, tsn);
+    return true;
+  }
+
+  if (tsn >= it->first && tsn <= it->last) {
+    // It's already in this block.
+    return false;
+  }
+
+  if (it->last.next_value() == tsn) {
+    // This block can be expanded to the right, or merged with the next.
+    auto next_it = it + 1;
+    if (next_it != blocks_.end() && tsn.next_value() == next_it->first) {
+      // Expanding it would make it adjacent to next block - merge those.
+      it->last = next_it->last;
+      blocks_.erase(next_it);
+      return true;
+    }
+
+    // Expand to the right
+    it->last = tsn;
+    return true;
+  }
+
+  if (it->first == tsn.next_value()) {
+    // This block can be expanded to the left. Merging to the left would've been
+    // covered by the above "merge to the right". Both blocks (expand a
+    // right-most block to the left and expand a left-most block to the right)
+    // would match, but the left-most would be returned by std::lower_bound.
+    RTC_DCHECK(it == blocks_.begin() || (it - 1)->last.next_value() != tsn);
+
+    // Expand to the left.
+    it->first = tsn;
+    return true;
+  }
+
+  // Need to create a new block in the middle.
+  blocks_.emplace(it, tsn, tsn);
+  return true;
+}
+
+void DataTracker::AdditionalTsnBlocks::EraseTo(UnwrappedTSN tsn) {
+  // Find the block that is greater than or equals `tsn`.
+  auto it = absl::c_lower_bound(
+      blocks_, tsn, [&](const TsnRange& elem, const UnwrappedTSN& t) {
+        return elem.last < t;
+      });
+
+  // The block that is found is greater or equal (or possibly ::end, when no
+  // block is greater or equal). All blocks before this block can be safely
+  // removed. the TSN might be within this block, so possibly truncate it.
+  bool tsn_is_within_block = it != blocks_.end() && tsn >= it->first;
+  blocks_.erase(blocks_.begin(), it);
+
+  if (tsn_is_within_block) {
+    blocks_.front().first = tsn.next_value();
+  }
+}
+
+void DataTracker::AdditionalTsnBlocks::PopFront() {
+  RTC_DCHECK(!blocks_.empty());
+  blocks_.erase(blocks_.begin());
+}
+
+bool DataTracker::IsTSNValid(TSN tsn) const {
+  UnwrappedTSN unwrapped_tsn = tsn_unwrapper_.PeekUnwrap(tsn);
+
+  // Note that this method doesn't return `false` for old DATA chunks, as those
+  // are actually valid, and receiving those may affect the generated SACK
+  // response (by setting "duplicate TSNs").
+
+  uint32_t difference =
+      UnwrappedTSN::Difference(unwrapped_tsn, last_cumulative_acked_tsn_);
+  if (difference > kMaxAcceptedOutstandingFragments) {
+    return false;
+  }
+  return true;
+}
+
+bool DataTracker::Observe(TSN tsn,
+                          AnyDataChunk::ImmediateAckFlag immediate_ack) {
+  bool is_duplicate = false;
+  UnwrappedTSN unwrapped_tsn = tsn_unwrapper_.Unwrap(tsn);
+
+  // IsTSNValid must be called prior to calling this method.
+  RTC_DCHECK(
+      UnwrappedTSN::Difference(unwrapped_tsn, last_cumulative_acked_tsn_) <=
+      kMaxAcceptedOutstandingFragments);
+
+  // Old chunk already seen before?
+  if (unwrapped_tsn <= last_cumulative_acked_tsn_) {
+    if (duplicate_tsns_.size() < kMaxDuplicateTsnReported) {
+      duplicate_tsns_.insert(unwrapped_tsn.Wrap());
+    }
+    // https://datatracker.ietf.org/doc/html/rfc4960#section-6.2
+    // "When a packet arrives with duplicate DATA chunk(s) and with no new DATA
+    // chunk(s), the endpoint MUST immediately send a SACK with no delay. If a
+    // packet arrives with duplicate DATA chunk(s) bundled with new DATA chunks,
+    // the endpoint MAY immediately send a SACK."
+    UpdateAckState(AckState::kImmediate, "duplicate data");
+    is_duplicate = true;
+  } else {
+    if (unwrapped_tsn == last_cumulative_acked_tsn_.next_value()) {
+      last_cumulative_acked_tsn_ = unwrapped_tsn;
+      // The cumulative acked tsn may be moved even further, if a gap was
+      // filled.
+      if (!additional_tsn_blocks_.empty() &&
+          additional_tsn_blocks_.front().first ==
+              last_cumulative_acked_tsn_.next_value()) {
+        last_cumulative_acked_tsn_ = additional_tsn_blocks_.front().last;
+        additional_tsn_blocks_.PopFront();
+      }
+    } else {
+      bool inserted = additional_tsn_blocks_.Add(unwrapped_tsn);
+      if (!inserted) {
+        // Already seen before.
+        if (duplicate_tsns_.size() < kMaxDuplicateTsnReported) {
+          duplicate_tsns_.insert(unwrapped_tsn.Wrap());
+        }
+        // https://datatracker.ietf.org/doc/html/rfc4960#section-6.2
+        // "When a packet arrives with duplicate DATA chunk(s) and with no new
+        // DATA chunk(s), the endpoint MUST immediately send a SACK with no
+        // delay. If a packet arrives with duplicate DATA chunk(s) bundled with
+        // new DATA chunks, the endpoint MAY immediately send a SACK."
+        // No need to do this. SACKs are sent immediately on packet loss below.
+        is_duplicate = true;
+      }
+    }
+  }
+
+  // https://tools.ietf.org/html/rfc4960#section-6.7
+  // "Upon the reception of a new DATA chunk, an endpoint shall examine the
+  // continuity of the TSNs received.  If the endpoint detects a gap in
+  // the received DATA chunk sequence, it SHOULD send a SACK with Gap Ack
+  // Blocks immediately.  The data receiver continues sending a SACK after
+  // receipt of each SCTP packet that doesn't fill the gap."
+  if (!additional_tsn_blocks_.empty()) {
+    UpdateAckState(AckState::kImmediate, "packet loss");
+  }
+
+  // https://tools.ietf.org/html/rfc7053#section-5.2
+  // "Upon receipt of an SCTP packet containing a DATA chunk with the I
+  // bit set, the receiver SHOULD NOT delay the sending of the corresponding
+  // SACK chunk, i.e., the receiver SHOULD immediately respond with the
+  // corresponding SACK chunk."
+  if (*immediate_ack) {
+    UpdateAckState(AckState::kImmediate, "immediate-ack bit set");
+  }
+
+  if (!seen_packet_) {
+    // https://tools.ietf.org/html/rfc4960#section-5.1
+    // "After the reception of the first DATA chunk in an association the
+    // endpoint MUST immediately respond with a SACK to acknowledge the DATA
+    // chunk."
+    seen_packet_ = true;
+    UpdateAckState(AckState::kImmediate, "first DATA chunk");
+  }
+
+  // https://tools.ietf.org/html/rfc4960#section-6.2
+  // "Specifically, an acknowledgement SHOULD be generated for at least
+  // every second packet (not every second DATA chunk) received, and SHOULD be
+  // generated within 200 ms of the arrival of any unacknowledged DATA chunk."
+  if (ack_state_ == AckState::kIdle) {
+    UpdateAckState(AckState::kBecomingDelayed, "received DATA when idle");
+  } else if (ack_state_ == AckState::kDelayed) {
+    UpdateAckState(AckState::kImmediate, "received DATA when already delayed");
+  }
+  return !is_duplicate;
+}
+
+void DataTracker::HandleForwardTsn(TSN new_cumulative_ack) {
+  // ForwardTSN is sent to make the receiver (this socket) "forget" about partly
+  // received (or not received at all) data, up until `new_cumulative_ack`.
+
+  UnwrappedTSN unwrapped_tsn = tsn_unwrapper_.Unwrap(new_cumulative_ack);
+  UnwrappedTSN prev_last_cum_ack_tsn = last_cumulative_acked_tsn_;
+
+  // Old chunk already seen before?
+  if (unwrapped_tsn <= last_cumulative_acked_tsn_) {
+    // https://tools.ietf.org/html/rfc3758#section-3.6
+    // "Note, if the "New Cumulative TSN" value carried in the arrived
+    // FORWARD TSN chunk is found to be behind or at the current cumulative TSN
+    // point, the data receiver MUST treat this FORWARD TSN as out-of-date and
+    // MUST NOT update its Cumulative TSN.  The receiver SHOULD send a SACK to
+    // its peer (the sender of the FORWARD TSN) since such a duplicate may
+    // indicate the previous SACK was lost in the network."
+    UpdateAckState(AckState::kImmediate,
+                   "FORWARD_TSN new_cumulative_tsn was behind");
+    return;
+  }
+
+  // https://tools.ietf.org/html/rfc3758#section-3.6
+  // "When a FORWARD TSN chunk arrives, the data receiver MUST first update
+  // its cumulative TSN point to the value carried in the FORWARD TSN chunk, and
+  // then MUST further advance its cumulative TSN point locally if possible, as
+  // shown by the following example..."
+
+  // The `new_cumulative_ack` will become the current
+  // `last_cumulative_acked_tsn_`, and if there have been prior "gaps" that are
+  // now overlapping with the new value, remove them.
+  last_cumulative_acked_tsn_ = unwrapped_tsn;
+  additional_tsn_blocks_.EraseTo(unwrapped_tsn);
+
+  // See if the `last_cumulative_acked_tsn_` can be moved even further:
+  if (!additional_tsn_blocks_.empty() &&
+      additional_tsn_blocks_.front().first ==
+          last_cumulative_acked_tsn_.next_value()) {
+    last_cumulative_acked_tsn_ = additional_tsn_blocks_.front().last;
+    additional_tsn_blocks_.PopFront();
+  }
+
+  RTC_DLOG(LS_VERBOSE) << log_prefix_ << "FORWARD_TSN, cum_ack_tsn="
+                       << *prev_last_cum_ack_tsn.Wrap() << "->"
+                       << *new_cumulative_ack << "->"
+                       << *last_cumulative_acked_tsn_.Wrap();
+
+  // https://tools.ietf.org/html/rfc3758#section-3.6
+  // "Any time a FORWARD TSN chunk arrives, for the purposes of sending a
+  // SACK, the receiver MUST follow the same rules as if a DATA chunk had been
+  // received (i.e., follow the delayed sack rules specified in ..."
+  if (ack_state_ == AckState::kIdle) {
+    UpdateAckState(AckState::kBecomingDelayed,
+                   "received FORWARD_TSN when idle");
+  } else if (ack_state_ == AckState::kDelayed) {
+    UpdateAckState(AckState::kImmediate,
+                   "received FORWARD_TSN when already delayed");
+  }
+}
+
+SackChunk DataTracker::CreateSelectiveAck(size_t a_rwnd) {
+  // Note that in SCTP, the receiver side is allowed to discard received data
+  // and signal that to the sender, but only chunks that have previously been
+  // reported in the gap-ack-blocks. However, this implementation will never do
+  // that. So this SACK produced is more like a NR-SACK as explained in
+  // https://ieeexplore.ieee.org/document/4697037 and which there is an RFC
+  // draft at https://tools.ietf.org/html/draft-tuexen-tsvwg-sctp-multipath-17.
+  std::set<TSN> duplicate_tsns;
+  duplicate_tsns_.swap(duplicate_tsns);
+
+  return SackChunk(last_cumulative_acked_tsn_.Wrap(), a_rwnd,
+                   CreateGapAckBlocks(), std::move(duplicate_tsns));
+}
+
+std::vector<SackChunk::GapAckBlock> DataTracker::CreateGapAckBlocks() const {
+  const auto& blocks = additional_tsn_blocks_.blocks();
+  std::vector<SackChunk::GapAckBlock> gap_ack_blocks;
+  gap_ack_blocks.reserve(std::min(blocks.size(), kMaxGapAckBlocksReported));
+  for (size_t i = 0; i < blocks.size() && i < kMaxGapAckBlocksReported; ++i) {
+    auto start_diff =
+        UnwrappedTSN::Difference(blocks[i].first, last_cumulative_acked_tsn_);
+    auto end_diff =
+        UnwrappedTSN::Difference(blocks[i].last, last_cumulative_acked_tsn_);
+    gap_ack_blocks.emplace_back(static_cast<uint16_t>(start_diff),
+                                static_cast<uint16_t>(end_diff));
+  }
+
+  return gap_ack_blocks;
+}
+
+bool DataTracker::ShouldSendAck(bool also_if_delayed) {
+  if (ack_state_ == AckState::kImmediate ||
+      (also_if_delayed && (ack_state_ == AckState::kBecomingDelayed ||
+                           ack_state_ == AckState::kDelayed))) {
+    UpdateAckState(AckState::kIdle, "sending SACK");
+    return true;
+  }
+
+  return false;
+}
+
+bool DataTracker::will_increase_cum_ack_tsn(TSN tsn) const {
+  UnwrappedTSN unwrapped = tsn_unwrapper_.PeekUnwrap(tsn);
+  return unwrapped == last_cumulative_acked_tsn_.next_value();
+}
+
+void DataTracker::ForceImmediateSack() {
+  ack_state_ = AckState::kImmediate;
+}
+
+void DataTracker::HandleDelayedAckTimerExpiry() {
+  UpdateAckState(AckState::kImmediate, "delayed ack timer expired");
+}
+
+void DataTracker::ObservePacketEnd() {
+  if (ack_state_ == AckState::kBecomingDelayed) {
+    UpdateAckState(AckState::kDelayed, "packet end");
+  }
+}
+
+void DataTracker::UpdateAckState(AckState new_state, absl::string_view reason) {
+  if (new_state != ack_state_) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix_ << "State changed from "
+                         << ToString(ack_state_) << " to "
+                         << ToString(new_state) << " due to " << reason;
+    if (ack_state_ == AckState::kDelayed) {
+      delayed_ack_timer_.Stop();
+    } else if (new_state == AckState::kDelayed) {
+      delayed_ack_timer_.Start();
+    }
+    ack_state_ = new_state;
+  }
+}
+
+absl::string_view DataTracker::ToString(AckState ack_state) {
+  switch (ack_state) {
+    case AckState::kIdle:
+      return "IDLE";
+    case AckState::kBecomingDelayed:
+      return "BECOMING_DELAYED";
+    case AckState::kDelayed:
+      return "DELAYED";
+    case AckState::kImmediate:
+      return "IMMEDIATE";
+  }
+}
+
+HandoverReadinessStatus DataTracker::GetHandoverReadiness() const {
+  HandoverReadinessStatus status;
+  if (!additional_tsn_blocks_.empty()) {
+    status.Add(HandoverUnreadinessReason::kDataTrackerTsnBlocksPending);
+  }
+  return status;
+}
+
+void DataTracker::AddHandoverState(DcSctpSocketHandoverState& state) {
+  state.rx.last_cumulative_acked_tsn = last_cumulative_acked_tsn().value();
+  state.rx.seen_packet = seen_packet_;
+}
+
+void DataTracker::RestoreFromState(const DcSctpSocketHandoverState& state) {
+  // Validate that the component is in pristine state.
+  RTC_DCHECK(additional_tsn_blocks_.empty());
+  RTC_DCHECK(duplicate_tsns_.empty());
+  RTC_DCHECK(!seen_packet_);
+
+  seen_packet_ = state.rx.seen_packet;
+  last_cumulative_acked_tsn_ =
+      tsn_unwrapper_.Unwrap(TSN(state.rx.last_cumulative_acked_tsn));
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/rx/data_tracker.h b/third_party/libwebrtc/net/dcsctp/rx/data_tracker.h
new file mode 100644
index 0000000000..ea077a9b57
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/data_tracker.h
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_RX_DATA_TRACKER_H_
+#define NET_DCSCTP_RX_DATA_TRACKER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <cstdint>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/data_common.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/dcsctp_handover_state.h"
+#include "net/dcsctp/timer/timer.h"
+
+namespace dcsctp {
+
+// Keeps track of received DATA chunks and handles all logic for _when_ to
+// create SACKs and also _how_ to generate them.
+//
+// It only uses TSNs to track delivery and doesn't need to be aware of streams.
+//
+// SACKs are optimally sent every second packet on connections with no packet
+// loss. When packet loss is detected, it's sent for every packet. When SACKs
+// are not sent directly, a timer is used to send a SACK delayed (by RTO/2, or
+// 200ms, whatever is smallest).
+class DataTracker {
+ public:
+  // The maximum number of duplicate TSNs that will be reported in a SACK.
+  static constexpr size_t kMaxDuplicateTsnReported = 20;
+  // The maximum number of gap-ack-blocks that will be reported in a SACK.
+  static constexpr size_t kMaxGapAckBlocksReported = 20;
+
+  // The maximum number of accepted in-flight DATA chunks. This indicates the
+  // maximum difference from this buffer's last cumulative ack TSN, and any
+  // received data. Data received beyond this limit will be dropped, which will
+  // force the transmitter to send data that actually increases the last
+  // cumulative acked TSN.
+  static constexpr uint32_t kMaxAcceptedOutstandingFragments = 100000;
+
+  DataTracker(absl::string_view log_prefix,
+              Timer* delayed_ack_timer,
+              TSN peer_initial_tsn)
+      : log_prefix_(std::string(log_prefix) + "dtrack: "),
+        seen_packet_(false),
+        delayed_ack_timer_(*delayed_ack_timer),
+        last_cumulative_acked_tsn_(
+            tsn_unwrapper_.Unwrap(TSN(*peer_initial_tsn - 1))) {}
+
+  // Indicates if the provided TSN is valid. If this return false, the data
+  // should be dropped and not added to any other buffers, which essentially
+  // means that there is intentional packet loss.
+  bool IsTSNValid(TSN tsn) const;
+
+  // Call for every incoming data chunk. Returns `true` if `tsn` was seen for
+  // the first time, and `false` if it has been seen before (a duplicate `tsn`).
+  bool Observe(TSN tsn,
+               AnyDataChunk::ImmediateAckFlag immediate_ack =
+                   AnyDataChunk::ImmediateAckFlag(false));
+  // Called at the end of processing an SCTP packet.
+  void ObservePacketEnd();
+
+  // Called for incoming FORWARD-TSN/I-FORWARD-TSN chunks
+  void HandleForwardTsn(TSN new_cumulative_ack);
+
+  // Indicates if a SACK should be sent. There may be other reasons to send a
+  // SACK, but if this function indicates so, it should be sent as soon as
+  // possible. Calling this function will make it clear a flag so that if it's
+  // called again, it will probably return false.
+  //
+  // If the delayed ack timer is running, this method will return false _unless_
+  // `also_if_delayed` is set to true. Then it will return true as well.
+  bool ShouldSendAck(bool also_if_delayed = false);
+
+  // Returns the last cumulative ack TSN - the last seen data chunk's TSN
+  // value before any packet loss was detected.
+  TSN last_cumulative_acked_tsn() const {
+    return TSN(last_cumulative_acked_tsn_.Wrap());
+  }
+
+  // Returns true if the received `tsn` would increase the cumulative ack TSN.
+  bool will_increase_cum_ack_tsn(TSN tsn) const;
+
+  // Forces `ShouldSendSack` to return true.
+  void ForceImmediateSack();
+
+  // Note that this will clear `duplicates_`, so every SackChunk that is
+  // consumed must be sent.
+  SackChunk CreateSelectiveAck(size_t a_rwnd);
+
+  void HandleDelayedAckTimerExpiry();
+
+  HandoverReadinessStatus GetHandoverReadiness() const;
+
+  void AddHandoverState(DcSctpSocketHandoverState& state);
+  void RestoreFromState(const DcSctpSocketHandoverState& state);
+
+ private:
+  enum class AckState {
+    // No need to send an ACK.
+    kIdle,
+
+    // Has received data chunks (but not yet end of packet).
+    kBecomingDelayed,
+
+    // Has received data chunks and the end of a packet. Delayed ack timer is
+    // running and a SACK will be sent on expiry, or if DATA is sent, or after
+    // next packet with data.
+    kDelayed,
+
+    // Send a SACK immediately after handling this packet.
+    kImmediate,
+  };
+
+  // Represents ranges of TSNs that have been received that are not directly
+  // following the last cumulative acked TSN. This information is returned to
+  // the sender in the "gap ack blocks" in the SACK chunk. The blocks are always
+  // non-overlapping and non-adjacent.
+  class AdditionalTsnBlocks {
+   public:
+    // Represents an inclusive range of received TSNs, i.e. [first, last].
+    struct TsnRange {
+      TsnRange(UnwrappedTSN first, UnwrappedTSN last)
+          : first(first), last(last) {}
+      UnwrappedTSN first;
+      UnwrappedTSN last;
+    };
+
+    // Adds a TSN to the set. This will try to expand any existing block and
+    // might merge blocks to ensure that all blocks are non-adjacent. If a
+    // current block can't be expanded, a new block is created.
+    //
+    // The return value indicates if `tsn` was added. If false is returned, the
+    // `tsn` was already represented in one of the blocks.
+    bool Add(UnwrappedTSN tsn);
+
+    // Erases all TSNs up to, and including `tsn`. This will remove all blocks
+    // that are completely below `tsn` and may truncate a block where `tsn` is
+    // within that block. In that case, the frontmost block's start TSN will be
+    // the next following tsn after `tsn`.
+    void EraseTo(UnwrappedTSN tsn);
+
+    // Removes the first block. Must not be called on an empty set.
+    void PopFront();
+
+    const std::vector<TsnRange>& blocks() const { return blocks_; }
+
+    bool empty() const { return blocks_.empty(); }
+
+    const TsnRange& front() const { return blocks_.front(); }
+
+   private:
+    // A sorted vector of non-overlapping and non-adjacent blocks.
+    std::vector<TsnRange> blocks_;
+  };
+
+  std::vector<SackChunk::GapAckBlock> CreateGapAckBlocks() const;
+  void UpdateAckState(AckState new_state, absl::string_view reason);
+  static absl::string_view ToString(AckState ack_state);
+
+  const std::string log_prefix_;
+  // If a packet has ever been seen.
+  bool seen_packet_;
+  Timer& delayed_ack_timer_;
+  AckState ack_state_ = AckState::kIdle;
+  UnwrappedTSN::Unwrapper tsn_unwrapper_;
+
+  // All TSNs up until (and including) this value have been seen.
+  UnwrappedTSN last_cumulative_acked_tsn_;
+  // Received TSNs that are not directly following `last_cumulative_acked_tsn_`.
+  AdditionalTsnBlocks additional_tsn_blocks_;
+  std::set<TSN> duplicate_tsns_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_RX_DATA_TRACKER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/rx/data_tracker_test.cc b/third_party/libwebrtc/net/dcsctp/rx/data_tracker_test.cc
new file mode 100644
index 0000000000..f74dd6eb0b
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/data_tracker_test.cc
@@ -0,0 +1,739 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/rx/data_tracker.h"
+
+#include <cstdint>
+#include <initializer_list>
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/common/handover_testing.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/timer/fake_timeout.h"
+#include "net/dcsctp/timer/timer.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+using ::testing::SizeIs;
+using ::testing::UnorderedElementsAre;
+
+constexpr size_t kArwnd = 10000;
+constexpr TSN kInitialTSN(11);
+
+class DataTrackerTest : public testing::Test {
+ protected:
+  DataTrackerTest()
+      : timeout_manager_([this]() { return now_; }),
+        timer_manager_([this](webrtc::TaskQueueBase::DelayPrecision precision) {
+          return timeout_manager_.CreateTimeout(precision);
+        }),
+        timer_(timer_manager_.CreateTimer(
+            "test/delayed_ack",
+            []() { return absl::nullopt; },
+            TimerOptions(DurationMs(0)))),
+        tracker_(
+            std::make_unique<DataTracker>("log: ", timer_.get(), kInitialTSN)) {
+  }
+
+  void Observer(std::initializer_list<uint32_t> tsns,
+                bool expect_as_duplicate = false) {
+    for (const uint32_t tsn : tsns) {
+      if (expect_as_duplicate) {
+        EXPECT_FALSE(
+            tracker_->Observe(TSN(tsn), AnyDataChunk::ImmediateAckFlag(false)));
+      } else {
+        EXPECT_TRUE(
+            tracker_->Observe(TSN(tsn), AnyDataChunk::ImmediateAckFlag(false)));
+      }
+    }
+  }
+
+  void HandoverTracker() {
+    EXPECT_TRUE(tracker_->GetHandoverReadiness().IsReady());
+    DcSctpSocketHandoverState state;
+    tracker_->AddHandoverState(state);
+    g_handover_state_transformer_for_test(&state);
+    tracker_ =
+        std::make_unique<DataTracker>("log: ", timer_.get(), kInitialTSN);
+    tracker_->RestoreFromState(state);
+  }
+
+  TimeMs now_ = TimeMs(0);
+  FakeTimeoutManager timeout_manager_;
+  TimerManager timer_manager_;
+  std::unique_ptr<Timer> timer_;
+  std::unique_ptr<DataTracker> tracker_;
+};
+
+TEST_F(DataTrackerTest, Empty) {
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(sack.gap_ack_blocks(), IsEmpty());
+  EXPECT_THAT(sack.duplicate_tsns(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, ObserverSingleInOrderPacket) {
+  Observer({11});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(11));
+  EXPECT_THAT(sack.gap_ack_blocks(), IsEmpty());
+  EXPECT_THAT(sack.duplicate_tsns(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, ObserverManyInOrderMovesCumulativeTsnAck) {
+  Observer({11, 12, 13});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(13));
+  EXPECT_THAT(sack.gap_ack_blocks(), IsEmpty());
+  EXPECT_THAT(sack.duplicate_tsns(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, ObserveOutOfOrderMovesCumulativeTsnAck) {
+  Observer({12, 13, 14, 11});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(14));
+  EXPECT_THAT(sack.gap_ack_blocks(), IsEmpty());
+  EXPECT_THAT(sack.duplicate_tsns(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, SingleGap) {
+  Observer({12});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(sack.gap_ack_blocks(), ElementsAre(SackChunk::GapAckBlock(2, 2)));
+  EXPECT_THAT(sack.duplicate_tsns(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, ExampleFromRFC4960Section334) {
+  Observer({11, 12, 14, 15, 17});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(12));
+  EXPECT_THAT(sack.gap_ack_blocks(), ElementsAre(SackChunk::GapAckBlock(2, 3),
+                                                 SackChunk::GapAckBlock(5, 5)));
+  EXPECT_THAT(sack.duplicate_tsns(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, AckAlreadyReceivedChunk) {
+  Observer({11});
+  SackChunk sack1 = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack1.cumulative_tsn_ack(), TSN(11));
+  EXPECT_THAT(sack1.gap_ack_blocks(), IsEmpty());
+
+  // Receive old chunk
+  Observer({8}, /*expect_as_duplicate=*/true);
+  SackChunk sack2 = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack2.cumulative_tsn_ack(), TSN(11));
+  EXPECT_THAT(sack2.gap_ack_blocks(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, DoubleSendRetransmittedChunk) {
+  Observer({11, 13, 14, 15});
+  SackChunk sack1 = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack1.cumulative_tsn_ack(), TSN(11));
+  EXPECT_THAT(sack1.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(2, 4)));
+
+  // Fill in the hole.
+  Observer({12, 16, 17, 18});
+  SackChunk sack2 = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack2.cumulative_tsn_ack(), TSN(18));
+  EXPECT_THAT(sack2.gap_ack_blocks(), IsEmpty());
+
+  // Receive chunk 12 again.
+  Observer({12}, /*expect_as_duplicate=*/true);
+  Observer({19, 20, 21});
+  SackChunk sack3 = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack3.cumulative_tsn_ack(), TSN(21));
+  EXPECT_THAT(sack3.gap_ack_blocks(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, ForwardTsnSimple) {
+  // Messages (11, 12, 13), (14, 15) - first message expires.
+  Observer({11, 12, 15});
+
+  tracker_->HandleForwardTsn(TSN(13));
+
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(13));
+  EXPECT_THAT(sack.gap_ack_blocks(), ElementsAre(SackChunk::GapAckBlock(2, 2)));
+}
+
+TEST_F(DataTrackerTest, ForwardTsnSkipsFromGapBlock) {
+  // Messages (11, 12, 13), (14, 15) - first message expires.
+  Observer({11, 12, 14});
+
+  tracker_->HandleForwardTsn(TSN(13));
+
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(14));
+  EXPECT_THAT(sack.gap_ack_blocks(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, ExampleFromRFC3758) {
+  tracker_->HandleForwardTsn(TSN(102));
+
+  Observer({102}, /*expect_as_duplicate=*/true);
+  Observer({104, 105, 107});
+
+  tracker_->HandleForwardTsn(TSN(103));
+
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(105));
+  EXPECT_THAT(sack.gap_ack_blocks(), ElementsAre(SackChunk::GapAckBlock(2, 2)));
+}
+
+TEST_F(DataTrackerTest, EmptyAllAcks) {
+  Observer({11, 13, 14, 15});
+
+  tracker_->HandleForwardTsn(TSN(100));
+
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(100));
+  EXPECT_THAT(sack.gap_ack_blocks(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, SetsArwndCorrectly) {
+  SackChunk sack1 = tracker_->CreateSelectiveAck(/*a_rwnd=*/100);
+  EXPECT_EQ(sack1.a_rwnd(), 100u);
+
+  SackChunk sack2 = tracker_->CreateSelectiveAck(/*a_rwnd=*/101);
+  EXPECT_EQ(sack2.a_rwnd(), 101u);
+}
+
+TEST_F(DataTrackerTest, WillIncreaseCumAckTsn) {
+  EXPECT_EQ(tracker_->last_cumulative_acked_tsn(), TSN(10));
+  EXPECT_FALSE(tracker_->will_increase_cum_ack_tsn(TSN(10)));
+  EXPECT_TRUE(tracker_->will_increase_cum_ack_tsn(TSN(11)));
+  EXPECT_FALSE(tracker_->will_increase_cum_ack_tsn(TSN(12)));
+
+  Observer({11, 12, 13, 14, 15});
+  EXPECT_EQ(tracker_->last_cumulative_acked_tsn(), TSN(15));
+  EXPECT_FALSE(tracker_->will_increase_cum_ack_tsn(TSN(15)));
+  EXPECT_TRUE(tracker_->will_increase_cum_ack_tsn(TSN(16)));
+  EXPECT_FALSE(tracker_->will_increase_cum_ack_tsn(TSN(17)));
+}
+
+TEST_F(DataTrackerTest, ForceShouldSendSackImmediately) {
+  EXPECT_FALSE(tracker_->ShouldSendAck());
+
+  tracker_->ForceImmediateSack();
+
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+}
+
+TEST_F(DataTrackerTest, WillAcceptValidTSNs) {
+  // The initial TSN is always one more than the last, which is our base.
+  TSN last_tsn = TSN(*kInitialTSN - 1);
+  int limit = static_cast<int>(DataTracker::kMaxAcceptedOutstandingFragments);
+
+  for (int i = -limit; i <= limit; ++i) {
+    EXPECT_TRUE(tracker_->IsTSNValid(TSN(*last_tsn + i)));
+  }
+}
+
+TEST_F(DataTrackerTest, WillNotAcceptInvalidTSNs) {
+  // The initial TSN is always one more than the last, which is our base.
+  TSN last_tsn = TSN(*kInitialTSN - 1);
+
+  size_t limit = DataTracker::kMaxAcceptedOutstandingFragments;
+  EXPECT_FALSE(tracker_->IsTSNValid(TSN(*last_tsn + limit + 1)));
+  EXPECT_FALSE(tracker_->IsTSNValid(TSN(*last_tsn - (limit + 1))));
+  EXPECT_FALSE(tracker_->IsTSNValid(TSN(*last_tsn + 0x8000000)));
+  EXPECT_FALSE(tracker_->IsTSNValid(TSN(*last_tsn - 0x8000000)));
+}
+
+TEST_F(DataTrackerTest, ReportSingleDuplicateTsns) {
+  Observer({11, 12});
+  Observer({11}, /*expect_as_duplicate=*/true);
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(12));
+  EXPECT_THAT(sack.gap_ack_blocks(), IsEmpty());
+  EXPECT_THAT(sack.duplicate_tsns(), UnorderedElementsAre(TSN(11)));
+}
+
+TEST_F(DataTrackerTest, ReportMultipleDuplicateTsns) {
+  Observer({11, 12, 13, 14});
+  Observer({12, 13, 12, 13}, /*expect_as_duplicate=*/true);
+  Observer({15, 16});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(16));
+  EXPECT_THAT(sack.gap_ack_blocks(), IsEmpty());
+  EXPECT_THAT(sack.duplicate_tsns(), UnorderedElementsAre(TSN(12), TSN(13)));
+}
+
+TEST_F(DataTrackerTest, ReportDuplicateTsnsInGapAckBlocks) {
+  Observer({11, /*12,*/ 13, 14});
+  Observer({13, 14}, /*expect_as_duplicate=*/true);
+  Observer({15, 16});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(11));
+  EXPECT_THAT(sack.gap_ack_blocks(), ElementsAre(SackChunk::GapAckBlock(2, 5)));
+  EXPECT_THAT(sack.duplicate_tsns(), UnorderedElementsAre(TSN(13), TSN(14)));
+}
+
+TEST_F(DataTrackerTest, ClearsDuplicateTsnsAfterCreatingSack) {
+  Observer({11, 12, 13, 14});
+  Observer({12, 13, 12, 13}, /*expect_as_duplicate=*/true);
+  Observer({15, 16});
+  SackChunk sack1 = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack1.cumulative_tsn_ack(), TSN(16));
+  EXPECT_THAT(sack1.gap_ack_blocks(), IsEmpty());
+  EXPECT_THAT(sack1.duplicate_tsns(), UnorderedElementsAre(TSN(12), TSN(13)));
+
+  Observer({17});
+  SackChunk sack2 = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack2.cumulative_tsn_ack(), TSN(17));
+  EXPECT_THAT(sack2.gap_ack_blocks(), IsEmpty());
+  EXPECT_THAT(sack2.duplicate_tsns(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, LimitsNumberOfDuplicatesReported) {
+  for (size_t i = 0; i < DataTracker::kMaxDuplicateTsnReported + 10; ++i) {
+    TSN tsn(11 + i);
+    tracker_->Observe(tsn, AnyDataChunk::ImmediateAckFlag(false));
+    tracker_->Observe(tsn, AnyDataChunk::ImmediateAckFlag(false));
+  }
+
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_THAT(sack.gap_ack_blocks(), IsEmpty());
+  EXPECT_THAT(sack.duplicate_tsns(),
+              SizeIs(DataTracker::kMaxDuplicateTsnReported));
+}
+
+TEST_F(DataTrackerTest, LimitsNumberOfGapAckBlocksReported) {
+  for (size_t i = 0; i < DataTracker::kMaxGapAckBlocksReported + 10; ++i) {
+    TSN tsn(11 + i * 2);
+    tracker_->Observe(tsn, AnyDataChunk::ImmediateAckFlag(false));
+  }
+
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(11));
+  EXPECT_THAT(sack.gap_ack_blocks(),
+              SizeIs(DataTracker::kMaxGapAckBlocksReported));
+}
+
+TEST_F(DataTrackerTest, SendsSackForFirstPacketObserved) {
+  Observer({11});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+}
+
+TEST_F(DataTrackerTest, SendsSackEverySecondPacketWhenThereIsNoPacketLoss) {
+  Observer({11});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  Observer({12});
+  tracker_->ObservePacketEnd();
+  EXPECT_FALSE(tracker_->ShouldSendAck());
+  EXPECT_TRUE(timer_->is_running());
+  Observer({13});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  Observer({14});
+  tracker_->ObservePacketEnd();
+  EXPECT_FALSE(tracker_->ShouldSendAck());
+  EXPECT_TRUE(timer_->is_running());
+  Observer({15});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+}
+
+TEST_F(DataTrackerTest, SendsSackEveryPacketOnPacketLoss) {
+  Observer({11});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  Observer({13});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  Observer({14});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  Observer({15});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  Observer({16});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  // Fill the hole.
+  Observer({12});
+  tracker_->ObservePacketEnd();
+  EXPECT_FALSE(tracker_->ShouldSendAck());
+  EXPECT_TRUE(timer_->is_running());
+  // Goes back to every second packet
+  Observer({17});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  Observer({18});
+  tracker_->ObservePacketEnd();
+  EXPECT_FALSE(tracker_->ShouldSendAck());
+  EXPECT_TRUE(timer_->is_running());
+}
+
+TEST_F(DataTrackerTest, SendsSackOnDuplicateDataChunks) {
+  Observer({11});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  Observer({11}, /*expect_as_duplicate=*/true);
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  Observer({12});
+  tracker_->ObservePacketEnd();
+  EXPECT_FALSE(tracker_->ShouldSendAck());
+  EXPECT_TRUE(timer_->is_running());
+  // Goes back to every second packet
+  Observer({13});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  // Duplicate again
+  Observer({12}, /*expect_as_duplicate=*/true);
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+}
+
+TEST_F(DataTrackerTest, GapAckBlockAddSingleBlock) {
+  Observer({12});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(sack.gap_ack_blocks(), ElementsAre(SackChunk::GapAckBlock(2, 2)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockAddsAnother) {
+  Observer({12});
+  Observer({14});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(sack.gap_ack_blocks(), ElementsAre(SackChunk::GapAckBlock(2, 2),
+                                                 SackChunk::GapAckBlock(4, 4)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockAddsDuplicate) {
+  Observer({12});
+  Observer({12}, /*expect_as_duplicate=*/true);
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(sack.gap_ack_blocks(), ElementsAre(SackChunk::GapAckBlock(2, 2)));
+  EXPECT_THAT(sack.duplicate_tsns(), ElementsAre(TSN(12)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockExpandsToRight) {
+  Observer({12});
+  Observer({13});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(sack.gap_ack_blocks(), ElementsAre(SackChunk::GapAckBlock(2, 3)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockExpandsToRightWithOther) {
+  Observer({12});
+  Observer({20});
+  Observer({30});
+  Observer({21});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(sack.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(2, 2),    //
+                          SackChunk::GapAckBlock(10, 11),  //
+                          SackChunk::GapAckBlock(20, 20)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockExpandsToLeft) {
+  Observer({13});
+  Observer({12});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(sack.gap_ack_blocks(), ElementsAre(SackChunk::GapAckBlock(2, 3)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockExpandsToLeftWithOther) {
+  Observer({12});
+  Observer({21});
+  Observer({30});
+  Observer({20});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(sack.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(2, 2),    //
+                          SackChunk::GapAckBlock(10, 11),  //
+                          SackChunk::GapAckBlock(20, 20)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockExpandsToLRightAndMerges) {
+  Observer({12});
+  Observer({20});
+  Observer({22});
+  Observer({30});
+  Observer({21});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(sack.gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(2, 2),    //
+                          SackChunk::GapAckBlock(10, 12),  //
+                          SackChunk::GapAckBlock(20, 20)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockMergesManyBlocksIntoOne) {
+  Observer({22});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(12, 12)));
+  Observer({30});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(12, 12),  //
+                          SackChunk::GapAckBlock(20, 20)));
+  Observer({24});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(12, 12),  //
+                          SackChunk::GapAckBlock(14, 14),  //
+                          SackChunk::GapAckBlock(20, 20)));
+  Observer({28});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(12, 12),  //
+                          SackChunk::GapAckBlock(14, 14),  //
+                          SackChunk::GapAckBlock(18, 18),  //
+                          SackChunk::GapAckBlock(20, 20)));
+  Observer({26});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(12, 12),  //
+                          SackChunk::GapAckBlock(14, 14),  //
+                          SackChunk::GapAckBlock(16, 16),  //
+                          SackChunk::GapAckBlock(18, 18),  //
+                          SackChunk::GapAckBlock(20, 20)));
+  Observer({29});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(12, 12),  //
+                          SackChunk::GapAckBlock(14, 14),  //
+                          SackChunk::GapAckBlock(16, 16),  //
+                          SackChunk::GapAckBlock(18, 20)));
+  Observer({23});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(12, 14),  //
+                          SackChunk::GapAckBlock(16, 16),  //
+                          SackChunk::GapAckBlock(18, 20)));
+  Observer({27});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(12, 14),  //
+                          SackChunk::GapAckBlock(16, 20)));
+
+  Observer({25});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(12, 20)));
+  Observer({20});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(10, 10),  //
+                          SackChunk::GapAckBlock(12, 20)));
+  Observer({32});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(10, 10),  //
+                          SackChunk::GapAckBlock(12, 20),  //
+                          SackChunk::GapAckBlock(22, 22)));
+  Observer({21});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(10, 20),  //
+                          SackChunk::GapAckBlock(22, 22)));
+  Observer({31});
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(10, 22)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveBeforeCumAckTsn) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+
+  tracker_->HandleForwardTsn(TSN(8));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(10));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(2, 4),  //
+                          SackChunk::GapAckBlock(10, 12),
+                          SackChunk::GapAckBlock(20, 21)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveBeforeFirstBlock) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+
+  tracker_->HandleForwardTsn(TSN(11));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(14));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(6, 8),  //
+                          SackChunk::GapAckBlock(16, 17)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveAtBeginningOfFirstBlock) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+
+  tracker_->HandleForwardTsn(TSN(12));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(14));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(6, 8),  //
+                          SackChunk::GapAckBlock(16, 17)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveAtMiddleOfFirstBlock) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+  tracker_->HandleForwardTsn(TSN(13));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(14));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(6, 8),  //
+                          SackChunk::GapAckBlock(16, 17)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveAtEndOfFirstBlock) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+  tracker_->HandleForwardTsn(TSN(14));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(14));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(6, 8),  //
+                          SackChunk::GapAckBlock(16, 17)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveRightAfterFirstBlock) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+
+  tracker_->HandleForwardTsn(TSN(18));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(18));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(2, 4),  //
+                          SackChunk::GapAckBlock(12, 13)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveRightBeforeSecondBlock) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+
+  tracker_->HandleForwardTsn(TSN(19));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(22));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(8, 9)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveRightAtStartOfSecondBlock) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+
+  tracker_->HandleForwardTsn(TSN(20));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(22));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(8, 9)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveRightAtMiddleOfSecondBlock) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+
+  tracker_->HandleForwardTsn(TSN(21));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(22));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(8, 9)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveRightAtEndOfSecondBlock) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+
+  tracker_->HandleForwardTsn(TSN(22));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(22));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(),
+              ElementsAre(SackChunk::GapAckBlock(8, 9)));
+}
+
+TEST_F(DataTrackerTest, GapAckBlockRemoveeFarAfterAllBlocks) {
+  Observer({12, 13, 14, 20, 21, 22, 30, 31});
+
+  tracker_->HandleForwardTsn(TSN(40));
+  EXPECT_EQ(tracker_->CreateSelectiveAck(kArwnd).cumulative_tsn_ack(), TSN(40));
+  EXPECT_THAT(tracker_->CreateSelectiveAck(kArwnd).gap_ack_blocks(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest, HandoverEmpty) {
+  HandoverTracker();
+  Observer({11});
+  SackChunk sack = tracker_->CreateSelectiveAck(kArwnd);
+  EXPECT_EQ(sack.cumulative_tsn_ack(), TSN(11));
+  EXPECT_THAT(sack.gap_ack_blocks(), IsEmpty());
+}
+
+TEST_F(DataTrackerTest,
+       HandoverWhileSendingSackEverySecondPacketWhenThereIsNoPacketLoss) {
+  Observer({11});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+
+  HandoverTracker();
+
+  Observer({12});
+  tracker_->ObservePacketEnd();
+  EXPECT_FALSE(tracker_->ShouldSendAck());
+  Observer({13});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+  Observer({14});
+  tracker_->ObservePacketEnd();
+  EXPECT_FALSE(tracker_->ShouldSendAck());
+  EXPECT_TRUE(timer_->is_running());
+  Observer({15});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_FALSE(timer_->is_running());
+}
+
+TEST_F(DataTrackerTest, HandoverWhileSendingSackEveryPacketOnPacketLoss) {
+  Observer({11});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  Observer({13});
+  EXPECT_EQ(tracker_->GetHandoverReadiness(),
+            HandoverReadinessStatus().Add(
+                HandoverUnreadinessReason::kDataTrackerTsnBlocksPending));
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  Observer({14});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  EXPECT_EQ(tracker_->GetHandoverReadiness(),
+            HandoverReadinessStatus(
+                HandoverUnreadinessReason::kDataTrackerTsnBlocksPending));
+  Observer({15});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  Observer({16});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+  // Fill the hole.
+  Observer({12});
+  tracker_->ObservePacketEnd();
+  EXPECT_FALSE(tracker_->ShouldSendAck());
+  // Goes back to every second packet
+  Observer({17});
+  tracker_->ObservePacketEnd();
+  EXPECT_TRUE(tracker_->ShouldSendAck());
+
+  HandoverTracker();
+
+  Observer({18});
+  tracker_->ObservePacketEnd();
+  EXPECT_FALSE(tracker_->ShouldSendAck());
+  EXPECT_TRUE(timer_->is_running());
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/rx/interleaved_reassembly_streams.cc b/third_party/libwebrtc/net/dcsctp/rx/interleaved_reassembly_streams.cc
new file mode 100644
index 0000000000..8b316de676
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/interleaved_reassembly_streams.cc
@@ -0,0 +1,272 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/rx/interleaved_reassembly_streams.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <functional>
+#include <iterator>
+#include <map>
+#include <numeric>
+#include <unordered_map>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/types.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+
+InterleavedReassemblyStreams::InterleavedReassemblyStreams(
+    absl::string_view log_prefix,
+    OnAssembledMessage on_assembled_message)
+    : log_prefix_(log_prefix), on_assembled_message_(on_assembled_message) {}
+
+size_t InterleavedReassemblyStreams::Stream::TryToAssembleMessage(
+    UnwrappedMID mid) {
+  std::map<UnwrappedMID, ChunkMap>::const_iterator it =
+      chunks_by_mid_.find(mid);
+  if (it == chunks_by_mid_.end()) {
+    RTC_DLOG(LS_VERBOSE) << parent_.log_prefix_ << "TryToAssembleMessage "
+                         << *mid.Wrap() << " - no chunks";
+    return 0;
+  }
+  const ChunkMap& chunks = it->second;
+  if (!chunks.begin()->second.second.is_beginning ||
+      !chunks.rbegin()->second.second.is_end) {
+    RTC_DLOG(LS_VERBOSE) << parent_.log_prefix_ << "TryToAssembleMessage "
+                         << *mid.Wrap() << "- missing beginning or end";
+    return 0;
+  }
+  int64_t fsn_diff = *chunks.rbegin()->first - *chunks.begin()->first;
+  if (fsn_diff != (static_cast<int64_t>(chunks.size()) - 1)) {
+    RTC_DLOG(LS_VERBOSE) << parent_.log_prefix_ << "TryToAssembleMessage "
+                         << *mid.Wrap() << "- not all chunks exist (have "
+                         << chunks.size() << ", expect " << (fsn_diff + 1)
+                         << ")";
+    return 0;
+  }
+
+  size_t removed_bytes = AssembleMessage(chunks);
+  RTC_DLOG(LS_VERBOSE) << parent_.log_prefix_ << "TryToAssembleMessage "
+                       << *mid.Wrap() << " - succeeded and removed "
+                       << removed_bytes;
+
+  chunks_by_mid_.erase(mid);
+  return removed_bytes;
+}
+
+size_t InterleavedReassemblyStreams::Stream::AssembleMessage(
+    const ChunkMap& tsn_chunks) {
+  size_t count = tsn_chunks.size();
+  if (count == 1) {
+    // Fast path - zero-copy
+    const Data& data = tsn_chunks.begin()->second.second;
+    size_t payload_size = data.size();
+    UnwrappedTSN tsns[1] = {tsn_chunks.begin()->second.first};
+    DcSctpMessage message(data.stream_id, data.ppid, std::move(data.payload));
+    parent_.on_assembled_message_(tsns, std::move(message));
+    return payload_size;
+  }
+
+  // Slow path - will need to concatenate the payload.
+  std::vector<UnwrappedTSN> tsns;
+  tsns.reserve(count);
+
+  std::vector<uint8_t> payload;
+  size_t payload_size = absl::c_accumulate(
+      tsn_chunks, 0,
+      [](size_t v, const auto& p) { return v + p.second.second.size(); });
+  payload.reserve(payload_size);
+
+  for (auto& item : tsn_chunks) {
+    const UnwrappedTSN tsn = item.second.first;
+    const Data& data = item.second.second;
+    tsns.push_back(tsn);
+    payload.insert(payload.end(), data.payload.begin(), data.payload.end());
+  }
+
+  const Data& data = tsn_chunks.begin()->second.second;
+
+  DcSctpMessage message(data.stream_id, data.ppid, std::move(payload));
+  parent_.on_assembled_message_(tsns, std::move(message));
+  return payload_size;
+}
+
+size_t InterleavedReassemblyStreams::Stream::EraseTo(MID message_id) {
+  UnwrappedMID unwrapped_mid = mid_unwrapper_.Unwrap(message_id);
+
+  size_t removed_bytes = 0;
+  auto it = chunks_by_mid_.begin();
+  while (it != chunks_by_mid_.end() && it->first <= unwrapped_mid) {
+    removed_bytes += absl::c_accumulate(
+        it->second, 0,
+        [](size_t r2, const auto& q) { return r2 + q.second.second.size(); });
+    it = chunks_by_mid_.erase(it);
+  }
+
+  if (!stream_id_.unordered) {
+    // For ordered streams, erasing a message might suddenly unblock that queue
+    // and allow it to deliver any following received messages.
+    if (unwrapped_mid >= next_mid_) {
+      next_mid_ = unwrapped_mid.next_value();
+    }
+
+    removed_bytes += TryToAssembleMessages();
+  }
+
+  return removed_bytes;
+}
+
+int InterleavedReassemblyStreams::Stream::Add(UnwrappedTSN tsn, Data data) {
+  RTC_DCHECK_EQ(*data.is_unordered, *stream_id_.unordered);
+  RTC_DCHECK_EQ(*data.stream_id, *stream_id_.stream_id);
+  int queued_bytes = data.size();
+  UnwrappedMID mid = mid_unwrapper_.Unwrap(data.message_id);
+  FSN fsn = data.fsn;
+  auto [unused, inserted] =
+      chunks_by_mid_[mid].emplace(fsn, std::make_pair(tsn, std::move(data)));
+  if (!inserted) {
+    return 0;
+  }
+
+  if (stream_id_.unordered) {
+    queued_bytes -= TryToAssembleMessage(mid);
+  } else {
+    if (mid == next_mid_) {
+      queued_bytes -= TryToAssembleMessages();
+    }
+  }
+
+  return queued_bytes;
+}
+
+size_t InterleavedReassemblyStreams::Stream::TryToAssembleMessages() {
+  size_t removed_bytes = 0;
+
+  for (;;) {
+    size_t removed_bytes_this_iter = TryToAssembleMessage(next_mid_);
+    if (removed_bytes_this_iter == 0) {
+      break;
+    }
+
+    removed_bytes += removed_bytes_this_iter;
+    next_mid_.Increment();
+  }
+  return removed_bytes;
+}
+
+void InterleavedReassemblyStreams::Stream::AddHandoverState(
+    DcSctpSocketHandoverState& state) const {
+  if (stream_id_.unordered) {
+    DcSctpSocketHandoverState::UnorderedStream state_stream;
+    state_stream.id = stream_id_.stream_id.value();
+    state.rx.unordered_streams.push_back(std::move(state_stream));
+  } else {
+    DcSctpSocketHandoverState::OrderedStream state_stream;
+    state_stream.id = stream_id_.stream_id.value();
+    state_stream.next_ssn = next_mid_.Wrap().value();
+    state.rx.ordered_streams.push_back(std::move(state_stream));
+  }
+}
+
+InterleavedReassemblyStreams::Stream&
+InterleavedReassemblyStreams::GetOrCreateStream(const FullStreamId& stream_id) {
+  auto it = streams_.find(stream_id);
+  if (it == streams_.end()) {
+    it =
+        streams_
+            .emplace(std::piecewise_construct, std::forward_as_tuple(stream_id),
+                     std::forward_as_tuple(stream_id, this))
+            .first;
+  }
+  return it->second;
+}
+
+int InterleavedReassemblyStreams::Add(UnwrappedTSN tsn, Data data) {
+  return GetOrCreateStream(FullStreamId(data.is_unordered, data.stream_id))
+      .Add(tsn, std::move(data));
+}
+
+size_t InterleavedReassemblyStreams::HandleForwardTsn(
+    UnwrappedTSN new_cumulative_ack_tsn,
+    rtc::ArrayView<const AnyForwardTsnChunk::SkippedStream> skipped_streams) {
+  size_t removed_bytes = 0;
+  for (const auto& skipped : skipped_streams) {
+    removed_bytes +=
+        GetOrCreateStream(FullStreamId(skipped.unordered, skipped.stream_id))
+            .EraseTo(skipped.message_id);
+  }
+  return removed_bytes;
+}
+
+void InterleavedReassemblyStreams::ResetStreams(
+    rtc::ArrayView<const StreamID> stream_ids) {
+  if (stream_ids.empty()) {
+    for (auto& entry : streams_) {
+      entry.second.Reset();
+    }
+  } else {
+    for (StreamID stream_id : stream_ids) {
+      GetOrCreateStream(FullStreamId(IsUnordered(true), stream_id)).Reset();
+      GetOrCreateStream(FullStreamId(IsUnordered(false), stream_id)).Reset();
+    }
+  }
+}
+
+HandoverReadinessStatus InterleavedReassemblyStreams::GetHandoverReadiness()
+    const {
+  HandoverReadinessStatus status;
+  for (const auto& [stream_id, stream] : streams_) {
+    if (stream.has_unassembled_chunks()) {
+      status.Add(
+          stream_id.unordered
+              ? HandoverUnreadinessReason::kUnorderedStreamHasUnassembledChunks
+              : HandoverUnreadinessReason::kOrderedStreamHasUnassembledChunks);
+      break;
+    }
+  }
+  return status;
+}
+
+void InterleavedReassemblyStreams::AddHandoverState(
+    DcSctpSocketHandoverState& state) {
+  for (const auto& [unused, stream] : streams_) {
+    stream.AddHandoverState(state);
+  }
+}
+
+void InterleavedReassemblyStreams::RestoreFromState(
+    const DcSctpSocketHandoverState& state) {
+  // Validate that the component is in pristine state.
+  RTC_DCHECK(streams_.empty());
+
+  for (const DcSctpSocketHandoverState::OrderedStream& state :
+       state.rx.ordered_streams) {
+    FullStreamId stream_id(IsUnordered(false), StreamID(state.id));
+    streams_.emplace(
+        std::piecewise_construct, std::forward_as_tuple(stream_id),
+        std::forward_as_tuple(stream_id, this, MID(state.next_ssn)));
+  }
+  for (const DcSctpSocketHandoverState::UnorderedStream& state :
+       state.rx.unordered_streams) {
+    FullStreamId stream_id(IsUnordered(true), StreamID(state.id));
+    streams_.emplace(std::piecewise_construct, std::forward_as_tuple(stream_id),
+                     std::forward_as_tuple(stream_id, this));
+  }
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/rx/interleaved_reassembly_streams.h b/third_party/libwebrtc/net/dcsctp/rx/interleaved_reassembly_streams.h
new file mode 100644
index 0000000000..a7b67707e9
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/interleaved_reassembly_streams.h
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_RX_INTERLEAVED_REASSEMBLY_STREAMS_H_
+#define NET_DCSCTP_RX_INTERLEAVED_REASSEMBLY_STREAMS_H_
+
+#include <cstdint>
+#include <map>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/rx/reassembly_streams.h"
+
+namespace dcsctp {
+
+// Handles reassembly of incoming data when interleaved message sending is
+// enabled on the association, i.e. when RFC8260 is in use.
+class InterleavedReassemblyStreams : public ReassemblyStreams {
+ public:
+  InterleavedReassemblyStreams(absl::string_view log_prefix,
+                               OnAssembledMessage on_assembled_message);
+
+  int Add(UnwrappedTSN tsn, Data data) override;
+
+  size_t HandleForwardTsn(
+      UnwrappedTSN new_cumulative_ack_tsn,
+      rtc::ArrayView<const AnyForwardTsnChunk::SkippedStream> skipped_streams)
+      override;
+
+  void ResetStreams(rtc::ArrayView<const StreamID> stream_ids) override;
+
+  HandoverReadinessStatus GetHandoverReadiness() const override;
+  void AddHandoverState(DcSctpSocketHandoverState& state) override;
+  void RestoreFromState(const DcSctpSocketHandoverState& state) override;
+
+ private:
+  struct FullStreamId {
+    const IsUnordered unordered;
+    const StreamID stream_id;
+
+    FullStreamId(IsUnordered unordered, StreamID stream_id)
+        : unordered(unordered), stream_id(stream_id) {}
+
+    friend bool operator<(FullStreamId a, FullStreamId b) {
+      return a.unordered < b.unordered ||
+             (!(a.unordered < b.unordered) && (a.stream_id < b.stream_id));
+    }
+  };
+
+  class Stream {
+   public:
+    Stream(FullStreamId stream_id,
+           InterleavedReassemblyStreams* parent,
+           MID next_mid = MID(0))
+        : stream_id_(stream_id),
+          parent_(*parent),
+          next_mid_(mid_unwrapper_.Unwrap(next_mid)) {}
+    int Add(UnwrappedTSN tsn, Data data);
+    size_t EraseTo(MID message_id);
+    void Reset() {
+      mid_unwrapper_.Reset();
+      next_mid_ = mid_unwrapper_.Unwrap(MID(0));
+    }
+    bool has_unassembled_chunks() const { return !chunks_by_mid_.empty(); }
+    void AddHandoverState(DcSctpSocketHandoverState& state) const;
+
+   private:
+    using ChunkMap = std::map<FSN, std::pair<UnwrappedTSN, Data>>;
+
+    // Try to assemble one message identified by `mid`.
+    // Returns the number of bytes assembled if a message was assembled.
+    size_t TryToAssembleMessage(UnwrappedMID mid);
+    size_t AssembleMessage(const ChunkMap& tsn_chunks);
+    // Try to assemble one or several messages in order from the stream.
+    // Returns the number of bytes assembled if one or more messages were
+    // assembled.
+    size_t TryToAssembleMessages();
+
+    const FullStreamId stream_id_;
+    InterleavedReassemblyStreams& parent_;
+    std::map<UnwrappedMID, ChunkMap> chunks_by_mid_;
+    UnwrappedMID::Unwrapper mid_unwrapper_;
+    UnwrappedMID next_mid_;
+  };
+
+  Stream& GetOrCreateStream(const FullStreamId& stream_id);
+
+  const std::string log_prefix_;
+
+  // Callback for when a message has been assembled.
+  const OnAssembledMessage on_assembled_message_;
+
+  // All unordered and ordered streams, managing not-yet-assembled data.
+  std::map<FullStreamId, Stream> streams_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_RX_INTERLEAVED_REASSEMBLY_STREAMS_H_
diff --git a/third_party/libwebrtc/net/dcsctp/rx/interleaved_reassembly_streams_test.cc b/third_party/libwebrtc/net/dcsctp/rx/interleaved_reassembly_streams_test.cc
new file mode 100644
index 0000000000..df4024ed60
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/interleaved_reassembly_streams_test.cc
@@ -0,0 +1,154 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/rx/interleaved_reassembly_streams.h"
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/rx/reassembly_streams.h"
+#include "net/dcsctp/testing/data_generator.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::MockFunction;
+using ::testing::NiceMock;
+
+class InterleavedReassemblyStreamsTest : public testing::Test {
+ protected:
+  UnwrappedTSN tsn(uint32_t value) { return tsn_.Unwrap(TSN(value)); }
+
+  InterleavedReassemblyStreamsTest() {}
+  DataGenerator gen_;
+  UnwrappedTSN::Unwrapper tsn_;
+};
+
+TEST_F(InterleavedReassemblyStreamsTest,
+       AddUnorderedMessageReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  InterleavedReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Unordered({1}, "B")), 1);
+  EXPECT_EQ(streams.Add(tsn(2), gen_.Unordered({2, 3, 4})), 3);
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Unordered({5, 6})), 2);
+  // Adding the end fragment should make it empty again.
+  EXPECT_EQ(streams.Add(tsn(4), gen_.Unordered({7}, "E")), -6);
+}
+
+TEST_F(InterleavedReassemblyStreamsTest,
+       AddSimpleOrderedMessageReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  InterleavedReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  EXPECT_EQ(streams.Add(tsn(2), gen_.Ordered({2, 3, 4})), 3);
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+  EXPECT_EQ(streams.Add(tsn(4), gen_.Ordered({7}, "E")), -6);
+}
+
+TEST_F(InterleavedReassemblyStreamsTest,
+       AddMoreComplexOrderedMessageReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  InterleavedReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  Data late = gen_.Ordered({2, 3, 4});
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+  EXPECT_EQ(streams.Add(tsn(4), gen_.Ordered({7}, "E")), 1);
+
+  EXPECT_EQ(streams.Add(tsn(5), gen_.Ordered({1}, "BE")), 1);
+  EXPECT_EQ(streams.Add(tsn(6), gen_.Ordered({5, 6}, "B")), 2);
+  EXPECT_EQ(streams.Add(tsn(7), gen_.Ordered({7}, "E")), 1);
+  EXPECT_EQ(streams.Add(tsn(2), std::move(late)), -8);
+}
+
+TEST_F(InterleavedReassemblyStreamsTest,
+       DeleteUnorderedMessageReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  InterleavedReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Unordered({1}, "B")), 1);
+  EXPECT_EQ(streams.Add(tsn(2), gen_.Unordered({2, 3, 4})), 3);
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Unordered({5, 6})), 2);
+
+  IForwardTsnChunk::SkippedStream skipped[] = {
+      IForwardTsnChunk::SkippedStream(IsUnordered(true), StreamID(1), MID(0))};
+  EXPECT_EQ(streams.HandleForwardTsn(tsn(3), skipped), 6u);
+}
+
+TEST_F(InterleavedReassemblyStreamsTest,
+       DeleteSimpleOrderedMessageReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  InterleavedReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  EXPECT_EQ(streams.Add(tsn(2), gen_.Ordered({2, 3, 4})), 3);
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+
+  IForwardTsnChunk::SkippedStream skipped[] = {
+      IForwardTsnChunk::SkippedStream(IsUnordered(false), StreamID(1), MID(0))};
+  EXPECT_EQ(streams.HandleForwardTsn(tsn(3), skipped), 6u);
+}
+
+TEST_F(InterleavedReassemblyStreamsTest,
+       DeleteManyOrderedMessagesReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  InterleavedReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  gen_.Ordered({2, 3, 4});
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+  EXPECT_EQ(streams.Add(tsn(4), gen_.Ordered({7}, "E")), 1);
+
+  EXPECT_EQ(streams.Add(tsn(5), gen_.Ordered({1}, "BE")), 1);
+  EXPECT_EQ(streams.Add(tsn(6), gen_.Ordered({5, 6}, "B")), 2);
+  EXPECT_EQ(streams.Add(tsn(7), gen_.Ordered({7}, "E")), 1);
+
+  // Expire all three messages
+  IForwardTsnChunk::SkippedStream skipped[] = {
+      IForwardTsnChunk::SkippedStream(IsUnordered(false), StreamID(1), MID(2))};
+  EXPECT_EQ(streams.HandleForwardTsn(tsn(8), skipped), 8u);
+}
+
+TEST_F(InterleavedReassemblyStreamsTest,
+       DeleteOrderedMessageDelivesTwoReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  InterleavedReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  gen_.Ordered({2, 3, 4});
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+  EXPECT_EQ(streams.Add(tsn(4), gen_.Ordered({7}, "E")), 1);
+
+  EXPECT_EQ(streams.Add(tsn(5), gen_.Ordered({1}, "BE")), 1);
+  EXPECT_EQ(streams.Add(tsn(6), gen_.Ordered({5, 6}, "B")), 2);
+  EXPECT_EQ(streams.Add(tsn(7), gen_.Ordered({7}, "E")), 1);
+
+  // The first ordered message expire, and the following two are delivered.
+  IForwardTsnChunk::SkippedStream skipped[] = {
+      IForwardTsnChunk::SkippedStream(IsUnordered(false), StreamID(1), MID(0))};
+  EXPECT_EQ(streams.HandleForwardTsn(tsn(4), skipped), 8u);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/rx/reassembly_queue.cc b/third_party/libwebrtc/net/dcsctp/rx/reassembly_queue.cc
new file mode 100644
index 0000000000..f72c5cb8c1
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/reassembly_queue.cc
@@ -0,0 +1,312 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/rx/reassembly_queue.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/common/str_join.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/reconfiguration_response_parameter.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/rx/interleaved_reassembly_streams.h"
+#include "net/dcsctp/rx/reassembly_streams.h"
+#include "net/dcsctp/rx/traditional_reassembly_streams.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+namespace {
+std::unique_ptr<ReassemblyStreams> CreateStreams(
+    absl::string_view log_prefix,
+    ReassemblyStreams::OnAssembledMessage on_assembled_message,
+    bool use_message_interleaving) {
+  if (use_message_interleaving) {
+    return std::make_unique<InterleavedReassemblyStreams>(
+        log_prefix, std::move(on_assembled_message));
+  }
+  return std::make_unique<TraditionalReassemblyStreams>(
+      log_prefix, std::move(on_assembled_message));
+}
+}  // namespace
+
+ReassemblyQueue::ReassemblyQueue(absl::string_view log_prefix,
+                                 TSN peer_initial_tsn,
+                                 size_t max_size_bytes,
+                                 bool use_message_interleaving)
+    : log_prefix_(std::string(log_prefix) + "reasm: "),
+      max_size_bytes_(max_size_bytes),
+      watermark_bytes_(max_size_bytes * kHighWatermarkLimit),
+      last_assembled_tsn_watermark_(
+          tsn_unwrapper_.Unwrap(TSN(*peer_initial_tsn - 1))),
+      last_completed_reset_req_seq_nbr_(ReconfigRequestSN(0)),
+      streams_(CreateStreams(
+          log_prefix_,
+          [this](rtc::ArrayView<const UnwrappedTSN> tsns,
+                 DcSctpMessage message) {
+            AddReassembledMessage(tsns, std::move(message));
+          },
+          use_message_interleaving)) {}
+
+void ReassemblyQueue::Add(TSN tsn, Data data) {
+  RTC_DCHECK(IsConsistent());
+  RTC_DLOG(LS_VERBOSE) << log_prefix_ << "added tsn=" << *tsn
+                       << ", stream=" << *data.stream_id << ":"
+                       << *data.message_id << ":" << *data.fsn << ", type="
+                       << (data.is_beginning && data.is_end ? "complete"
+                           : data.is_beginning              ? "first"
+                           : data.is_end                    ? "last"
+                                                            : "middle");
+
+  UnwrappedTSN unwrapped_tsn = tsn_unwrapper_.Unwrap(tsn);
+
+  if (unwrapped_tsn <= last_assembled_tsn_watermark_ ||
+      delivered_tsns_.find(unwrapped_tsn) != delivered_tsns_.end()) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix_
+                         << "Chunk has already been delivered - skipping";
+    return;
+  }
+
+  // If a stream reset has been received with a "sender's last assigned tsn" in
+  // the future, the socket is in "deferred reset processing" mode and must
+  // buffer chunks until it's exited.
+  if (deferred_reset_streams_.has_value() &&
+      unwrapped_tsn >
+          tsn_unwrapper_.Unwrap(
+              deferred_reset_streams_->req.sender_last_assigned_tsn())) {
+    RTC_DLOG(LS_VERBOSE)
+        << log_prefix_ << "Deferring chunk with tsn=" << *tsn
+        << " until cum_ack_tsn="
+        << *deferred_reset_streams_->req.sender_last_assigned_tsn();
+    // https://tools.ietf.org/html/rfc6525#section-5.2.2
+    // "In this mode, any data arriving with a TSN larger than the
+    // Sender's Last Assigned TSN for the affected stream(s) MUST be queued
+    // locally and held until the cumulative acknowledgment point reaches the
+    // Sender's Last Assigned TSN."
+    queued_bytes_ += data.size();
+    deferred_reset_streams_->deferred_chunks.emplace_back(
+        std::make_pair(tsn, std::move(data)));
+  } else {
+    queued_bytes_ += streams_->Add(unwrapped_tsn, std::move(data));
+  }
+
+  // https://tools.ietf.org/html/rfc4960#section-6.9
+  // "Note: If the data receiver runs out of buffer space while still
+  // waiting for more fragments to complete the reassembly of the message, it
+  // should dispatch part of its inbound message through a partial delivery
+  // API (see Section 10), freeing some of its receive buffer space so that
+  // the rest of the message may be received."
+
+  // TODO(boivie): Support EOR flag and partial delivery?
+  RTC_DCHECK(IsConsistent());
+}
+
+ReconfigurationResponseParameter::Result ReassemblyQueue::ResetStreams(
+    const OutgoingSSNResetRequestParameter& req,
+    TSN cum_tsn_ack) {
+  RTC_DCHECK(IsConsistent());
+  if (deferred_reset_streams_.has_value()) {
+    // In deferred mode already.
+    return ReconfigurationResponseParameter::Result::kInProgress;
+  } else if (req.request_sequence_number() <=
+             last_completed_reset_req_seq_nbr_) {
+    // Already performed at some time previously.
+    return ReconfigurationResponseParameter::Result::kSuccessPerformed;
+  }
+
+  UnwrappedTSN sla_tsn = tsn_unwrapper_.Unwrap(req.sender_last_assigned_tsn());
+  UnwrappedTSN unwrapped_cum_tsn_ack = tsn_unwrapper_.Unwrap(cum_tsn_ack);
+
+  // https://tools.ietf.org/html/rfc6525#section-5.2.2
+  // "If the Sender's Last Assigned TSN is greater than the
+  // cumulative acknowledgment point, then the endpoint MUST enter "deferred
+  // reset processing"."
+  if (sla_tsn > unwrapped_cum_tsn_ack) {
+    RTC_DLOG(LS_VERBOSE)
+        << log_prefix_
+        << "Entering deferred reset processing mode until cum_tsn_ack="
+        << *req.sender_last_assigned_tsn();
+    deferred_reset_streams_ = absl::make_optional<DeferredResetStreams>(req);
+    return ReconfigurationResponseParameter::Result::kInProgress;
+  }
+
+  // https://tools.ietf.org/html/rfc6525#section-5.2.2
+  // "... streams MUST be reset to 0 as the next expected SSN."
+  streams_->ResetStreams(req.stream_ids());
+  last_completed_reset_req_seq_nbr_ = req.request_sequence_number();
+  RTC_DCHECK(IsConsistent());
+  return ReconfigurationResponseParameter::Result::kSuccessPerformed;
+}
+
+bool ReassemblyQueue::MaybeResetStreamsDeferred(TSN cum_ack_tsn) {
+  RTC_DCHECK(IsConsistent());
+  if (deferred_reset_streams_.has_value()) {
+    UnwrappedTSN unwrapped_cum_ack_tsn = tsn_unwrapper_.Unwrap(cum_ack_tsn);
+    UnwrappedTSN unwrapped_sla_tsn = tsn_unwrapper_.Unwrap(
+        deferred_reset_streams_->req.sender_last_assigned_tsn());
+    if (unwrapped_cum_ack_tsn >= unwrapped_sla_tsn) {
+      RTC_DLOG(LS_VERBOSE) << log_prefix_
+                           << "Leaving deferred reset processing with tsn="
+                           << *cum_ack_tsn << ", feeding back "
+                           << deferred_reset_streams_->deferred_chunks.size()
+                           << " chunks";
+      // https://tools.ietf.org/html/rfc6525#section-5.2.2
+      // "... streams MUST be reset to 0 as the next expected SSN."
+      streams_->ResetStreams(deferred_reset_streams_->req.stream_ids());
+      std::vector<std::pair<TSN, Data>> deferred_chunks =
+          std::move(deferred_reset_streams_->deferred_chunks);
+      // The response will not be sent now, but as a reply to the retried
+      // request, which will come as "in progress" has been sent prior.
+      last_completed_reset_req_seq_nbr_ =
+          deferred_reset_streams_->req.request_sequence_number();
+      deferred_reset_streams_ = absl::nullopt;
+
+      // https://tools.ietf.org/html/rfc6525#section-5.2.2
+      // "Any queued TSNs (queued at step E2) MUST now be released and processed
+      // normally."
+      for (auto& [tsn, data] : deferred_chunks) {
+        queued_bytes_ -= data.size();
+        Add(tsn, std::move(data));
+      }
+
+      RTC_DCHECK(IsConsistent());
+      return true;
+    } else {
+      RTC_DLOG(LS_VERBOSE) << "Staying in deferred reset processing. tsn="
+                           << *cum_ack_tsn;
+    }
+  }
+
+  return false;
+}
+
+std::vector<DcSctpMessage> ReassemblyQueue::FlushMessages() {
+  std::vector<DcSctpMessage> ret;
+  reassembled_messages_.swap(ret);
+  return ret;
+}
+
+void ReassemblyQueue::AddReassembledMessage(
+    rtc::ArrayView<const UnwrappedTSN> tsns,
+    DcSctpMessage message) {
+  RTC_DLOG(LS_VERBOSE) << log_prefix_ << "Assembled message from TSN=["
+                       << StrJoin(tsns, ",",
+                                  [](rtc::StringBuilder& sb, UnwrappedTSN tsn) {
+                                    sb << *tsn.Wrap();
+                                  })
+                       << "], message; stream_id=" << *message.stream_id()
+                       << ", ppid=" << *message.ppid()
+                       << ", payload=" << message.payload().size() << " bytes";
+
+  for (const UnwrappedTSN tsn : tsns) {
+    if (tsn <= last_assembled_tsn_watermark_) {
+      // This can be provoked by a misbehaving peer by sending FORWARD-TSN with
+      // invalid SSNs, allowing ordered messages to stay in the queue that
+      // should've been discarded.
+      RTC_DLOG(LS_VERBOSE)
+          << log_prefix_
+          << "Message is built from fragments already seen - skipping";
+      return;
+    } else if (tsn == last_assembled_tsn_watermark_.next_value()) {
+      // Update watermark, or insert into delivered_tsns_
+      last_assembled_tsn_watermark_.Increment();
+    } else {
+      delivered_tsns_.insert(tsn);
+    }
+  }
+
+  // With new TSNs in delivered_tsns, gaps might be filled.
+  MaybeMoveLastAssembledWatermarkFurther();
+
+  reassembled_messages_.emplace_back(std::move(message));
+}
+
+void ReassemblyQueue::MaybeMoveLastAssembledWatermarkFurther() {
+  // `delivered_tsns_` contain TSNS when there is a gap between ranges of
+  // assembled TSNs. `last_assembled_tsn_watermark_` should not be adjacent to
+  // that list, because if so, it can be moved.
+  while (!delivered_tsns_.empty() &&
+         *delivered_tsns_.begin() ==
+             last_assembled_tsn_watermark_.next_value()) {
+    last_assembled_tsn_watermark_.Increment();
+    delivered_tsns_.erase(delivered_tsns_.begin());
+  }
+}
+
+void ReassemblyQueue::Handle(const AnyForwardTsnChunk& forward_tsn) {
+  RTC_DCHECK(IsConsistent());
+  UnwrappedTSN tsn = tsn_unwrapper_.Unwrap(forward_tsn.new_cumulative_tsn());
+
+  last_assembled_tsn_watermark_ = std::max(last_assembled_tsn_watermark_, tsn);
+  delivered_tsns_.erase(delivered_tsns_.begin(),
+                        delivered_tsns_.upper_bound(tsn));
+
+  MaybeMoveLastAssembledWatermarkFurther();
+
+  queued_bytes_ -=
+      streams_->HandleForwardTsn(tsn, forward_tsn.skipped_streams());
+  RTC_DCHECK(IsConsistent());
+}
+
+bool ReassemblyQueue::IsConsistent() const {
+  // `delivered_tsns_` and `last_assembled_tsn_watermark_` mustn't overlap or be
+  // adjacent.
+  if (!delivered_tsns_.empty() &&
+      last_assembled_tsn_watermark_.next_value() >= *delivered_tsns_.begin()) {
+    return false;
+  }
+
+  // Allow queued_bytes_ to be larger than max_size_bytes, as it's not actively
+  // enforced in this class. This comparison will still trigger if queued_bytes_
+  // became "negative".
+  return (queued_bytes_ >= 0 && queued_bytes_ <= 2 * max_size_bytes_);
+}
+
+HandoverReadinessStatus ReassemblyQueue::GetHandoverReadiness() const {
+  HandoverReadinessStatus status = streams_->GetHandoverReadiness();
+  if (!delivered_tsns_.empty()) {
+    status.Add(HandoverUnreadinessReason::kReassemblyQueueDeliveredTSNsGap);
+  }
+  if (deferred_reset_streams_.has_value()) {
+    status.Add(HandoverUnreadinessReason::kStreamResetDeferred);
+  }
+  return status;
+}
+
+void ReassemblyQueue::AddHandoverState(DcSctpSocketHandoverState& state) {
+  state.rx.last_assembled_tsn = last_assembled_tsn_watermark_.Wrap().value();
+  state.rx.last_completed_deferred_reset_req_sn =
+      last_completed_reset_req_seq_nbr_.value();
+  streams_->AddHandoverState(state);
+}
+
+void ReassemblyQueue::RestoreFromState(const DcSctpSocketHandoverState& state) {
+  // Validate that the component is in pristine state.
+  RTC_DCHECK(last_completed_reset_req_seq_nbr_ == ReconfigRequestSN(0));
+
+  last_assembled_tsn_watermark_ =
+      tsn_unwrapper_.Unwrap(TSN(state.rx.last_assembled_tsn));
+  last_completed_reset_req_seq_nbr_ =
+      ReconfigRequestSN(state.rx.last_completed_deferred_reset_req_sn);
+  streams_->RestoreFromState(state);
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/rx/reassembly_queue.h b/third_party/libwebrtc/net/dcsctp/rx/reassembly_queue.h
new file mode 100644
index 0000000000..91f30a3f69
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/reassembly_queue.h
@@ -0,0 +1,171 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_RX_REASSEMBLY_QUEUE_H_
+#define NET_DCSCTP_RX_REASSEMBLY_QUEUE_H_
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/reconfiguration_response_parameter.h"
+#include "net/dcsctp/public/dcsctp_handover_state.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/rx/reassembly_streams.h"
+
+namespace dcsctp {
+
+// Contains the received DATA chunks that haven't yet been reassembled, and
+// reassembles chunks when possible.
+//
+// The actual assembly is handled by an implementation of the
+// `ReassemblyStreams` interface.
+//
+// Except for reassembling fragmented messages, this class will also handle two
+// less common operations; To handle the receiver-side of partial reliability
+// (limited number of retransmissions or limited message lifetime) as well as
+// stream resetting, which is used when a sender wishes to close a data channel.
+//
+// Partial reliability is handled when a FORWARD-TSN or I-FORWARD-TSN chunk is
+// received, and it will simply delete any chunks matching the parameters in
+// that chunk. This is mainly implemented in ReassemblyStreams.
+//
+// Resetting streams is handled when a RECONFIG chunks is received, with an
+// "Outgoing SSN Reset Request" parameter. That parameter will contain a list of
+// streams to reset, and a `sender_last_assigned_tsn`. If this TSN is not yet
+// seen, the stream cannot be directly reset, and this class will respond that
+// the reset is "deferred". But if this TSN provided is known, the stream can be
+// immediately be reset.
+//
+// The ReassemblyQueue has a maximum size, as it would otherwise be an DoS
+// attack vector where a peer could consume all memory of the other peer by
+// sending a lot of ordered chunks, but carefully withholding an early one. It
+// also has a watermark limit, which the caller can query is the number of bytes
+// is above that limit. This is used by the caller to be selective in what to
+// add to the reassembly queue, so that it's not exhausted. The caller is
+// expected to call `is_full` prior to adding data to the queue and to act
+// accordingly if the queue is full.
+class ReassemblyQueue {
+ public:
+  // When the queue is filled over this fraction (of its maximum size), the
+  // socket should restrict incoming data to avoid filling up the queue.
+  static constexpr float kHighWatermarkLimit = 0.9;
+
+  ReassemblyQueue(absl::string_view log_prefix,
+                  TSN peer_initial_tsn,
+                  size_t max_size_bytes,
+                  bool use_message_interleaving = false);
+
+  // Adds a data chunk to the queue, with a `tsn` and other parameters in
+  // `data`.
+  void Add(TSN tsn, Data data);
+
+  // Indicates if the reassembly queue has any reassembled messages that can be
+  // retrieved by calling `FlushMessages`.
+  bool HasMessages() const { return !reassembled_messages_.empty(); }
+
+  // Returns any reassembled messages.
+  std::vector<DcSctpMessage> FlushMessages();
+
+  // Handle a ForwardTSN chunk, when the sender has indicated that the received
+  // (this class) should forget about some chunks. This is used to implement
+  // partial reliability.
+  void Handle(const AnyForwardTsnChunk& forward_tsn);
+
+  // Given the reset stream request and the current cum_tsn_ack, might either
+  // reset the streams directly (returns kSuccessPerformed), or at a later time,
+  // by entering the "deferred reset processing" mode (returns kInProgress).
+  ReconfigurationResponseParameter::Result ResetStreams(
+      const OutgoingSSNResetRequestParameter& req,
+      TSN cum_tsn_ack);
+
+  // Given the current (updated) cum_tsn_ack, might leave "defererred reset
+  // processing" mode and reset streams. Returns true if so.
+  bool MaybeResetStreamsDeferred(TSN cum_ack_tsn);
+
+  // The number of payload bytes that have been queued. Note that the actual
+  // memory usage is higher due to additional overhead of tracking received
+  // data.
+  size_t queued_bytes() const { return queued_bytes_; }
+
+  // The remaining bytes until the queue has reached the watermark limit.
+  size_t remaining_bytes() const { return watermark_bytes_ - queued_bytes_; }
+
+  // Indicates if the queue is full. Data should not be added to the queue when
+  // it's full.
+  bool is_full() const { return queued_bytes_ >= max_size_bytes_; }
+
+  // Indicates if the queue is above the watermark limit, which is a certain
+  // percentage of its size.
+  bool is_above_watermark() const { return queued_bytes_ >= watermark_bytes_; }
+
+  // Returns the watermark limit, in bytes.
+  size_t watermark_bytes() const { return watermark_bytes_; }
+
+  HandoverReadinessStatus GetHandoverReadiness() const;
+
+  void AddHandoverState(DcSctpSocketHandoverState& state);
+  void RestoreFromState(const DcSctpSocketHandoverState& state);
+
+ private:
+  bool IsConsistent() const;
+  void AddReassembledMessage(rtc::ArrayView<const UnwrappedTSN> tsns,
+                             DcSctpMessage message);
+  void MaybeMoveLastAssembledWatermarkFurther();
+
+  struct DeferredResetStreams {
+    explicit DeferredResetStreams(OutgoingSSNResetRequestParameter req)
+        : req(std::move(req)) {}
+    OutgoingSSNResetRequestParameter req;
+    std::vector<std::pair<TSN, Data>> deferred_chunks;
+  };
+
+  const std::string log_prefix_;
+  const size_t max_size_bytes_;
+  const size_t watermark_bytes_;
+  UnwrappedTSN::Unwrapper tsn_unwrapper_;
+
+  // Whenever a message has been assembled, either increase
+  // `last_assembled_tsn_watermark_` or - if there are gaps - add the message's
+  // TSNs into delivered_tsns_ so that messages are not re-delivered on
+  // duplicate chunks.
+  UnwrappedTSN last_assembled_tsn_watermark_;
+  std::set<UnwrappedTSN> delivered_tsns_;
+  // Messages that have been reassembled, and will be returned by
+  // `FlushMessages`.
+  std::vector<DcSctpMessage> reassembled_messages_;
+
+  // If present, "deferred reset processing" mode is active.
+  absl::optional<DeferredResetStreams> deferred_reset_streams_;
+
+  // Contains the last request sequence number of the
+  // OutgoingSSNResetRequestParameter that was performed.
+  ReconfigRequestSN last_completed_reset_req_seq_nbr_;
+
+  // The number of "payload bytes" that are in this queue, in total.
+  size_t queued_bytes_ = 0;
+
+  // The actual implementation of ReassemblyStreams.
+  std::unique_ptr<ReassemblyStreams> streams_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_RX_REASSEMBLY_QUEUE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/rx/reassembly_queue_test.cc b/third_party/libwebrtc/net/dcsctp/rx/reassembly_queue_test.cc
new file mode 100644
index 0000000000..549bc6fce1
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/reassembly_queue_test.cc
@@ -0,0 +1,509 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/rx/reassembly_queue.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <array>
+#include <cstdint>
+#include <iterator>
+#include <vector>
+
+#include "api/array_view.h"
+#include "net/dcsctp/common/handover_testing.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/testing/data_generator.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+using ::testing::SizeIs;
+using ::testing::UnorderedElementsAre;
+
+// The default maximum size of the Reassembly Queue.
+static constexpr size_t kBufferSize = 10000;
+
+static constexpr StreamID kStreamID(1);
+static constexpr SSN kSSN(0);
+static constexpr MID kMID(0);
+static constexpr FSN kFSN(0);
+static constexpr PPID kPPID(53);
+
+static constexpr std::array<uint8_t, 4> kShortPayload = {1, 2, 3, 4};
+static constexpr std::array<uint8_t, 4> kMessage2Payload = {5, 6, 7, 8};
+static constexpr std::array<uint8_t, 6> kSixBytePayload = {1, 2, 3, 4, 5, 6};
+static constexpr std::array<uint8_t, 8> kMediumPayload1 = {1, 2, 3, 4,
+                                                           5, 6, 7, 8};
+static constexpr std::array<uint8_t, 8> kMediumPayload2 = {9,  10, 11, 12,
+                                                           13, 14, 15, 16};
+static constexpr std::array<uint8_t, 16> kLongPayload = {
+    1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16};
+
+MATCHER_P3(SctpMessageIs, stream_id, ppid, expected_payload, "") {
+  if (arg.stream_id() != stream_id) {
+    *result_listener << "the stream_id is " << *arg.stream_id();
+    return false;
+  }
+
+  if (arg.ppid() != ppid) {
+    *result_listener << "the ppid is " << *arg.ppid();
+    return false;
+  }
+
+  if (std::vector<uint8_t>(arg.payload().begin(), arg.payload().end()) !=
+      std::vector<uint8_t>(expected_payload.begin(), expected_payload.end())) {
+    *result_listener << "the payload is wrong";
+    return false;
+  }
+  return true;
+}
+
+class ReassemblyQueueTest : public testing::Test {
+ protected:
+  ReassemblyQueueTest() {}
+  DataGenerator gen_;
+};
+
+TEST_F(ReassemblyQueueTest, EmptyQueue) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  EXPECT_FALSE(reasm.HasMessages());
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+}
+
+TEST_F(ReassemblyQueueTest, SingleUnorderedChunkMessage) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Add(TSN(10), gen_.Unordered({1, 2, 3, 4}, "BE"));
+  EXPECT_TRUE(reasm.HasMessages());
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(kStreamID, kPPID, kShortPayload)));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+}
+
+TEST_F(ReassemblyQueueTest, LargeUnorderedChunkAllPermutations) {
+  std::vector<uint32_t> tsns = {10, 11, 12, 13};
+  rtc::ArrayView<const uint8_t> payload(kLongPayload);
+  do {
+    ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+
+    for (size_t i = 0; i < tsns.size(); i++) {
+      auto span = payload.subview((tsns[i] - 10) * 4, 4);
+      Data::IsBeginning is_beginning(tsns[i] == 10);
+      Data::IsEnd is_end(tsns[i] == 13);
+
+      reasm.Add(TSN(tsns[i]),
+                Data(kStreamID, kSSN, kMID, kFSN, kPPID,
+                     std::vector<uint8_t>(span.begin(), span.end()),
+                     is_beginning, is_end, IsUnordered(false)));
+      if (i < 3) {
+        EXPECT_FALSE(reasm.HasMessages());
+      } else {
+        EXPECT_TRUE(reasm.HasMessages());
+        EXPECT_THAT(reasm.FlushMessages(),
+                    ElementsAre(SctpMessageIs(kStreamID, kPPID, kLongPayload)));
+        EXPECT_EQ(reasm.queued_bytes(), 0u);
+      }
+    }
+  } while (std::next_permutation(std::begin(tsns), std::end(tsns)));
+}
+
+TEST_F(ReassemblyQueueTest, SingleOrderedChunkMessage) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Add(TSN(10), gen_.Ordered({1, 2, 3, 4}, "BE"));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+  EXPECT_TRUE(reasm.HasMessages());
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(kStreamID, kPPID, kShortPayload)));
+}
+
+TEST_F(ReassemblyQueueTest, ManySmallOrderedMessages) {
+  std::vector<uint32_t> tsns = {10, 11, 12, 13};
+  rtc::ArrayView<const uint8_t> payload(kLongPayload);
+  do {
+    ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+    for (size_t i = 0; i < tsns.size(); i++) {
+      auto span = payload.subview((tsns[i] - 10) * 4, 4);
+      Data::IsBeginning is_beginning(true);
+      Data::IsEnd is_end(true);
+
+      SSN ssn(static_cast<uint16_t>(tsns[i] - 10));
+      reasm.Add(TSN(tsns[i]),
+                Data(kStreamID, ssn, kMID, kFSN, kPPID,
+                     std::vector<uint8_t>(span.begin(), span.end()),
+                     is_beginning, is_end, IsUnordered(false)));
+    }
+    EXPECT_THAT(
+        reasm.FlushMessages(),
+        ElementsAre(SctpMessageIs(kStreamID, kPPID, payload.subview(0, 4)),
+                    SctpMessageIs(kStreamID, kPPID, payload.subview(4, 4)),
+                    SctpMessageIs(kStreamID, kPPID, payload.subview(8, 4)),
+                    SctpMessageIs(kStreamID, kPPID, payload.subview(12, 4))));
+    EXPECT_EQ(reasm.queued_bytes(), 0u);
+  } while (std::next_permutation(std::begin(tsns), std::end(tsns)));
+}
+
+TEST_F(ReassemblyQueueTest, RetransmissionInLargeOrdered) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Add(TSN(10), gen_.Ordered({1}, "B"));
+  reasm.Add(TSN(12), gen_.Ordered({3}));
+  reasm.Add(TSN(13), gen_.Ordered({4}));
+  reasm.Add(TSN(14), gen_.Ordered({5}));
+  reasm.Add(TSN(15), gen_.Ordered({6}));
+  reasm.Add(TSN(16), gen_.Ordered({7}));
+  reasm.Add(TSN(17), gen_.Ordered({8}));
+  EXPECT_EQ(reasm.queued_bytes(), 7u);
+
+  // lost and retransmitted
+  reasm.Add(TSN(11), gen_.Ordered({2}));
+  reasm.Add(TSN(18), gen_.Ordered({9}));
+  reasm.Add(TSN(19), gen_.Ordered({10}));
+  EXPECT_EQ(reasm.queued_bytes(), 10u);
+  EXPECT_FALSE(reasm.HasMessages());
+
+  reasm.Add(TSN(20), gen_.Ordered({11, 12, 13, 14, 15, 16}, "E"));
+  EXPECT_TRUE(reasm.HasMessages());
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(kStreamID, kPPID, kLongPayload)));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+}
+
+TEST_F(ReassemblyQueueTest, ForwardTSNRemoveUnordered) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Add(TSN(10), gen_.Unordered({1}, "B"));
+  reasm.Add(TSN(12), gen_.Unordered({3}));
+  reasm.Add(TSN(13), gen_.Unordered({4}, "E"));
+
+  reasm.Add(TSN(14), gen_.Unordered({5}, "B"));
+  reasm.Add(TSN(15), gen_.Unordered({6}));
+  reasm.Add(TSN(17), gen_.Unordered({8}, "E"));
+  EXPECT_EQ(reasm.queued_bytes(), 6u);
+
+  EXPECT_FALSE(reasm.HasMessages());
+
+  reasm.Handle(ForwardTsnChunk(TSN(13), {}));
+  EXPECT_EQ(reasm.queued_bytes(), 3u);
+
+  // The lost chunk comes, but too late.
+  reasm.Add(TSN(11), gen_.Unordered({2}));
+  EXPECT_FALSE(reasm.HasMessages());
+  EXPECT_EQ(reasm.queued_bytes(), 3u);
+
+  // The second lost chunk comes, message is assembled.
+  reasm.Add(TSN(16), gen_.Unordered({7}));
+  EXPECT_TRUE(reasm.HasMessages());
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+}
+
+TEST_F(ReassemblyQueueTest, ForwardTSNRemoveOrdered) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Add(TSN(10), gen_.Ordered({1}, "B"));
+  reasm.Add(TSN(12), gen_.Ordered({3}));
+  reasm.Add(TSN(13), gen_.Ordered({4}, "E"));
+
+  reasm.Add(TSN(14), gen_.Ordered({5}, "B"));
+  reasm.Add(TSN(15), gen_.Ordered({6}));
+  reasm.Add(TSN(16), gen_.Ordered({7}));
+  reasm.Add(TSN(17), gen_.Ordered({8}, "E"));
+  EXPECT_EQ(reasm.queued_bytes(), 7u);
+
+  EXPECT_FALSE(reasm.HasMessages());
+
+  reasm.Handle(ForwardTsnChunk(
+      TSN(13), {ForwardTsnChunk::SkippedStream(kStreamID, kSSN)}));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+
+  // The lost chunk comes, but too late.
+  EXPECT_TRUE(reasm.HasMessages());
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(kStreamID, kPPID, kMessage2Payload)));
+}
+
+TEST_F(ReassemblyQueueTest, ForwardTSNRemoveALotOrdered) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Add(TSN(10), gen_.Ordered({1}, "B"));
+  reasm.Add(TSN(12), gen_.Ordered({3}));
+  reasm.Add(TSN(13), gen_.Ordered({4}, "E"));
+
+  reasm.Add(TSN(15), gen_.Ordered({5}, "B"));
+  reasm.Add(TSN(16), gen_.Ordered({6}));
+  reasm.Add(TSN(17), gen_.Ordered({7}));
+  reasm.Add(TSN(18), gen_.Ordered({8}, "E"));
+  EXPECT_EQ(reasm.queued_bytes(), 7u);
+
+  EXPECT_FALSE(reasm.HasMessages());
+
+  reasm.Handle(ForwardTsnChunk(
+      TSN(13), {ForwardTsnChunk::SkippedStream(kStreamID, kSSN)}));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+
+  // The lost chunk comes, but too late.
+  EXPECT_TRUE(reasm.HasMessages());
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(kStreamID, kPPID, kMessage2Payload)));
+}
+
+TEST_F(ReassemblyQueueTest, ShouldntDeliverMessagesBeforeInitialTsn) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Add(TSN(5), gen_.Unordered({1, 2, 3, 4}, "BE"));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+  EXPECT_FALSE(reasm.HasMessages());
+}
+
+TEST_F(ReassemblyQueueTest, ShouldntRedeliverUnorderedMessages) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Add(TSN(10), gen_.Unordered({1, 2, 3, 4}, "BE"));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+  EXPECT_TRUE(reasm.HasMessages());
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(kStreamID, kPPID, kShortPayload)));
+  reasm.Add(TSN(10), gen_.Unordered({1, 2, 3, 4}, "BE"));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+  EXPECT_FALSE(reasm.HasMessages());
+}
+
+TEST_F(ReassemblyQueueTest, ShouldntRedeliverUnorderedMessagesReallyUnordered) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Add(TSN(10), gen_.Unordered({1, 2, 3, 4}, "B"));
+  EXPECT_EQ(reasm.queued_bytes(), 4u);
+
+  EXPECT_FALSE(reasm.HasMessages());
+
+  reasm.Add(TSN(12), gen_.Unordered({1, 2, 3, 4}, "BE"));
+  EXPECT_EQ(reasm.queued_bytes(), 4u);
+  EXPECT_TRUE(reasm.HasMessages());
+
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(kStreamID, kPPID, kShortPayload)));
+  reasm.Add(TSN(12), gen_.Unordered({1, 2, 3, 4}, "BE"));
+  EXPECT_EQ(reasm.queued_bytes(), 4u);
+  EXPECT_FALSE(reasm.HasMessages());
+}
+
+TEST_F(ReassemblyQueueTest, ShouldntDeliverBeforeForwardedTsn) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Handle(ForwardTsnChunk(TSN(12), {}));
+
+  reasm.Add(TSN(12), gen_.Unordered({1, 2, 3, 4}, "BE"));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+  EXPECT_FALSE(reasm.HasMessages());
+}
+
+TEST_F(ReassemblyQueueTest, NotReadyForHandoverWhenDeliveredTsnsHaveGap) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  reasm.Add(TSN(10), gen_.Unordered({1, 2, 3, 4}, "B"));
+  EXPECT_FALSE(reasm.HasMessages());
+
+  reasm.Add(TSN(12), gen_.Unordered({1, 2, 3, 4}, "BE"));
+  EXPECT_TRUE(reasm.HasMessages());
+  EXPECT_EQ(
+      reasm.GetHandoverReadiness(),
+      HandoverReadinessStatus()
+          .Add(HandoverUnreadinessReason::kReassemblyQueueDeliveredTSNsGap)
+          .Add(
+              HandoverUnreadinessReason::kUnorderedStreamHasUnassembledChunks));
+
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(kStreamID, kPPID, kShortPayload)));
+  EXPECT_EQ(
+      reasm.GetHandoverReadiness(),
+      HandoverReadinessStatus()
+          .Add(HandoverUnreadinessReason::kReassemblyQueueDeliveredTSNsGap)
+          .Add(
+              HandoverUnreadinessReason::kUnorderedStreamHasUnassembledChunks));
+
+  reasm.Handle(ForwardTsnChunk(TSN(13), {}));
+  EXPECT_EQ(reasm.GetHandoverReadiness(), HandoverReadinessStatus());
+}
+
+TEST_F(ReassemblyQueueTest, NotReadyForHandoverWhenResetStreamIsDeferred) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  DataGeneratorOptions opts;
+  opts.message_id = MID(0);
+  reasm.Add(TSN(10), gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+  opts.message_id = MID(1);
+  reasm.Add(TSN(11), gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+  EXPECT_THAT(reasm.FlushMessages(), SizeIs(2));
+
+  reasm.ResetStreams(
+      OutgoingSSNResetRequestParameter(
+          ReconfigRequestSN(10), ReconfigRequestSN(3), TSN(13), {StreamID(1)}),
+      TSN(11));
+  EXPECT_EQ(reasm.GetHandoverReadiness(),
+            HandoverReadinessStatus().Add(
+                HandoverUnreadinessReason::kStreamResetDeferred));
+
+  opts.message_id = MID(3);
+  opts.ppid = PPID(3);
+  reasm.Add(TSN(13), gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+  reasm.MaybeResetStreamsDeferred(TSN(11));
+
+  opts.message_id = MID(2);
+  opts.ppid = PPID(2);
+  reasm.Add(TSN(13), gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+  reasm.MaybeResetStreamsDeferred(TSN(15));
+  EXPECT_EQ(reasm.GetHandoverReadiness(),
+            HandoverReadinessStatus().Add(
+                HandoverUnreadinessReason::kReassemblyQueueDeliveredTSNsGap));
+
+  EXPECT_THAT(reasm.FlushMessages(), SizeIs(2));
+  EXPECT_EQ(reasm.GetHandoverReadiness(),
+            HandoverReadinessStatus().Add(
+                HandoverUnreadinessReason::kReassemblyQueueDeliveredTSNsGap));
+
+  reasm.Handle(ForwardTsnChunk(TSN(15), {}));
+  EXPECT_EQ(reasm.GetHandoverReadiness(), HandoverReadinessStatus());
+}
+
+TEST_F(ReassemblyQueueTest, HandoverInInitialState) {
+  ReassemblyQueue reasm1("log: ", TSN(10), kBufferSize);
+
+  EXPECT_EQ(reasm1.GetHandoverReadiness(), HandoverReadinessStatus());
+  DcSctpSocketHandoverState state;
+  reasm1.AddHandoverState(state);
+  g_handover_state_transformer_for_test(&state);
+  ReassemblyQueue reasm2("log: ", TSN(100), kBufferSize,
+                         /*use_message_interleaving=*/false);
+  reasm2.RestoreFromState(state);
+
+  reasm2.Add(TSN(10), gen_.Ordered({1, 2, 3, 4}, "BE"));
+  EXPECT_THAT(reasm2.FlushMessages(), SizeIs(1));
+}
+
+TEST_F(ReassemblyQueueTest, HandoverAfterHavingAssembedOneMessage) {
+  ReassemblyQueue reasm1("log: ", TSN(10), kBufferSize);
+  reasm1.Add(TSN(10), gen_.Ordered({1, 2, 3, 4}, "BE"));
+  EXPECT_THAT(reasm1.FlushMessages(), SizeIs(1));
+
+  EXPECT_EQ(reasm1.GetHandoverReadiness(), HandoverReadinessStatus());
+  DcSctpSocketHandoverState state;
+  reasm1.AddHandoverState(state);
+  g_handover_state_transformer_for_test(&state);
+  ReassemblyQueue reasm2("log: ", TSN(100), kBufferSize,
+                         /*use_message_interleaving=*/false);
+  reasm2.RestoreFromState(state);
+
+  reasm2.Add(TSN(11), gen_.Ordered({1, 2, 3, 4}, "BE"));
+  EXPECT_THAT(reasm2.FlushMessages(), SizeIs(1));
+}
+
+TEST_F(ReassemblyQueueTest, HandleInconsistentForwardTSN) {
+  // Found when fuzzing.
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize);
+  // Add TSN=43, SSN=7. Can't be reassembled as previous SSNs aren't known.
+  reasm.Add(TSN(43), Data(kStreamID, SSN(7), MID(0), FSN(0), kPPID,
+                          std::vector<uint8_t>(10), Data::IsBeginning(true),
+                          Data::IsEnd(true), IsUnordered(false)));
+
+  // Invalid, as TSN=44 have to have SSN>=7, but peer says 6.
+  reasm.Handle(ForwardTsnChunk(
+      TSN(44), {ForwardTsnChunk::SkippedStream(kStreamID, SSN(6))}));
+
+  // Don't assemble SSN=7, as that TSN is skipped.
+  EXPECT_FALSE(reasm.HasMessages());
+}
+
+TEST_F(ReassemblyQueueTest, SingleUnorderedChunkMessageInRfc8260) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize,
+                        /*use_message_interleaving=*/true);
+  reasm.Add(TSN(10), Data(StreamID(1), SSN(0), MID(0), FSN(0), kPPID,
+                          {1, 2, 3, 4}, Data::IsBeginning(true),
+                          Data::IsEnd(true), IsUnordered(true)));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+  EXPECT_TRUE(reasm.HasMessages());
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(kStreamID, kPPID, kShortPayload)));
+}
+
+TEST_F(ReassemblyQueueTest, TwoInterleavedChunks) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize,
+                        /*use_message_interleaving=*/true);
+  reasm.Add(TSN(10), Data(StreamID(1), SSN(0), MID(0), FSN(0), kPPID,
+                          {1, 2, 3, 4}, Data::IsBeginning(true),
+                          Data::IsEnd(false), IsUnordered(true)));
+  reasm.Add(TSN(11), Data(StreamID(2), SSN(0), MID(0), FSN(0), kPPID,
+                          {9, 10, 11, 12}, Data::IsBeginning(true),
+                          Data::IsEnd(false), IsUnordered(true)));
+  EXPECT_EQ(reasm.queued_bytes(), 8u);
+  reasm.Add(TSN(12), Data(StreamID(1), SSN(0), MID(0), FSN(1), kPPID,
+                          {5, 6, 7, 8}, Data::IsBeginning(false),
+                          Data::IsEnd(true), IsUnordered(true)));
+  EXPECT_EQ(reasm.queued_bytes(), 4u);
+  reasm.Add(TSN(13), Data(StreamID(2), SSN(0), MID(0), FSN(1), kPPID,
+                          {13, 14, 15, 16}, Data::IsBeginning(false),
+                          Data::IsEnd(true), IsUnordered(true)));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+  EXPECT_TRUE(reasm.HasMessages());
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(StreamID(1), kPPID, kMediumPayload1),
+                          SctpMessageIs(StreamID(2), kPPID, kMediumPayload2)));
+}
+
+TEST_F(ReassemblyQueueTest, UnorderedInterleavedMessagesAllPermutations) {
+  std::vector<int> indexes = {0, 1, 2, 3, 4, 5};
+  TSN tsns[] = {TSN(10), TSN(11), TSN(12), TSN(13), TSN(14), TSN(15)};
+  StreamID stream_ids[] = {StreamID(1), StreamID(2), StreamID(1),
+                           StreamID(1), StreamID(2), StreamID(2)};
+  FSN fsns[] = {FSN(0), FSN(0), FSN(1), FSN(2), FSN(1), FSN(2)};
+  rtc::ArrayView<const uint8_t> payload(kSixBytePayload);
+  do {
+    ReassemblyQueue reasm("log: ", TSN(10), kBufferSize,
+                          /*use_message_interleaving=*/true);
+    for (int i : indexes) {
+      auto span = payload.subview(*fsns[i] * 2, 2);
+      Data::IsBeginning is_beginning(fsns[i] == FSN(0));
+      Data::IsEnd is_end(fsns[i] == FSN(2));
+      reasm.Add(tsns[i], Data(stream_ids[i], SSN(0), MID(0), fsns[i], kPPID,
+                              std::vector<uint8_t>(span.begin(), span.end()),
+                              is_beginning, is_end, IsUnordered(true)));
+    }
+    EXPECT_TRUE(reasm.HasMessages());
+    EXPECT_THAT(reasm.FlushMessages(),
+                UnorderedElementsAre(
+                    SctpMessageIs(StreamID(1), kPPID, kSixBytePayload),
+                    SctpMessageIs(StreamID(2), kPPID, kSixBytePayload)));
+    EXPECT_EQ(reasm.queued_bytes(), 0u);
+  } while (std::next_permutation(std::begin(indexes), std::end(indexes)));
+}
+
+TEST_F(ReassemblyQueueTest, IForwardTSNRemoveALotOrdered) {
+  ReassemblyQueue reasm("log: ", TSN(10), kBufferSize,
+                        /*use_message_interleaving=*/true);
+  reasm.Add(TSN(10), gen_.Ordered({1}, "B"));
+  gen_.Ordered({2}, "");
+  reasm.Add(TSN(12), gen_.Ordered({3}, ""));
+  reasm.Add(TSN(13), gen_.Ordered({4}, "E"));
+  reasm.Add(TSN(15), gen_.Ordered({5}, "B"));
+  reasm.Add(TSN(16), gen_.Ordered({6}, ""));
+  reasm.Add(TSN(17), gen_.Ordered({7}, ""));
+  reasm.Add(TSN(18), gen_.Ordered({8}, "E"));
+
+  ASSERT_FALSE(reasm.HasMessages());
+  EXPECT_EQ(reasm.queued_bytes(), 7u);
+
+  reasm.Handle(
+      IForwardTsnChunk(TSN(13), {IForwardTsnChunk::SkippedStream(
+                                    IsUnordered(false), kStreamID, MID(0))}));
+  EXPECT_EQ(reasm.queued_bytes(), 0u);
+
+  // The lost chunk comes, but too late.
+  ASSERT_TRUE(reasm.HasMessages());
+  EXPECT_THAT(reasm.FlushMessages(),
+              ElementsAre(SctpMessageIs(kStreamID, kPPID, kMessage2Payload)));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/rx/reassembly_streams.cc b/third_party/libwebrtc/net/dcsctp/rx/reassembly_streams.cc
new file mode 100644
index 0000000000..9fd52fb15d
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/reassembly_streams.cc
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/rx/reassembly_streams.h"
+
+#include <cstddef>
+#include <map>
+#include <utility>
+
+namespace dcsctp {
+
+ReassembledMessage AssembleMessage(std::map<UnwrappedTSN, Data>::iterator start,
+                                   std::map<UnwrappedTSN, Data>::iterator end) {
+  size_t count = std::distance(start, end);
+
+  if (count == 1) {
+    // Fast path - zero-copy
+    Data& data = start->second;
+
+    return ReassembledMessage{
+        .tsns = {start->first},
+        .message = DcSctpMessage(data.stream_id, data.ppid,
+                                 std::move(start->second.payload)),
+    };
+  }
+
+  // Slow path - will need to concatenate the payload.
+  std::vector<UnwrappedTSN> tsns;
+  std::vector<uint8_t> payload;
+
+  size_t payload_size = std::accumulate(
+      start, end, 0,
+      [](size_t v, const auto& p) { return v + p.second.size(); });
+
+  tsns.reserve(count);
+  payload.reserve(payload_size);
+  for (auto it = start; it != end; ++it) {
+    Data& data = it->second;
+    tsns.push_back(it->first);
+    payload.insert(payload.end(), data.payload.begin(), data.payload.end());
+  }
+
+  return ReassembledMessage{
+      .tsns = std::move(tsns),
+      .message = DcSctpMessage(start->second.stream_id, start->second.ppid,
+                               std::move(payload)),
+  };
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/rx/reassembly_streams.h b/third_party/libwebrtc/net/dcsctp/rx/reassembly_streams.h
new file mode 100644
index 0000000000..0ecfac0c0a
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/reassembly_streams.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_RX_REASSEMBLY_STREAMS_H_
+#define NET_DCSCTP_RX_REASSEMBLY_STREAMS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <functional>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/dcsctp_handover_state.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+
+namespace dcsctp {
+
+// Implementations of this interface will be called when data is received, when
+// data should be skipped/forgotten or when sequence number should be reset.
+//
+// As a result of these operations - mainly when data is received - the
+// implementations of this interface should notify when a message has been
+// assembled, by calling the provided callback of type `OnAssembledMessage`. How
+// it assembles messages will depend on e.g. if a message was sent on an ordered
+// or unordered stream.
+//
+// Implementations will - for each operation - indicate how much additional
+// memory that has been used as a result of performing the operation. This is
+// used to limit the maximum amount of memory used, to prevent out-of-memory
+// situations.
+class ReassemblyStreams {
+ public:
+  // This callback will be provided as an argument to the constructor of the
+  // concrete class implementing this interface and should be called when a
+  // message has been assembled as well as indicating from which TSNs this
+  // message was assembled from.
+  using OnAssembledMessage =
+      std::function<void(rtc::ArrayView<const UnwrappedTSN> tsns,
+                         DcSctpMessage message)>;
+
+  virtual ~ReassemblyStreams() = default;
+
+  // Adds a data chunk to a stream as identified in `data`.
+  // If it was the last remaining chunk in a message, reassemble one (or
+  // several, in case of ordered chunks) messages.
+  //
+  // Returns the additional number of bytes added to the queue as a result of
+  // performing this operation. If this addition resulted in messages being
+  // assembled and delivered, this may be negative.
+  virtual int Add(UnwrappedTSN tsn, Data data) = 0;
+
+  // Called for incoming FORWARD-TSN/I-FORWARD-TSN chunks - when the sender
+  // wishes the received to skip/forget about data up until the provided TSN.
+  // This is used to implement partial reliability, such as limiting the number
+  // of retransmissions or the an expiration duration. As a result of skipping
+  // data, this may result in the implementation being able to assemble messages
+  // in ordered streams.
+  //
+  // Returns the number of bytes removed from the queue as a result of
+  // this operation.
+  virtual size_t HandleForwardTsn(
+      UnwrappedTSN new_cumulative_ack_tsn,
+      rtc::ArrayView<const AnyForwardTsnChunk::SkippedStream>
+          skipped_streams) = 0;
+
+  // Called for incoming (possibly deferred) RE_CONFIG chunks asking for
+  // either a few streams, or all streams (when the list is empty) to be
+  // reset - to have their next SSN or Message ID to be zero.
+  virtual void ResetStreams(rtc::ArrayView<const StreamID> stream_ids) = 0;
+
+  virtual HandoverReadinessStatus GetHandoverReadiness() const = 0;
+  virtual void AddHandoverState(DcSctpSocketHandoverState& state) = 0;
+  virtual void RestoreFromState(const DcSctpSocketHandoverState& state) = 0;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_RX_REASSEMBLY_STREAMS_H_
diff --git a/third_party/libwebrtc/net/dcsctp/rx/traditional_reassembly_streams.cc b/third_party/libwebrtc/net/dcsctp/rx/traditional_reassembly_streams.cc
new file mode 100644
index 0000000000..dce6c90131
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/traditional_reassembly_streams.cc
@@ -0,0 +1,348 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/rx/traditional_reassembly_streams.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <functional>
+#include <iterator>
+#include <map>
+#include <numeric>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+namespace {
+
+// Given a map (`chunks`) and an iterator to within that map (`iter`), this
+// function will return an iterator to the first chunk in that message, which
+// has the `is_beginning` flag set. If there are any gaps, or if the beginning
+// can't be found, `absl::nullopt` is returned.
+absl::optional<std::map<UnwrappedTSN, Data>::iterator> FindBeginning(
+    const std::map<UnwrappedTSN, Data>& chunks,
+    std::map<UnwrappedTSN, Data>::iterator iter) {
+  UnwrappedTSN prev_tsn = iter->first;
+  for (;;) {
+    if (iter->second.is_beginning) {
+      return iter;
+    }
+    if (iter == chunks.begin()) {
+      return absl::nullopt;
+    }
+    --iter;
+    if (iter->first.next_value() != prev_tsn) {
+      return absl::nullopt;
+    }
+    prev_tsn = iter->first;
+  }
+}
+
+// Given a map (`chunks`) and an iterator to within that map (`iter`), this
+// function will return an iterator to the chunk after the last chunk in that
+// message, which has the `is_end` flag set. If there are any gaps, or if the
+// end can't be found, `absl::nullopt` is returned.
+absl::optional<std::map<UnwrappedTSN, Data>::iterator> FindEnd(
+    std::map<UnwrappedTSN, Data>& chunks,
+    std::map<UnwrappedTSN, Data>::iterator iter) {
+  UnwrappedTSN prev_tsn = iter->first;
+  for (;;) {
+    if (iter->second.is_end) {
+      return ++iter;
+    }
+    ++iter;
+    if (iter == chunks.end()) {
+      return absl::nullopt;
+    }
+    if (iter->first != prev_tsn.next_value()) {
+      return absl::nullopt;
+    }
+    prev_tsn = iter->first;
+  }
+}
+}  // namespace
+
+TraditionalReassemblyStreams::TraditionalReassemblyStreams(
+    absl::string_view log_prefix,
+    OnAssembledMessage on_assembled_message)
+    : log_prefix_(log_prefix),
+      on_assembled_message_(std::move(on_assembled_message)) {}
+
+int TraditionalReassemblyStreams::UnorderedStream::Add(UnwrappedTSN tsn,
+                                                       Data data) {
+  int queued_bytes = data.size();
+  auto [it, inserted] = chunks_.emplace(tsn, std::move(data));
+  if (!inserted) {
+    return 0;
+  }
+
+  queued_bytes -= TryToAssembleMessage(it);
+
+  return queued_bytes;
+}
+
+size_t TraditionalReassemblyStreams::UnorderedStream::TryToAssembleMessage(
+    ChunkMap::iterator iter) {
+  // TODO(boivie): This method is O(N) with the number of fragments in a
+  // message, which can be inefficient for very large values of N. This could be
+  // optimized by e.g. only trying to assemble a message once _any_ beginning
+  // and _any_ end has been found.
+  absl::optional<ChunkMap::iterator> start = FindBeginning(chunks_, iter);
+  if (!start.has_value()) {
+    return 0;
+  }
+  absl::optional<ChunkMap::iterator> end = FindEnd(chunks_, iter);
+  if (!end.has_value()) {
+    return 0;
+  }
+
+  size_t bytes_assembled = AssembleMessage(*start, *end);
+  chunks_.erase(*start, *end);
+  return bytes_assembled;
+}
+
+size_t TraditionalReassemblyStreams::StreamBase::AssembleMessage(
+    const ChunkMap::iterator start,
+    const ChunkMap::iterator end) {
+  size_t count = std::distance(start, end);
+
+  if (count == 1) {
+    // Fast path - zero-copy
+    const Data& data = start->second;
+    size_t payload_size = start->second.size();
+    UnwrappedTSN tsns[1] = {start->first};
+    DcSctpMessage message(data.stream_id, data.ppid, std::move(data.payload));
+    parent_.on_assembled_message_(tsns, std::move(message));
+    return payload_size;
+  }
+
+  // Slow path - will need to concatenate the payload.
+  std::vector<UnwrappedTSN> tsns;
+  std::vector<uint8_t> payload;
+
+  size_t payload_size = std::accumulate(
+      start, end, 0,
+      [](size_t v, const auto& p) { return v + p.second.size(); });
+
+  tsns.reserve(count);
+  payload.reserve(payload_size);
+  for (auto it = start; it != end; ++it) {
+    const Data& data = it->second;
+    tsns.push_back(it->first);
+    payload.insert(payload.end(), data.payload.begin(), data.payload.end());
+  }
+
+  DcSctpMessage message(start->second.stream_id, start->second.ppid,
+                        std::move(payload));
+  parent_.on_assembled_message_(tsns, std::move(message));
+
+  return payload_size;
+}
+
+size_t TraditionalReassemblyStreams::UnorderedStream::EraseTo(
+    UnwrappedTSN tsn) {
+  auto end_iter = chunks_.upper_bound(tsn);
+  size_t removed_bytes = std::accumulate(
+      chunks_.begin(), end_iter, 0,
+      [](size_t r, const auto& p) { return r + p.second.size(); });
+
+  chunks_.erase(chunks_.begin(), end_iter);
+  return removed_bytes;
+}
+
+size_t TraditionalReassemblyStreams::OrderedStream::TryToAssembleMessage() {
+  if (chunks_by_ssn_.empty() || chunks_by_ssn_.begin()->first != next_ssn_) {
+    return 0;
+  }
+
+  ChunkMap& chunks = chunks_by_ssn_.begin()->second;
+
+  if (!chunks.begin()->second.is_beginning || !chunks.rbegin()->second.is_end) {
+    return 0;
+  }
+
+  uint32_t tsn_diff =
+      UnwrappedTSN::Difference(chunks.rbegin()->first, chunks.begin()->first);
+  if (tsn_diff != chunks.size() - 1) {
+    return 0;
+  }
+
+  size_t assembled_bytes = AssembleMessage(chunks.begin(), chunks.end());
+  chunks_by_ssn_.erase(chunks_by_ssn_.begin());
+  next_ssn_.Increment();
+  return assembled_bytes;
+}
+
+size_t TraditionalReassemblyStreams::OrderedStream::TryToAssembleMessages() {
+  size_t assembled_bytes = 0;
+
+  for (;;) {
+    size_t assembled_bytes_this_iter = TryToAssembleMessage();
+    if (assembled_bytes_this_iter == 0) {
+      break;
+    }
+    assembled_bytes += assembled_bytes_this_iter;
+  }
+  return assembled_bytes;
+}
+
+int TraditionalReassemblyStreams::OrderedStream::Add(UnwrappedTSN tsn,
+                                                     Data data) {
+  int queued_bytes = data.size();
+
+  UnwrappedSSN ssn = ssn_unwrapper_.Unwrap(data.ssn);
+  auto [unused, inserted] = chunks_by_ssn_[ssn].emplace(tsn, std::move(data));
+  if (!inserted) {
+    return 0;
+  }
+
+  if (ssn == next_ssn_) {
+    queued_bytes -= TryToAssembleMessages();
+  }
+
+  return queued_bytes;
+}
+
+size_t TraditionalReassemblyStreams::OrderedStream::EraseTo(SSN ssn) {
+  UnwrappedSSN unwrapped_ssn = ssn_unwrapper_.Unwrap(ssn);
+
+  auto end_iter = chunks_by_ssn_.upper_bound(unwrapped_ssn);
+  size_t removed_bytes = std::accumulate(
+      chunks_by_ssn_.begin(), end_iter, 0, [](size_t r1, const auto& p) {
+        return r1 +
+               absl::c_accumulate(p.second, 0, [](size_t r2, const auto& q) {
+                 return r2 + q.second.size();
+               });
+      });
+  chunks_by_ssn_.erase(chunks_by_ssn_.begin(), end_iter);
+
+  if (unwrapped_ssn >= next_ssn_) {
+    unwrapped_ssn.Increment();
+    next_ssn_ = unwrapped_ssn;
+  }
+
+  removed_bytes += TryToAssembleMessages();
+  return removed_bytes;
+}
+
+int TraditionalReassemblyStreams::Add(UnwrappedTSN tsn, Data data) {
+  if (data.is_unordered) {
+    auto it = unordered_streams_.try_emplace(data.stream_id, this).first;
+    return it->second.Add(tsn, std::move(data));
+  }
+
+  auto it = ordered_streams_.try_emplace(data.stream_id, this).first;
+  return it->second.Add(tsn, std::move(data));
+}
+
+size_t TraditionalReassemblyStreams::HandleForwardTsn(
+    UnwrappedTSN new_cumulative_ack_tsn,
+    rtc::ArrayView<const AnyForwardTsnChunk::SkippedStream> skipped_streams) {
+  size_t bytes_removed = 0;
+  // The `skipped_streams` only cover ordered messages - need to
+  // iterate all unordered streams manually to remove those chunks.
+  for (auto& [unused, stream] : unordered_streams_) {
+    bytes_removed += stream.EraseTo(new_cumulative_ack_tsn);
+  }
+
+  for (const auto& skipped_stream : skipped_streams) {
+    auto it =
+        ordered_streams_.try_emplace(skipped_stream.stream_id, this).first;
+    bytes_removed += it->second.EraseTo(skipped_stream.ssn);
+  }
+
+  return bytes_removed;
+}
+
+void TraditionalReassemblyStreams::ResetStreams(
+    rtc::ArrayView<const StreamID> stream_ids) {
+  if (stream_ids.empty()) {
+    for (auto& [stream_id, stream] : ordered_streams_) {
+      RTC_DLOG(LS_VERBOSE) << log_prefix_
+                           << "Resetting implicit stream_id=" << *stream_id;
+      stream.Reset();
+    }
+  } else {
+    for (StreamID stream_id : stream_ids) {
+      auto it = ordered_streams_.find(stream_id);
+      if (it != ordered_streams_.end()) {
+        RTC_DLOG(LS_VERBOSE)
+            << log_prefix_ << "Resetting explicit stream_id=" << *stream_id;
+        it->second.Reset();
+      }
+    }
+  }
+}
+
+HandoverReadinessStatus TraditionalReassemblyStreams::GetHandoverReadiness()
+    const {
+  HandoverReadinessStatus status;
+  for (const auto& [unused, stream] : ordered_streams_) {
+    if (stream.has_unassembled_chunks()) {
+      status.Add(HandoverUnreadinessReason::kOrderedStreamHasUnassembledChunks);
+      break;
+    }
+  }
+  for (const auto& [unused, stream] : unordered_streams_) {
+    if (stream.has_unassembled_chunks()) {
+      status.Add(
+          HandoverUnreadinessReason::kUnorderedStreamHasUnassembledChunks);
+      break;
+    }
+  }
+  return status;
+}
+
+void TraditionalReassemblyStreams::AddHandoverState(
+    DcSctpSocketHandoverState& state) {
+  for (const auto& [stream_id, stream] : ordered_streams_) {
+    DcSctpSocketHandoverState::OrderedStream state_stream;
+    state_stream.id = stream_id.value();
+    state_stream.next_ssn = stream.next_ssn().value();
+    state.rx.ordered_streams.push_back(std::move(state_stream));
+  }
+  for (const auto& [stream_id, unused] : unordered_streams_) {
+    DcSctpSocketHandoverState::UnorderedStream state_stream;
+    state_stream.id = stream_id.value();
+    state.rx.unordered_streams.push_back(std::move(state_stream));
+  }
+}
+
+void TraditionalReassemblyStreams::RestoreFromState(
+    const DcSctpSocketHandoverState& state) {
+  // Validate that the component is in pristine state.
+  RTC_DCHECK(ordered_streams_.empty());
+  RTC_DCHECK(unordered_streams_.empty());
+
+  for (const DcSctpSocketHandoverState::OrderedStream& state_stream :
+       state.rx.ordered_streams) {
+    ordered_streams_.emplace(
+        std::piecewise_construct,
+        std::forward_as_tuple(StreamID(state_stream.id)),
+        std::forward_as_tuple(this, SSN(state_stream.next_ssn)));
+  }
+  for (const DcSctpSocketHandoverState::UnorderedStream& state_stream :
+       state.rx.unordered_streams) {
+    unordered_streams_.emplace(std::piecewise_construct,
+                               std::forward_as_tuple(StreamID(state_stream.id)),
+                               std::forward_as_tuple(this));
+  }
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/rx/traditional_reassembly_streams.h b/third_party/libwebrtc/net/dcsctp/rx/traditional_reassembly_streams.h
new file mode 100644
index 0000000000..4825afd1ba
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/traditional_reassembly_streams.h
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_RX_TRADITIONAL_REASSEMBLY_STREAMS_H_
+#define NET_DCSCTP_RX_TRADITIONAL_REASSEMBLY_STREAMS_H_
+#include <stddef.h>
+#include <stdint.h>
+
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/rx/reassembly_streams.h"
+
+namespace dcsctp {
+
+// Handles reassembly of incoming data when interleaved message sending
+// is not enabled on the association, i.e. when RFC8260 is not in use and
+// RFC4960 is to be followed.
+class TraditionalReassemblyStreams : public ReassemblyStreams {
+ public:
+  TraditionalReassemblyStreams(absl::string_view log_prefix,
+                               OnAssembledMessage on_assembled_message);
+
+  int Add(UnwrappedTSN tsn, Data data) override;
+
+  size_t HandleForwardTsn(
+      UnwrappedTSN new_cumulative_ack_tsn,
+      rtc::ArrayView<const AnyForwardTsnChunk::SkippedStream> skipped_streams)
+      override;
+
+  void ResetStreams(rtc::ArrayView<const StreamID> stream_ids) override;
+
+  HandoverReadinessStatus GetHandoverReadiness() const override;
+  void AddHandoverState(DcSctpSocketHandoverState& state) override;
+  void RestoreFromState(const DcSctpSocketHandoverState& state) override;
+
+ private:
+  using ChunkMap = std::map<UnwrappedTSN, Data>;
+
+  // Base class for `UnorderedStream` and `OrderedStream`.
+  class StreamBase {
+   protected:
+    explicit StreamBase(TraditionalReassemblyStreams* parent)
+        : parent_(*parent) {}
+
+    size_t AssembleMessage(ChunkMap::iterator start, ChunkMap::iterator end);
+    TraditionalReassemblyStreams& parent_;
+  };
+
+  // Manages all received data for a specific unordered stream, and assembles
+  // messages when possible.
+  class UnorderedStream : StreamBase {
+   public:
+    explicit UnorderedStream(TraditionalReassemblyStreams* parent)
+        : StreamBase(parent) {}
+    int Add(UnwrappedTSN tsn, Data data);
+    // Returns the number of bytes removed from the queue.
+    size_t EraseTo(UnwrappedTSN tsn);
+    bool has_unassembled_chunks() const { return !chunks_.empty(); }
+
+   private:
+    // Given an iterator to any chunk within the map, try to assemble a message
+    // into `reassembled_messages` containing it and - if successful - erase
+    // those chunks from the stream chunks map.
+    //
+    // Returns the number of bytes that were assembled.
+    size_t TryToAssembleMessage(ChunkMap::iterator iter);
+
+    ChunkMap chunks_;
+  };
+
+  // Manages all received data for a specific ordered stream, and assembles
+  // messages when possible.
+  class OrderedStream : StreamBase {
+   public:
+    explicit OrderedStream(TraditionalReassemblyStreams* parent,
+                           SSN next_ssn = SSN(0))
+        : StreamBase(parent), next_ssn_(ssn_unwrapper_.Unwrap(next_ssn)) {}
+    int Add(UnwrappedTSN tsn, Data data);
+    size_t EraseTo(SSN ssn);
+    void Reset() {
+      ssn_unwrapper_.Reset();
+      next_ssn_ = ssn_unwrapper_.Unwrap(SSN(0));
+    }
+    SSN next_ssn() const { return next_ssn_.Wrap(); }
+    bool has_unassembled_chunks() const { return !chunks_by_ssn_.empty(); }
+
+   private:
+    // Try to assemble one or several messages in order from the stream.
+    // Returns the number of bytes assembled if a message was assembled.
+    size_t TryToAssembleMessage();
+    size_t TryToAssembleMessages();
+    // This must be an ordered container to be able to iterate in SSN order.
+    std::map<UnwrappedSSN, ChunkMap> chunks_by_ssn_;
+    UnwrappedSSN::Unwrapper ssn_unwrapper_;
+    UnwrappedSSN next_ssn_;
+  };
+
+  const std::string log_prefix_;
+
+  // Callback for when a message has been assembled.
+  const OnAssembledMessage on_assembled_message_;
+
+  // All unordered and ordered streams, managing not-yet-assembled data.
+  std::map<StreamID, UnorderedStream> unordered_streams_;
+  std::map<StreamID, OrderedStream> ordered_streams_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_RX_TRADITIONAL_REASSEMBLY_STREAMS_H_
diff --git a/third_party/libwebrtc/net/dcsctp/rx/traditional_reassembly_streams_test.cc b/third_party/libwebrtc/net/dcsctp/rx/traditional_reassembly_streams_test.cc
new file mode 100644
index 0000000000..341870442d
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/rx/traditional_reassembly_streams_test.cc
@@ -0,0 +1,257 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/rx/traditional_reassembly_streams.h"
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "net/dcsctp/common/handover_testing.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/rx/reassembly_streams.h"
+#include "net/dcsctp/testing/data_generator.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+using ::testing::MockFunction;
+using ::testing::NiceMock;
+using ::testing::Property;
+
+class TraditionalReassemblyStreamsTest : public testing::Test {
+ protected:
+  UnwrappedTSN tsn(uint32_t value) { return tsn_.Unwrap(TSN(value)); }
+
+  TraditionalReassemblyStreamsTest() {}
+  DataGenerator gen_;
+  UnwrappedTSN::Unwrapper tsn_;
+};
+
+TEST_F(TraditionalReassemblyStreamsTest,
+       AddUnorderedMessageReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  TraditionalReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Unordered({1}, "B")), 1);
+  EXPECT_EQ(streams.Add(tsn(2), gen_.Unordered({2, 3, 4})), 3);
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Unordered({5, 6})), 2);
+  // Adding the end fragment should make it empty again.
+  EXPECT_EQ(streams.Add(tsn(4), gen_.Unordered({7}, "E")), -6);
+}
+
+TEST_F(TraditionalReassemblyStreamsTest,
+       AddSimpleOrderedMessageReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  TraditionalReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  EXPECT_EQ(streams.Add(tsn(2), gen_.Ordered({2, 3, 4})), 3);
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+  EXPECT_EQ(streams.Add(tsn(4), gen_.Ordered({7}, "E")), -6);
+}
+
+TEST_F(TraditionalReassemblyStreamsTest,
+       AddMoreComplexOrderedMessageReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  TraditionalReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  Data late = gen_.Ordered({2, 3, 4});
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+  EXPECT_EQ(streams.Add(tsn(4), gen_.Ordered({7}, "E")), 1);
+
+  EXPECT_EQ(streams.Add(tsn(5), gen_.Ordered({1}, "BE")), 1);
+  EXPECT_EQ(streams.Add(tsn(6), gen_.Ordered({5, 6}, "B")), 2);
+  EXPECT_EQ(streams.Add(tsn(7), gen_.Ordered({7}, "E")), 1);
+  EXPECT_EQ(streams.Add(tsn(2), std::move(late)), -8);
+}
+
+TEST_F(TraditionalReassemblyStreamsTest,
+       DeleteUnorderedMessageReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  TraditionalReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Unordered({1}, "B")), 1);
+  EXPECT_EQ(streams.Add(tsn(2), gen_.Unordered({2, 3, 4})), 3);
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Unordered({5, 6})), 2);
+
+  EXPECT_EQ(streams.HandleForwardTsn(tsn(3), {}), 6u);
+}
+
+TEST_F(TraditionalReassemblyStreamsTest,
+       DeleteSimpleOrderedMessageReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  TraditionalReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  EXPECT_EQ(streams.Add(tsn(2), gen_.Ordered({2, 3, 4})), 3);
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+
+  ForwardTsnChunk::SkippedStream skipped[] = {
+      ForwardTsnChunk::SkippedStream(StreamID(1), SSN(0))};
+  EXPECT_EQ(streams.HandleForwardTsn(tsn(3), skipped), 6u);
+}
+
+TEST_F(TraditionalReassemblyStreamsTest,
+       DeleteManyOrderedMessagesReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  TraditionalReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  gen_.Ordered({2, 3, 4});
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+  EXPECT_EQ(streams.Add(tsn(4), gen_.Ordered({7}, "E")), 1);
+
+  EXPECT_EQ(streams.Add(tsn(5), gen_.Ordered({1}, "BE")), 1);
+  EXPECT_EQ(streams.Add(tsn(6), gen_.Ordered({5, 6}, "B")), 2);
+  EXPECT_EQ(streams.Add(tsn(7), gen_.Ordered({7}, "E")), 1);
+
+  // Expire all three messages
+  ForwardTsnChunk::SkippedStream skipped[] = {
+      ForwardTsnChunk::SkippedStream(StreamID(1), SSN(2))};
+  EXPECT_EQ(streams.HandleForwardTsn(tsn(8), skipped), 8u);
+}
+
+TEST_F(TraditionalReassemblyStreamsTest,
+       DeleteOrderedMessageDelivesTwoReturnsCorrectSize) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  TraditionalReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  gen_.Ordered({2, 3, 4});
+  EXPECT_EQ(streams.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+  EXPECT_EQ(streams.Add(tsn(4), gen_.Ordered({7}, "E")), 1);
+
+  EXPECT_EQ(streams.Add(tsn(5), gen_.Ordered({1}, "BE")), 1);
+  EXPECT_EQ(streams.Add(tsn(6), gen_.Ordered({5, 6}, "B")), 2);
+  EXPECT_EQ(streams.Add(tsn(7), gen_.Ordered({7}, "E")), 1);
+
+  // The first ordered message expire, and the following two are delivered.
+  ForwardTsnChunk::SkippedStream skipped[] = {
+      ForwardTsnChunk::SkippedStream(StreamID(1), SSN(0))};
+  EXPECT_EQ(streams.HandleForwardTsn(tsn(4), skipped), 8u);
+}
+
+TEST_F(TraditionalReassemblyStreamsTest, NoStreamsCanBeHandedOver) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  TraditionalReassemblyStreams streams1("", on_assembled.AsStdFunction());
+  EXPECT_TRUE(streams1.GetHandoverReadiness().IsReady());
+
+  DcSctpSocketHandoverState state;
+  streams1.AddHandoverState(state);
+  g_handover_state_transformer_for_test(&state);
+  TraditionalReassemblyStreams streams2("", on_assembled.AsStdFunction());
+  streams2.RestoreFromState(state);
+
+  EXPECT_EQ(streams2.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  EXPECT_EQ(streams2.Add(tsn(2), gen_.Ordered({2, 3, 4})), 3);
+  EXPECT_EQ(streams2.Add(tsn(1), gen_.Unordered({1}, "B")), 1);
+  EXPECT_EQ(streams2.Add(tsn(2), gen_.Unordered({2, 3, 4})), 3);
+}
+
+TEST_F(TraditionalReassemblyStreamsTest,
+       OrderedStreamsCanBeHandedOverWhenNoUnassembledChunksExist) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  TraditionalReassemblyStreams streams1("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams1.Add(tsn(1), gen_.Ordered({1}, "B")), 1);
+  EXPECT_EQ(streams1.GetHandoverReadiness(),
+            HandoverReadinessStatus(
+                HandoverUnreadinessReason::kOrderedStreamHasUnassembledChunks));
+  EXPECT_EQ(streams1.Add(tsn(2), gen_.Ordered({2, 3, 4})), 3);
+  EXPECT_EQ(streams1.GetHandoverReadiness(),
+            HandoverReadinessStatus(
+                HandoverUnreadinessReason::kOrderedStreamHasUnassembledChunks));
+  EXPECT_EQ(streams1.Add(tsn(3), gen_.Ordered({5, 6})), 2);
+  EXPECT_EQ(streams1.GetHandoverReadiness(),
+            HandoverReadinessStatus(
+                HandoverUnreadinessReason::kOrderedStreamHasUnassembledChunks));
+
+  ForwardTsnChunk::SkippedStream skipped[] = {
+      ForwardTsnChunk::SkippedStream(StreamID(1), SSN(0))};
+  EXPECT_EQ(streams1.HandleForwardTsn(tsn(3), skipped), 6u);
+  EXPECT_TRUE(streams1.GetHandoverReadiness().IsReady());
+
+  DcSctpSocketHandoverState state;
+  streams1.AddHandoverState(state);
+  g_handover_state_transformer_for_test(&state);
+  TraditionalReassemblyStreams streams2("", on_assembled.AsStdFunction());
+  streams2.RestoreFromState(state);
+  EXPECT_EQ(streams2.Add(tsn(4), gen_.Ordered({7})), 1);
+}
+
+TEST_F(TraditionalReassemblyStreamsTest,
+       UnorderedStreamsCanBeHandedOverWhenNoUnassembledChunksExist) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+
+  TraditionalReassemblyStreams streams1("", on_assembled.AsStdFunction());
+
+  EXPECT_EQ(streams1.Add(tsn(1), gen_.Unordered({1}, "B")), 1);
+  EXPECT_EQ(
+      streams1.GetHandoverReadiness(),
+      HandoverReadinessStatus(
+          HandoverUnreadinessReason::kUnorderedStreamHasUnassembledChunks));
+  EXPECT_EQ(streams1.Add(tsn(2), gen_.Unordered({2, 3, 4})), 3);
+  EXPECT_EQ(
+      streams1.GetHandoverReadiness(),
+      HandoverReadinessStatus(
+          HandoverUnreadinessReason::kUnorderedStreamHasUnassembledChunks));
+  EXPECT_EQ(streams1.Add(tsn(3), gen_.Unordered({5, 6})), 2);
+  EXPECT_EQ(
+      streams1.GetHandoverReadiness(),
+      HandoverReadinessStatus(
+          HandoverUnreadinessReason::kUnorderedStreamHasUnassembledChunks));
+
+  EXPECT_EQ(streams1.HandleForwardTsn(tsn(3), {}), 6u);
+  EXPECT_TRUE(streams1.GetHandoverReadiness().IsReady());
+
+  DcSctpSocketHandoverState state;
+  streams1.AddHandoverState(state);
+  g_handover_state_transformer_for_test(&state);
+  TraditionalReassemblyStreams streams2("", on_assembled.AsStdFunction());
+  streams2.RestoreFromState(state);
+  EXPECT_EQ(streams2.Add(tsn(4), gen_.Unordered({7})), 1);
+}
+
+TEST_F(TraditionalReassemblyStreamsTest, CanDeleteFirstOrderedMessage) {
+  NiceMock<MockFunction<ReassemblyStreams::OnAssembledMessage>> on_assembled;
+  EXPECT_CALL(on_assembled,
+              Call(ElementsAre(tsn(2)),
+                   Property(&DcSctpMessage::payload, ElementsAre(2, 3, 4))));
+
+  TraditionalReassemblyStreams streams("", on_assembled.AsStdFunction());
+
+  // Not received, SID=1. TSN=1, SSN=0
+  gen_.Ordered({1}, "BE");
+  // And deleted (SID=1, TSN=1, SSN=0)
+  ForwardTsnChunk::SkippedStream skipped[] = {
+      ForwardTsnChunk::SkippedStream(StreamID(1), SSN(0))};
+  EXPECT_EQ(streams.HandleForwardTsn(tsn(1), skipped), 0u);
+
+  // Receive SID=1, TSN=2, SSN=1
+  EXPECT_EQ(streams.Add(tsn(2), gen_.Ordered({2, 3, 4}, "BE")), 0);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/BUILD.gn b/third_party/libwebrtc/net/dcsctp/socket/BUILD.gn
new file mode 100644
index 0000000000..92ce413d0d
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/BUILD.gn
@@ -0,0 +1,278 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("context") {
+  sources = [ "context.h" ]
+  deps = [
+    "../common:internal_types",
+    "../packet:sctp_packet",
+    "../public:socket",
+    "../public:types",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_library("heartbeat_handler") {
+  deps = [
+    ":context",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../packet:bounded_io",
+    "../packet:chunk",
+    "../packet:parameter",
+    "../packet:sctp_packet",
+    "../public:socket",
+    "../public:types",
+    "../timer",
+  ]
+  sources = [
+    "heartbeat_handler.cc",
+    "heartbeat_handler.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/functional:bind_front",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("stream_reset_handler") {
+  deps = [
+    ":context",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base/containers:flat_set",
+    "../common:internal_types",
+    "../common:str_join",
+    "../packet:chunk",
+    "../packet:parameter",
+    "../packet:sctp_packet",
+    "../packet:tlv_trait",
+    "../public:socket",
+    "../public:types",
+    "../rx:data_tracker",
+    "../rx:reassembly_queue",
+    "../timer",
+    "../tx:retransmission_queue",
+  ]
+  sources = [
+    "stream_reset_handler.cc",
+    "stream_reset_handler.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/functional:bind_front",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("packet_sender") {
+  deps = [
+    "../packet:sctp_packet",
+    "../public:socket",
+    "../public:types",
+    "../timer",
+  ]
+  sources = [
+    "packet_sender.cc",
+    "packet_sender.h",
+  ]
+  absl_deps = []
+}
+
+rtc_library("transmission_control_block") {
+  deps = [
+    ":context",
+    ":heartbeat_handler",
+    ":packet_sender",
+    ":stream_reset_handler",
+    "../../../api:array_view",
+    "../../../api/task_queue:task_queue",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:stringutils",
+    "../common:sequence_numbers",
+    "../packet:chunk",
+    "../packet:sctp_packet",
+    "../public:socket",
+    "../public:types",
+    "../rx:data_tracker",
+    "../rx:reassembly_queue",
+    "../timer",
+    "../tx:retransmission_error_counter",
+    "../tx:retransmission_queue",
+    "../tx:retransmission_timeout",
+    "../tx:send_queue",
+  ]
+  sources = [
+    "capabilities.h",
+    "transmission_control_block.cc",
+    "transmission_control_block.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/functional:bind_front",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("dcsctp_socket") {
+  deps = [
+    ":context",
+    ":heartbeat_handler",
+    ":packet_sender",
+    ":stream_reset_handler",
+    ":transmission_control_block",
+    "../../../api:array_view",
+    "../../../api:make_ref_counted",
+    "../../../api:refcountedbase",
+    "../../../api:scoped_refptr",
+    "../../../api:sequence_checker",
+    "../../../api/task_queue:task_queue",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:stringutils",
+    "../common:internal_types",
+    "../packet:bounded_io",
+    "../packet:chunk",
+    "../packet:chunk_validators",
+    "../packet:data",
+    "../packet:error_cause",
+    "../packet:parameter",
+    "../packet:sctp_packet",
+    "../packet:tlv_trait",
+    "../public:socket",
+    "../public:types",
+    "../rx:data_tracker",
+    "../rx:reassembly_queue",
+    "../timer",
+    "../tx:retransmission_error_counter",
+    "../tx:retransmission_queue",
+    "../tx:retransmission_timeout",
+    "../tx:rr_send_queue",
+    "../tx:send_queue",
+  ]
+  sources = [
+    "callback_deferrer.cc",
+    "callback_deferrer.h",
+    "dcsctp_socket.cc",
+    "dcsctp_socket.h",
+    "state_cookie.cc",
+    "state_cookie.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/functional:bind_front",
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_source_set("mock_callbacks") {
+    testonly = true
+    sources = [ "mock_dcsctp_socket_callbacks.h" ]
+    deps = [
+      "../../../api:array_view",
+      "../../../api/task_queue:task_queue",
+      "../../../rtc_base:logging",
+      "../../../rtc_base:random",
+      "../../../test:test_support",
+      "../public:socket",
+      "../public:types",
+      "../timer",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+
+  rtc_source_set("mock_context") {
+    testonly = true
+    sources = [ "mock_context.h" ]
+    deps = [
+      ":context",
+      ":mock_callbacks",
+      "../../../test:test_support",
+      "../common:internal_types",
+      "../packet:sctp_packet",
+      "../public:socket",
+      "../public:types",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+
+  rtc_library("dcsctp_socket_unittests") {
+    testonly = true
+
+    deps = [
+      ":dcsctp_socket",
+      ":heartbeat_handler",
+      ":mock_callbacks",
+      ":mock_context",
+      ":packet_sender",
+      ":stream_reset_handler",
+      "../../../api:array_view",
+      "../../../api:create_network_emulation_manager",
+      "../../../api:network_emulation_manager_api",
+      "../../../api/task_queue",
+      "../../../api/task_queue:pending_task_safety_flag",
+      "../../../api/units:time_delta",
+      "../../../call:simulated_network",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:copy_on_write_buffer",
+      "../../../rtc_base:gunit_helpers",
+      "../../../rtc_base:logging",
+      "../../../rtc_base:rtc_base_tests_utils",
+      "../../../rtc_base:socket_address",
+      "../../../rtc_base:stringutils",
+      "../../../rtc_base:timeutils",
+      "../../../test:test_support",
+      "../common:handover_testing",
+      "../common:internal_types",
+      "../packet:chunk",
+      "../packet:error_cause",
+      "../packet:parameter",
+      "../packet:sctp_packet",
+      "../packet:tlv_trait",
+      "../public:socket",
+      "../public:types",
+      "../public:utils",
+      "../rx:data_tracker",
+      "../rx:reassembly_queue",
+      "../testing:data_generator",
+      "../testing:testing_macros",
+      "../timer",
+      "../timer:task_queue_timeout",
+      "../tx:mock_send_queue",
+      "../tx:retransmission_queue",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/memory",
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+    sources = [
+      "dcsctp_socket_network_test.cc",
+      "dcsctp_socket_test.cc",
+      "heartbeat_handler_test.cc",
+      "packet_sender_test.cc",
+      "state_cookie_test.cc",
+      "stream_reset_handler_test.cc",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/net/dcsctp/socket/DEPS b/third_party/libwebrtc/net/dcsctp/socket/DEPS
new file mode 100644
index 0000000000..d4966290e3
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/DEPS
@@ -0,0 +1,5 @@
+specific_include_rules = {
+  "dcsctp_socket_network_test.cc": [
+    "+call",
+  ]
+}
diff --git a/third_party/libwebrtc/net/dcsctp/socket/callback_deferrer.cc b/third_party/libwebrtc/net/dcsctp/socket/callback_deferrer.cc
new file mode 100644
index 0000000000..123526e782
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/callback_deferrer.cc
@@ -0,0 +1,181 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/callback_deferrer.h"
+
+#include "api/make_ref_counted.h"
+
+namespace dcsctp {
+namespace {
+// A wrapper around the move-only DcSctpMessage, to let it be captured in a
+// lambda.
+class MessageDeliverer {
+ public:
+  explicit MessageDeliverer(DcSctpMessage&& message)
+      : state_(rtc::make_ref_counted<State>(std::move(message))) {}
+
+  void Deliver(DcSctpSocketCallbacks& c) {
+    // Really ensure that it's only called once.
+    RTC_DCHECK(!state_->has_delivered);
+    state_->has_delivered = true;
+    c.OnMessageReceived(std::move(state_->message));
+  }
+
+ private:
+  struct State : public rtc::RefCountInterface {
+    explicit State(DcSctpMessage&& m)
+        : has_delivered(false), message(std::move(m)) {}
+    bool has_delivered;
+    DcSctpMessage message;
+  };
+  rtc::scoped_refptr<State> state_;
+};
+}  // namespace
+
+void CallbackDeferrer::Prepare() {
+  RTC_DCHECK(!prepared_);
+  prepared_ = true;
+}
+
+void CallbackDeferrer::TriggerDeferred() {
+  // Need to swap here. The client may call into the library from within a
+  // callback, and that might result in adding new callbacks to this instance,
+  // and the vector can't be modified while iterated on.
+  RTC_DCHECK(prepared_);
+  std::vector<std::function<void(DcSctpSocketCallbacks & cb)>> deferred;
+  deferred.swap(deferred_);
+  prepared_ = false;
+
+  for (auto& cb : deferred) {
+    cb(underlying_);
+  }
+}
+
+SendPacketStatus CallbackDeferrer::SendPacketWithStatus(
+    rtc::ArrayView<const uint8_t> data) {
+  // Will not be deferred - call directly.
+  return underlying_.SendPacketWithStatus(data);
+}
+
+std::unique_ptr<Timeout> CallbackDeferrer::CreateTimeout(
+    webrtc::TaskQueueBase::DelayPrecision precision) {
+  // Will not be deferred - call directly.
+  return underlying_.CreateTimeout(precision);
+}
+
+TimeMs CallbackDeferrer::TimeMillis() {
+  // Will not be deferred - call directly.
+  return underlying_.TimeMillis();
+}
+
+uint32_t CallbackDeferrer::GetRandomInt(uint32_t low, uint32_t high) {
+  // Will not be deferred - call directly.
+  return underlying_.GetRandomInt(low, high);
+}
+
+void CallbackDeferrer::OnMessageReceived(DcSctpMessage message) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      [deliverer = MessageDeliverer(std::move(message))](
+          DcSctpSocketCallbacks& cb) mutable { deliverer.Deliver(cb); });
+}
+
+void CallbackDeferrer::OnError(ErrorKind error, absl::string_view message) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      [error, message = std::string(message)](DcSctpSocketCallbacks& cb) {
+        cb.OnError(error, message);
+      });
+}
+
+void CallbackDeferrer::OnAborted(ErrorKind error, absl::string_view message) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      [error, message = std::string(message)](DcSctpSocketCallbacks& cb) {
+        cb.OnAborted(error, message);
+      });
+}
+
+void CallbackDeferrer::OnConnected() {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back([](DcSctpSocketCallbacks& cb) { cb.OnConnected(); });
+}
+
+void CallbackDeferrer::OnClosed() {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back([](DcSctpSocketCallbacks& cb) { cb.OnClosed(); });
+}
+
+void CallbackDeferrer::OnConnectionRestarted() {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      [](DcSctpSocketCallbacks& cb) { cb.OnConnectionRestarted(); });
+}
+
+void CallbackDeferrer::OnStreamsResetFailed(
+    rtc::ArrayView<const StreamID> outgoing_streams,
+    absl::string_view reason) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      [streams = std::vector<StreamID>(outgoing_streams.begin(),
+                                       outgoing_streams.end()),
+       reason = std::string(reason)](DcSctpSocketCallbacks& cb) {
+        cb.OnStreamsResetFailed(streams, reason);
+      });
+}
+
+void CallbackDeferrer::OnStreamsResetPerformed(
+    rtc::ArrayView<const StreamID> outgoing_streams) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      [streams = std::vector<StreamID>(outgoing_streams.begin(),
+                                       outgoing_streams.end())](
+          DcSctpSocketCallbacks& cb) { cb.OnStreamsResetPerformed(streams); });
+}
+
+void CallbackDeferrer::OnIncomingStreamsReset(
+    rtc::ArrayView<const StreamID> incoming_streams) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      [streams = std::vector<StreamID>(incoming_streams.begin(),
+                                       incoming_streams.end())](
+          DcSctpSocketCallbacks& cb) { cb.OnIncomingStreamsReset(streams); });
+}
+
+void CallbackDeferrer::OnBufferedAmountLow(StreamID stream_id) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back([stream_id](DcSctpSocketCallbacks& cb) {
+    cb.OnBufferedAmountLow(stream_id);
+  });
+}
+
+void CallbackDeferrer::OnTotalBufferedAmountLow() {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      [](DcSctpSocketCallbacks& cb) { cb.OnTotalBufferedAmountLow(); });
+}
+
+void CallbackDeferrer::OnLifecycleMessageExpired(LifecycleId lifecycle_id,
+                                                 bool maybe_delivered) {
+  // Will not be deferred - call directly.
+  underlying_.OnLifecycleMessageExpired(lifecycle_id, maybe_delivered);
+}
+void CallbackDeferrer::OnLifecycleMessageFullySent(LifecycleId lifecycle_id) {
+  // Will not be deferred - call directly.
+  underlying_.OnLifecycleMessageFullySent(lifecycle_id);
+}
+void CallbackDeferrer::OnLifecycleMessageDelivered(LifecycleId lifecycle_id) {
+  // Will not be deferred - call directly.
+  underlying_.OnLifecycleMessageDelivered(lifecycle_id);
+}
+void CallbackDeferrer::OnLifecycleEnd(LifecycleId lifecycle_id) {
+  // Will not be deferred - call directly.
+  underlying_.OnLifecycleEnd(lifecycle_id);
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/callback_deferrer.h b/third_party/libwebrtc/net/dcsctp/socket/callback_deferrer.h
new file mode 100644
index 0000000000..1c35dda6cf
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/callback_deferrer.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_CALLBACK_DEFERRER_H_
+#define NET_DCSCTP_SOCKET_CALLBACK_DEFERRER_H_
+
+#include <cstdint>
+#include <functional>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+
+namespace dcsctp {
+// Defers callbacks until they can be safely triggered.
+//
+// There are a lot of callbacks from the dcSCTP library to the client,
+// such as when messages are received or streams are closed. When the client
+// receives these callbacks, the client is expected to be able to call into the
+// library - from within the callback. For example, sending a reply message when
+// a certain SCTP message has been received, or to reconnect when the connection
+// was closed for any reason. This means that the dcSCTP library must always be
+// in a consistent and stable state when these callbacks are delivered, and to
+// ensure that's the case, callbacks are not immediately delivered from where
+// they originate, but instead queued (deferred) by this class. At the end of
+// any public API method that may result in callbacks, they are triggered and
+// then delivered.
+//
+// There are a number of exceptions, which is clearly annotated in the API.
+class CallbackDeferrer : public DcSctpSocketCallbacks {
+ public:
+  class ScopedDeferrer {
+   public:
+    explicit ScopedDeferrer(CallbackDeferrer& callback_deferrer)
+        : callback_deferrer_(callback_deferrer) {
+      callback_deferrer_.Prepare();
+    }
+
+    ~ScopedDeferrer() { callback_deferrer_.TriggerDeferred(); }
+
+   private:
+    CallbackDeferrer& callback_deferrer_;
+  };
+
+  explicit CallbackDeferrer(DcSctpSocketCallbacks& underlying)
+      : underlying_(underlying) {}
+
+  // Implementation of DcSctpSocketCallbacks
+  SendPacketStatus SendPacketWithStatus(
+      rtc::ArrayView<const uint8_t> data) override;
+  std::unique_ptr<Timeout> CreateTimeout(
+      webrtc::TaskQueueBase::DelayPrecision precision) override;
+  TimeMs TimeMillis() override;
+  uint32_t GetRandomInt(uint32_t low, uint32_t high) override;
+  void OnMessageReceived(DcSctpMessage message) override;
+  void OnError(ErrorKind error, absl::string_view message) override;
+  void OnAborted(ErrorKind error, absl::string_view message) override;
+  void OnConnected() override;
+  void OnClosed() override;
+  void OnConnectionRestarted() override;
+  void OnStreamsResetFailed(rtc::ArrayView<const StreamID> outgoing_streams,
+                            absl::string_view reason) override;
+  void OnStreamsResetPerformed(
+      rtc::ArrayView<const StreamID> outgoing_streams) override;
+  void OnIncomingStreamsReset(
+      rtc::ArrayView<const StreamID> incoming_streams) override;
+  void OnBufferedAmountLow(StreamID stream_id) override;
+  void OnTotalBufferedAmountLow() override;
+
+  void OnLifecycleMessageExpired(LifecycleId lifecycle_id,
+                                 bool maybe_delivered) override;
+  void OnLifecycleMessageFullySent(LifecycleId lifecycle_id) override;
+  void OnLifecycleMessageDelivered(LifecycleId lifecycle_id) override;
+  void OnLifecycleEnd(LifecycleId lifecycle_id) override;
+
+ private:
+  void Prepare();
+  void TriggerDeferred();
+
+  DcSctpSocketCallbacks& underlying_;
+  bool prepared_ = false;
+  std::vector<std::function<void(DcSctpSocketCallbacks& cb)>> deferred_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_CALLBACK_DEFERRER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/socket/capabilities.h b/third_party/libwebrtc/net/dcsctp/socket/capabilities.h
new file mode 100644
index 0000000000..fa3be37d12
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/capabilities.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_CAPABILITIES_H_
+#define NET_DCSCTP_SOCKET_CAPABILITIES_H_
+
+#include <cstdint>
+namespace dcsctp {
+// Indicates what the association supports, meaning that both parties
+// support it and that feature can be used.
+struct Capabilities {
+  // RFC3758 Partial Reliability Extension
+  bool partial_reliability = false;
+  // RFC8260 Stream Schedulers and User Message Interleaving
+  bool message_interleaving = false;
+  // RFC6525 Stream Reconfiguration
+  bool reconfig = false;
+  // Negotiated maximum incoming and outgoing stream count.
+  uint16_t negotiated_maximum_incoming_streams = 0;
+  uint16_t negotiated_maximum_outgoing_streams = 0;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_CAPABILITIES_H_
diff --git a/third_party/libwebrtc/net/dcsctp/socket/context.h b/third_party/libwebrtc/net/dcsctp/socket/context.h
new file mode 100644
index 0000000000..eca5b9e4fb
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/context.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_CONTEXT_H_
+#define NET_DCSCTP_SOCKET_CONTEXT_H_
+
+#include <cstdint>
+
+#include "absl/strings/string_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// A set of helper methods used by handlers to e.g. send packets.
+//
+// Implemented by the TransmissionControlBlock.
+class Context {
+ public:
+  virtual ~Context() = default;
+
+  // Indicates if a connection has been established.
+  virtual bool is_connection_established() const = 0;
+
+  // Returns this side's initial TSN value.
+  virtual TSN my_initial_tsn() const = 0;
+
+  // Returns the peer's initial TSN value.
+  virtual TSN peer_initial_tsn() const = 0;
+
+  // Returns the socket callbacks.
+  virtual DcSctpSocketCallbacks& callbacks() const = 0;
+
+  // Observes a measured RTT value, in milliseconds.
+  virtual void ObserveRTT(DurationMs rtt_ms) = 0;
+
+  // Returns the current Retransmission Timeout (rto) value, in milliseconds.
+  virtual DurationMs current_rto() const = 0;
+
+  // Increments the transmission error counter, given a human readable reason.
+  virtual bool IncrementTxErrorCounter(absl::string_view reason) = 0;
+
+  // Clears the transmission error counter.
+  virtual void ClearTxErrorCounter() = 0;
+
+  // Returns true if there have been too many retransmission errors.
+  virtual bool HasTooManyTxErrors() const = 0;
+
+  // Returns a PacketBuilder, filled in with the correct verification tag.
+  virtual SctpPacket::Builder PacketBuilder() const = 0;
+
+  // Builds the packet from `builder` and sends it.
+  virtual void Send(SctpPacket::Builder& builder) = 0;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_CONTEXT_H_
diff --git a/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket.cc b/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket.cc
new file mode 100644
index 0000000000..f831ba090c
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket.cc
@@ -0,0 +1,1781 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/dcsctp_socket.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <limits>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/bind_front.h"
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/packet/chunk/abort_chunk.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/data_common.h"
+#include "net/dcsctp/packet/chunk/error_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/init_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/init_chunk.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_complete_chunk.h"
+#include "net/dcsctp/packet/chunk_validators.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/error_cause/no_user_data_cause.h"
+#include "net/dcsctp/packet/error_cause/out_of_resource_error_cause.h"
+#include "net/dcsctp/packet/error_cause/protocol_violation_cause.h"
+#include "net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h"
+#include "net/dcsctp/packet/error_cause/user_initiated_abort_cause.h"
+#include "net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/parameter/state_cookie_parameter.h"
+#include "net/dcsctp/packet/parameter/supported_extensions_parameter.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/packet_observer.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/callback_deferrer.h"
+#include "net/dcsctp/socket/capabilities.h"
+#include "net/dcsctp/socket/heartbeat_handler.h"
+#include "net/dcsctp/socket/state_cookie.h"
+#include "net/dcsctp/socket/stream_reset_handler.h"
+#include "net/dcsctp/socket/transmission_control_block.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "net/dcsctp/tx/send_queue.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/strings/string_format.h"
+
+namespace dcsctp {
+namespace {
+
+// https://tools.ietf.org/html/rfc4960#section-5.1
+constexpr uint32_t kMinVerificationTag = 1;
+constexpr uint32_t kMaxVerificationTag = std::numeric_limits<uint32_t>::max();
+
+// https://tools.ietf.org/html/rfc4960#section-3.3.2
+constexpr uint32_t kMinInitialTsn = 0;
+constexpr uint32_t kMaxInitialTsn = std::numeric_limits<uint32_t>::max();
+
+Capabilities ComputeCapabilities(const DcSctpOptions& options,
+                                 uint16_t peer_nbr_outbound_streams,
+                                 uint16_t peer_nbr_inbound_streams,
+                                 const Parameters& parameters) {
+  Capabilities capabilities;
+  absl::optional<SupportedExtensionsParameter> supported_extensions =
+      parameters.get<SupportedExtensionsParameter>();
+
+  if (options.enable_partial_reliability) {
+    capabilities.partial_reliability =
+        parameters.get<ForwardTsnSupportedParameter>().has_value();
+    if (supported_extensions.has_value()) {
+      capabilities.partial_reliability |=
+          supported_extensions->supports(ForwardTsnChunk::kType);
+    }
+  }
+
+  if (options.enable_message_interleaving && supported_extensions.has_value()) {
+    capabilities.message_interleaving =
+        supported_extensions->supports(IDataChunk::kType) &&
+        supported_extensions->supports(IForwardTsnChunk::kType);
+  }
+  if (supported_extensions.has_value() &&
+      supported_extensions->supports(ReConfigChunk::kType)) {
+    capabilities.reconfig = true;
+  }
+
+  capabilities.negotiated_maximum_incoming_streams = std::min(
+      options.announced_maximum_incoming_streams, peer_nbr_outbound_streams);
+  capabilities.negotiated_maximum_outgoing_streams = std::min(
+      options.announced_maximum_outgoing_streams, peer_nbr_inbound_streams);
+
+  return capabilities;
+}
+
+void AddCapabilityParameters(const DcSctpOptions& options,
+                             Parameters::Builder& builder) {
+  std::vector<uint8_t> chunk_types = {ReConfigChunk::kType};
+
+  if (options.enable_partial_reliability) {
+    builder.Add(ForwardTsnSupportedParameter());
+    chunk_types.push_back(ForwardTsnChunk::kType);
+  }
+  if (options.enable_message_interleaving) {
+    chunk_types.push_back(IDataChunk::kType);
+    chunk_types.push_back(IForwardTsnChunk::kType);
+  }
+  builder.Add(SupportedExtensionsParameter(std::move(chunk_types)));
+}
+
+TieTag MakeTieTag(DcSctpSocketCallbacks& cb) {
+  uint32_t tie_tag_upper =
+      cb.GetRandomInt(0, std::numeric_limits<uint32_t>::max());
+  uint32_t tie_tag_lower =
+      cb.GetRandomInt(1, std::numeric_limits<uint32_t>::max());
+  return TieTag(static_cast<uint64_t>(tie_tag_upper) << 32 |
+                static_cast<uint64_t>(tie_tag_lower));
+}
+
+SctpImplementation DeterminePeerImplementation(
+    rtc::ArrayView<const uint8_t> cookie) {
+  if (cookie.size() > 8) {
+    absl::string_view magic(reinterpret_cast<const char*>(cookie.data()), 8);
+    if (magic == "dcSCTP00") {
+      return SctpImplementation::kDcsctp;
+    }
+    if (magic == "KAME-BSD") {
+      return SctpImplementation::kUsrSctp;
+    }
+  }
+  return SctpImplementation::kOther;
+}
+}  // namespace
+
+DcSctpSocket::DcSctpSocket(absl::string_view log_prefix,
+                           DcSctpSocketCallbacks& callbacks,
+                           std::unique_ptr<PacketObserver> packet_observer,
+                           const DcSctpOptions& options)
+    : log_prefix_(std::string(log_prefix) + ": "),
+      packet_observer_(std::move(packet_observer)),
+      options_(options),
+      callbacks_(callbacks),
+      timer_manager_([this](webrtc::TaskQueueBase::DelayPrecision precision) {
+        return callbacks_.CreateTimeout(precision);
+      }),
+      t1_init_(timer_manager_.CreateTimer(
+          "t1-init",
+          absl::bind_front(&DcSctpSocket::OnInitTimerExpiry, this),
+          TimerOptions(options.t1_init_timeout,
+                       TimerBackoffAlgorithm::kExponential,
+                       options.max_init_retransmits))),
+      t1_cookie_(timer_manager_.CreateTimer(
+          "t1-cookie",
+          absl::bind_front(&DcSctpSocket::OnCookieTimerExpiry, this),
+          TimerOptions(options.t1_cookie_timeout,
+                       TimerBackoffAlgorithm::kExponential,
+                       options.max_init_retransmits))),
+      t2_shutdown_(timer_manager_.CreateTimer(
+          "t2-shutdown",
+          absl::bind_front(&DcSctpSocket::OnShutdownTimerExpiry, this),
+          TimerOptions(options.t2_shutdown_timeout,
+                       TimerBackoffAlgorithm::kExponential,
+                       options.max_retransmissions))),
+      packet_sender_(callbacks_,
+                     absl::bind_front(&DcSctpSocket::OnSentPacket, this)),
+      send_queue_(log_prefix_,
+                  &callbacks_,
+                  options_.max_send_buffer_size,
+                  options_.mtu,
+                  options_.default_stream_priority,
+                  options_.total_buffered_amount_low_threshold) {}
+
+std::string DcSctpSocket::log_prefix() const {
+  return log_prefix_ + "[" + std::string(ToString(state_)) + "] ";
+}
+
+bool DcSctpSocket::IsConsistent() const {
+  if (tcb_ != nullptr && tcb_->reassembly_queue().HasMessages()) {
+    return false;
+  }
+  switch (state_) {
+    case State::kClosed:
+      return (tcb_ == nullptr && !t1_init_->is_running() &&
+              !t1_cookie_->is_running() && !t2_shutdown_->is_running());
+    case State::kCookieWait:
+      return (tcb_ == nullptr && t1_init_->is_running() &&
+              !t1_cookie_->is_running() && !t2_shutdown_->is_running());
+    case State::kCookieEchoed:
+      return (tcb_ != nullptr && !t1_init_->is_running() &&
+              t1_cookie_->is_running() && !t2_shutdown_->is_running() &&
+              tcb_->has_cookie_echo_chunk());
+    case State::kEstablished:
+      return (tcb_ != nullptr && !t1_init_->is_running() &&
+              !t1_cookie_->is_running() && !t2_shutdown_->is_running());
+    case State::kShutdownPending:
+      return (tcb_ != nullptr && !t1_init_->is_running() &&
+              !t1_cookie_->is_running() && !t2_shutdown_->is_running());
+    case State::kShutdownSent:
+      return (tcb_ != nullptr && !t1_init_->is_running() &&
+              !t1_cookie_->is_running() && t2_shutdown_->is_running());
+    case State::kShutdownReceived:
+      return (tcb_ != nullptr && !t1_init_->is_running() &&
+              !t1_cookie_->is_running() && !t2_shutdown_->is_running());
+    case State::kShutdownAckSent:
+      return (tcb_ != nullptr && !t1_init_->is_running() &&
+              !t1_cookie_->is_running() && t2_shutdown_->is_running());
+  }
+}
+
+constexpr absl::string_view DcSctpSocket::ToString(DcSctpSocket::State state) {
+  switch (state) {
+    case DcSctpSocket::State::kClosed:
+      return "CLOSED";
+    case DcSctpSocket::State::kCookieWait:
+      return "COOKIE_WAIT";
+    case DcSctpSocket::State::kCookieEchoed:
+      return "COOKIE_ECHOED";
+    case DcSctpSocket::State::kEstablished:
+      return "ESTABLISHED";
+    case DcSctpSocket::State::kShutdownPending:
+      return "SHUTDOWN_PENDING";
+    case DcSctpSocket::State::kShutdownSent:
+      return "SHUTDOWN_SENT";
+    case DcSctpSocket::State::kShutdownReceived:
+      return "SHUTDOWN_RECEIVED";
+    case DcSctpSocket::State::kShutdownAckSent:
+      return "SHUTDOWN_ACK_SENT";
+  }
+}
+
+void DcSctpSocket::SetState(State state, absl::string_view reason) {
+  if (state_ != state) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix_ << "Socket state changed from "
+                         << ToString(state_) << " to " << ToString(state)
+                         << " due to " << reason;
+    state_ = state;
+  }
+}
+
+void DcSctpSocket::SendInit() {
+  Parameters::Builder params_builder;
+  AddCapabilityParameters(options_, params_builder);
+  InitChunk init(/*initiate_tag=*/connect_params_.verification_tag,
+                 /*a_rwnd=*/options_.max_receiver_window_buffer_size,
+                 options_.announced_maximum_outgoing_streams,
+                 options_.announced_maximum_incoming_streams,
+                 connect_params_.initial_tsn, params_builder.Build());
+  SctpPacket::Builder b(VerificationTag(0), options_);
+  b.Add(init);
+  packet_sender_.Send(b);
+}
+
+void DcSctpSocket::MakeConnectionParameters() {
+  VerificationTag new_verification_tag(
+      callbacks_.GetRandomInt(kMinVerificationTag, kMaxVerificationTag));
+  TSN initial_tsn(callbacks_.GetRandomInt(kMinInitialTsn, kMaxInitialTsn));
+  connect_params_.initial_tsn = initial_tsn;
+  connect_params_.verification_tag = new_verification_tag;
+}
+
+void DcSctpSocket::Connect() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
+
+  if (state_ == State::kClosed) {
+    MakeConnectionParameters();
+    RTC_DLOG(LS_INFO)
+        << log_prefix()
+        << rtc::StringFormat(
+               "Connecting. my_verification_tag=%08x, my_initial_tsn=%u",
+               *connect_params_.verification_tag, *connect_params_.initial_tsn);
+    SendInit();
+    t1_init_->Start();
+    SetState(State::kCookieWait, "Connect called");
+  } else {
+    RTC_DLOG(LS_WARNING) << log_prefix()
+                         << "Called Connect on a socket that is not closed";
+  }
+  RTC_DCHECK(IsConsistent());
+}
+
+void DcSctpSocket::CreateTransmissionControlBlock(
+    const Capabilities& capabilities,
+    VerificationTag my_verification_tag,
+    TSN my_initial_tsn,
+    VerificationTag peer_verification_tag,
+    TSN peer_initial_tsn,
+    size_t a_rwnd,
+    TieTag tie_tag) {
+  metrics_.uses_message_interleaving = capabilities.message_interleaving;
+  metrics_.negotiated_maximum_incoming_streams =
+      capabilities.negotiated_maximum_incoming_streams;
+  metrics_.negotiated_maximum_outgoing_streams =
+      capabilities.negotiated_maximum_outgoing_streams;
+  tcb_ = std::make_unique<TransmissionControlBlock>(
+      timer_manager_, log_prefix_, options_, capabilities, callbacks_,
+      send_queue_, my_verification_tag, my_initial_tsn, peer_verification_tag,
+      peer_initial_tsn, a_rwnd, tie_tag, packet_sender_,
+      [this]() { return state_ == State::kEstablished; });
+  RTC_DLOG(LS_VERBOSE) << log_prefix() << "Created TCB: " << tcb_->ToString();
+}
+
+void DcSctpSocket::RestoreFromState(const DcSctpSocketHandoverState& state) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
+
+  if (state_ != State::kClosed) {
+    callbacks_.OnError(ErrorKind::kUnsupportedOperation,
+                       "Only closed socket can be restored from state");
+  } else {
+    if (state.socket_state ==
+        DcSctpSocketHandoverState::SocketState::kConnected) {
+      VerificationTag my_verification_tag =
+          VerificationTag(state.my_verification_tag);
+      connect_params_.verification_tag = my_verification_tag;
+
+      Capabilities capabilities;
+      capabilities.partial_reliability = state.capabilities.partial_reliability;
+      capabilities.message_interleaving =
+          state.capabilities.message_interleaving;
+      capabilities.reconfig = state.capabilities.reconfig;
+      capabilities.negotiated_maximum_incoming_streams =
+          state.capabilities.negotiated_maximum_incoming_streams;
+      capabilities.negotiated_maximum_outgoing_streams =
+          state.capabilities.negotiated_maximum_outgoing_streams;
+
+      send_queue_.RestoreFromState(state);
+
+      CreateTransmissionControlBlock(
+          capabilities, my_verification_tag, TSN(state.my_initial_tsn),
+          VerificationTag(state.peer_verification_tag),
+          TSN(state.peer_initial_tsn), static_cast<size_t>(0),
+          TieTag(state.tie_tag));
+
+      tcb_->RestoreFromState(state);
+
+      SetState(State::kEstablished, "restored from handover state");
+      callbacks_.OnConnected();
+    }
+  }
+
+  RTC_DCHECK(IsConsistent());
+}
+
+void DcSctpSocket::Shutdown() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
+
+  if (tcb_ != nullptr) {
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "Upon receipt of the SHUTDOWN primitive from its upper layer, the
+    // endpoint enters the SHUTDOWN-PENDING state and remains there until all
+    // outstanding data has been acknowledged by its peer."
+
+    // TODO(webrtc:12739): Remove this check, as it just hides the problem that
+    // the socket can transition from ShutdownSent to ShutdownPending, or
+    // ShutdownAckSent to ShutdownPending which is illegal.
+    if (state_ != State::kShutdownSent && state_ != State::kShutdownAckSent) {
+      SetState(State::kShutdownPending, "Shutdown called");
+      t1_init_->Stop();
+      t1_cookie_->Stop();
+      MaybeSendShutdownOrAck();
+    }
+  } else {
+    // Connection closed before even starting to connect, or during the initial
+    // connection phase. There is no outstanding data, so the socket can just
+    // be closed (stopping any connection timers, if any), as this is the
+    // client's intention, by calling Shutdown.
+    InternalClose(ErrorKind::kNoError, "");
+  }
+  RTC_DCHECK(IsConsistent());
+}
+
+void DcSctpSocket::Close() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
+
+  if (state_ != State::kClosed) {
+    if (tcb_ != nullptr) {
+      SctpPacket::Builder b = tcb_->PacketBuilder();
+      b.Add(AbortChunk(/*filled_in_verification_tag=*/true,
+                       Parameters::Builder()
+                           .Add(UserInitiatedAbortCause("Close called"))
+                           .Build()));
+      packet_sender_.Send(b);
+    }
+    InternalClose(ErrorKind::kNoError, "");
+  } else {
+    RTC_DLOG(LS_INFO) << log_prefix() << "Called Close on a closed socket";
+  }
+  RTC_DCHECK(IsConsistent());
+}
+
+void DcSctpSocket::CloseConnectionBecauseOfTooManyTransmissionErrors() {
+  packet_sender_.Send(tcb_->PacketBuilder().Add(AbortChunk(
+      true, Parameters::Builder()
+                .Add(UserInitiatedAbortCause("Too many retransmissions"))
+                .Build())));
+  InternalClose(ErrorKind::kTooManyRetries, "Too many retransmissions");
+}
+
+void DcSctpSocket::InternalClose(ErrorKind error, absl::string_view message) {
+  if (state_ != State::kClosed) {
+    t1_init_->Stop();
+    t1_cookie_->Stop();
+    t2_shutdown_->Stop();
+    tcb_ = nullptr;
+
+    if (error == ErrorKind::kNoError) {
+      callbacks_.OnClosed();
+    } else {
+      callbacks_.OnAborted(error, message);
+    }
+    SetState(State::kClosed, message);
+  }
+  // This method's purpose is to abort/close and make it consistent by ensuring
+  // that e.g. all timers really are stopped.
+  RTC_DCHECK(IsConsistent());
+}
+
+void DcSctpSocket::SetStreamPriority(StreamID stream_id,
+                                     StreamPriority priority) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  send_queue_.SetStreamPriority(stream_id, priority);
+}
+StreamPriority DcSctpSocket::GetStreamPriority(StreamID stream_id) const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return send_queue_.GetStreamPriority(stream_id);
+}
+
+SendStatus DcSctpSocket::Send(DcSctpMessage message,
+                              const SendOptions& send_options) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
+  LifecycleId lifecycle_id = send_options.lifecycle_id;
+
+  if (message.payload().empty()) {
+    if (lifecycle_id.IsSet()) {
+      callbacks_.OnLifecycleEnd(lifecycle_id);
+    }
+    callbacks_.OnError(ErrorKind::kProtocolViolation,
+                       "Unable to send empty message");
+    return SendStatus::kErrorMessageEmpty;
+  }
+  if (message.payload().size() > options_.max_message_size) {
+    if (lifecycle_id.IsSet()) {
+      callbacks_.OnLifecycleEnd(lifecycle_id);
+    }
+    callbacks_.OnError(ErrorKind::kProtocolViolation,
+                       "Unable to send too large message");
+    return SendStatus::kErrorMessageTooLarge;
+  }
+  if (state_ == State::kShutdownPending || state_ == State::kShutdownSent ||
+      state_ == State::kShutdownReceived || state_ == State::kShutdownAckSent) {
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "An endpoint should reject any new data request from its upper layer
+    // if it is in the SHUTDOWN-PENDING, SHUTDOWN-SENT, SHUTDOWN-RECEIVED, or
+    // SHUTDOWN-ACK-SENT state."
+    if (lifecycle_id.IsSet()) {
+      callbacks_.OnLifecycleEnd(lifecycle_id);
+    }
+    callbacks_.OnError(ErrorKind::kWrongSequence,
+                       "Unable to send message as the socket is shutting down");
+    return SendStatus::kErrorShuttingDown;
+  }
+  if (send_queue_.IsFull()) {
+    if (lifecycle_id.IsSet()) {
+      callbacks_.OnLifecycleEnd(lifecycle_id);
+    }
+    callbacks_.OnError(ErrorKind::kResourceExhaustion,
+                       "Unable to send message as the send queue is full");
+    return SendStatus::kErrorResourceExhaustion;
+  }
+
+  TimeMs now = callbacks_.TimeMillis();
+  ++metrics_.tx_messages_count;
+  send_queue_.Add(now, std::move(message), send_options);
+  if (tcb_ != nullptr) {
+    tcb_->SendBufferedPackets(now);
+  }
+
+  RTC_DCHECK(IsConsistent());
+  return SendStatus::kSuccess;
+}
+
+ResetStreamsStatus DcSctpSocket::ResetStreams(
+    rtc::ArrayView<const StreamID> outgoing_streams) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
+
+  if (tcb_ == nullptr) {
+    callbacks_.OnError(ErrorKind::kWrongSequence,
+                       "Can't reset streams as the socket is not connected");
+    return ResetStreamsStatus::kNotConnected;
+  }
+  if (!tcb_->capabilities().reconfig) {
+    callbacks_.OnError(ErrorKind::kUnsupportedOperation,
+                       "Can't reset streams as the peer doesn't support it");
+    return ResetStreamsStatus::kNotSupported;
+  }
+
+  tcb_->stream_reset_handler().ResetStreams(outgoing_streams);
+  MaybeSendResetStreamsRequest();
+
+  RTC_DCHECK(IsConsistent());
+  return ResetStreamsStatus::kPerformed;
+}
+
+SocketState DcSctpSocket::state() const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  switch (state_) {
+    case State::kClosed:
+      return SocketState::kClosed;
+    case State::kCookieWait:
+    case State::kCookieEchoed:
+      return SocketState::kConnecting;
+    case State::kEstablished:
+      return SocketState::kConnected;
+    case State::kShutdownPending:
+    case State::kShutdownSent:
+    case State::kShutdownReceived:
+    case State::kShutdownAckSent:
+      return SocketState::kShuttingDown;
+  }
+}
+
+void DcSctpSocket::SetMaxMessageSize(size_t max_message_size) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  options_.max_message_size = max_message_size;
+}
+
+size_t DcSctpSocket::buffered_amount(StreamID stream_id) const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return send_queue_.buffered_amount(stream_id);
+}
+
+size_t DcSctpSocket::buffered_amount_low_threshold(StreamID stream_id) const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return send_queue_.buffered_amount_low_threshold(stream_id);
+}
+
+void DcSctpSocket::SetBufferedAmountLowThreshold(StreamID stream_id,
+                                                 size_t bytes) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  send_queue_.SetBufferedAmountLowThreshold(stream_id, bytes);
+}
+
+absl::optional<Metrics> DcSctpSocket::GetMetrics() const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  if (tcb_ == nullptr) {
+    return absl::nullopt;
+  }
+
+  Metrics metrics = metrics_;
+  metrics.cwnd_bytes = tcb_->cwnd();
+  metrics.srtt_ms = tcb_->current_srtt().value();
+  size_t packet_payload_size =
+      options_.mtu - SctpPacket::kHeaderSize - DataChunk::kHeaderSize;
+  metrics.unack_data_count =
+      tcb_->retransmission_queue().outstanding_items() +
+      (send_queue_.total_buffered_amount() + packet_payload_size - 1) /
+          packet_payload_size;
+  metrics.peer_rwnd_bytes = tcb_->retransmission_queue().rwnd();
+  metrics.negotiated_maximum_incoming_streams =
+      tcb_->capabilities().negotiated_maximum_incoming_streams;
+  metrics.negotiated_maximum_incoming_streams =
+      tcb_->capabilities().negotiated_maximum_incoming_streams;
+
+  return metrics;
+}
+
+void DcSctpSocket::MaybeSendShutdownOnPacketReceived(const SctpPacket& packet) {
+  if (state_ == State::kShutdownSent) {
+    bool has_data_chunk =
+        std::find_if(packet.descriptors().begin(), packet.descriptors().end(),
+                     [](const SctpPacket::ChunkDescriptor& descriptor) {
+                       return descriptor.type == DataChunk::kType;
+                     }) != packet.descriptors().end();
+    if (has_data_chunk) {
+      // https://tools.ietf.org/html/rfc4960#section-9.2
+      // "While in the SHUTDOWN-SENT state, the SHUTDOWN sender MUST immediately
+      // respond to each received packet containing one or more DATA chunks with
+      // a SHUTDOWN chunk and restart the T2-shutdown timer.""
+      SendShutdown();
+      t2_shutdown_->set_duration(tcb_->current_rto());
+      t2_shutdown_->Start();
+    }
+  }
+}
+
+void DcSctpSocket::MaybeSendResetStreamsRequest() {
+  absl::optional<ReConfigChunk> reconfig =
+      tcb_->stream_reset_handler().MakeStreamResetRequest();
+  if (reconfig.has_value()) {
+    SctpPacket::Builder builder = tcb_->PacketBuilder();
+    builder.Add(*reconfig);
+    packet_sender_.Send(builder);
+  }
+}
+
+bool DcSctpSocket::ValidatePacket(const SctpPacket& packet) {
+  const CommonHeader& header = packet.common_header();
+  VerificationTag my_verification_tag =
+      tcb_ != nullptr ? tcb_->my_verification_tag() : VerificationTag(0);
+
+  if (header.verification_tag == VerificationTag(0)) {
+    if (packet.descriptors().size() == 1 &&
+        packet.descriptors()[0].type == InitChunk::kType) {
+      // https://tools.ietf.org/html/rfc4960#section-8.5.1
+      // "When an endpoint receives an SCTP packet with the Verification Tag
+      // set to 0, it should verify that the packet contains only an INIT chunk.
+      // Otherwise, the receiver MUST silently discard the packet.""
+      return true;
+    }
+    callbacks_.OnError(
+        ErrorKind::kParseFailed,
+        "Only a single INIT chunk can be present in packets sent on "
+        "verification_tag = 0");
+    return false;
+  }
+
+  if (packet.descriptors().size() == 1 &&
+      packet.descriptors()[0].type == AbortChunk::kType) {
+    // https://tools.ietf.org/html/rfc4960#section-8.5.1
+    // "The receiver of an ABORT MUST accept the packet if the Verification
+    // Tag field of the packet matches its own tag and the T bit is not set OR
+    // if it is set to its peer's tag and the T bit is set in the Chunk Flags.
+    // Otherwise, the receiver MUST silently discard the packet and take no
+    // further action."
+    bool t_bit = (packet.descriptors()[0].flags & 0x01) != 0;
+    if (t_bit && tcb_ == nullptr) {
+      // Can't verify the tag - assume it's okey.
+      return true;
+    }
+    if ((!t_bit && header.verification_tag == my_verification_tag) ||
+        (t_bit && header.verification_tag == tcb_->peer_verification_tag())) {
+      return true;
+    }
+    callbacks_.OnError(ErrorKind::kParseFailed,
+                       "ABORT chunk verification tag was wrong");
+    return false;
+  }
+
+  if (packet.descriptors()[0].type == InitAckChunk::kType) {
+    if (header.verification_tag == connect_params_.verification_tag) {
+      return true;
+    }
+    callbacks_.OnError(
+        ErrorKind::kParseFailed,
+        rtc::StringFormat(
+            "Packet has invalid verification tag: %08x, expected %08x",
+            *header.verification_tag, *connect_params_.verification_tag));
+    return false;
+  }
+
+  if (packet.descriptors()[0].type == CookieEchoChunk::kType) {
+    // Handled in chunk handler (due to RFC 4960, section 5.2.4).
+    return true;
+  }
+
+  if (packet.descriptors().size() == 1 &&
+      packet.descriptors()[0].type == ShutdownCompleteChunk::kType) {
+    // https://tools.ietf.org/html/rfc4960#section-8.5.1
+    // "The receiver of a SHUTDOWN COMPLETE shall accept the packet if the
+    // Verification Tag field of the packet matches its own tag and the T bit is
+    // not set OR if it is set to its peer's tag and the T bit is set in the
+    // Chunk Flags.  Otherwise, the receiver MUST silently discard the packet
+    // and take no further action."
+    bool t_bit = (packet.descriptors()[0].flags & 0x01) != 0;
+    if (t_bit && tcb_ == nullptr) {
+      // Can't verify the tag - assume it's okey.
+      return true;
+    }
+    if ((!t_bit && header.verification_tag == my_verification_tag) ||
+        (t_bit && header.verification_tag == tcb_->peer_verification_tag())) {
+      return true;
+    }
+    callbacks_.OnError(ErrorKind::kParseFailed,
+                       "SHUTDOWN_COMPLETE chunk verification tag was wrong");
+    return false;
+  }
+
+  // https://tools.ietf.org/html/rfc4960#section-8.5
+  // "When receiving an SCTP packet, the endpoint MUST ensure that the value
+  // in the Verification Tag field of the received SCTP packet matches its own
+  // tag.  If the received Verification Tag value does not match the receiver's
+  // own tag value, the receiver shall silently discard the packet and shall not
+  // process it any further..."
+  if (header.verification_tag == my_verification_tag) {
+    return true;
+  }
+
+  callbacks_.OnError(
+      ErrorKind::kParseFailed,
+      rtc::StringFormat(
+          "Packet has invalid verification tag: %08x, expected %08x",
+          *header.verification_tag, *my_verification_tag));
+  return false;
+}
+
+void DcSctpSocket::HandleTimeout(TimeoutID timeout_id) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
+
+  timer_manager_.HandleTimeout(timeout_id);
+
+  if (tcb_ != nullptr && tcb_->HasTooManyTxErrors()) {
+    // Tearing down the TCB has to be done outside the handlers.
+    CloseConnectionBecauseOfTooManyTransmissionErrors();
+  }
+
+  RTC_DCHECK(IsConsistent());
+}
+
+void DcSctpSocket::ReceivePacket(rtc::ArrayView<const uint8_t> data) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
+
+  ++metrics_.rx_packets_count;
+
+  if (packet_observer_ != nullptr) {
+    packet_observer_->OnReceivedPacket(callbacks_.TimeMillis(), data);
+  }
+
+  absl::optional<SctpPacket> packet =
+      SctpPacket::Parse(data, options_.disable_checksum_verification);
+  if (!packet.has_value()) {
+    // https://tools.ietf.org/html/rfc4960#section-6.8
+    // "The default procedure for handling invalid SCTP packets is to
+    // silently discard them."
+    callbacks_.OnError(ErrorKind::kParseFailed,
+                       "Failed to parse received SCTP packet");
+    RTC_DCHECK(IsConsistent());
+    return;
+  }
+
+  if (RTC_DLOG_IS_ON) {
+    for (const auto& descriptor : packet->descriptors()) {
+      RTC_DLOG(LS_VERBOSE) << log_prefix() << "Received "
+                           << DebugConvertChunkToString(descriptor.data);
+    }
+  }
+
+  if (!ValidatePacket(*packet)) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Packet failed verification tag check - dropping";
+    RTC_DCHECK(IsConsistent());
+    return;
+  }
+
+  MaybeSendShutdownOnPacketReceived(*packet);
+
+  for (const auto& descriptor : packet->descriptors()) {
+    if (!Dispatch(packet->common_header(), descriptor)) {
+      break;
+    }
+  }
+
+  if (tcb_ != nullptr) {
+    tcb_->data_tracker().ObservePacketEnd();
+    tcb_->MaybeSendSack();
+  }
+
+  RTC_DCHECK(IsConsistent());
+}
+
+void DcSctpSocket::DebugPrintOutgoing(rtc::ArrayView<const uint8_t> payload) {
+  auto packet = SctpPacket::Parse(payload);
+  RTC_DCHECK(packet.has_value());
+
+  for (const auto& desc : packet->descriptors()) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix() << "Sent "
+                         << DebugConvertChunkToString(desc.data);
+  }
+}
+
+bool DcSctpSocket::Dispatch(const CommonHeader& header,
+                            const SctpPacket::ChunkDescriptor& descriptor) {
+  switch (descriptor.type) {
+    case DataChunk::kType:
+      HandleData(header, descriptor);
+      break;
+    case InitChunk::kType:
+      HandleInit(header, descriptor);
+      break;
+    case InitAckChunk::kType:
+      HandleInitAck(header, descriptor);
+      break;
+    case SackChunk::kType:
+      HandleSack(header, descriptor);
+      break;
+    case HeartbeatRequestChunk::kType:
+      HandleHeartbeatRequest(header, descriptor);
+      break;
+    case HeartbeatAckChunk::kType:
+      HandleHeartbeatAck(header, descriptor);
+      break;
+    case AbortChunk::kType:
+      HandleAbort(header, descriptor);
+      break;
+    case ErrorChunk::kType:
+      HandleError(header, descriptor);
+      break;
+    case CookieEchoChunk::kType:
+      HandleCookieEcho(header, descriptor);
+      break;
+    case CookieAckChunk::kType:
+      HandleCookieAck(header, descriptor);
+      break;
+    case ShutdownChunk::kType:
+      HandleShutdown(header, descriptor);
+      break;
+    case ShutdownAckChunk::kType:
+      HandleShutdownAck(header, descriptor);
+      break;
+    case ShutdownCompleteChunk::kType:
+      HandleShutdownComplete(header, descriptor);
+      break;
+    case ReConfigChunk::kType:
+      HandleReconfig(header, descriptor);
+      break;
+    case ForwardTsnChunk::kType:
+      HandleForwardTsn(header, descriptor);
+      break;
+    case IDataChunk::kType:
+      HandleIData(header, descriptor);
+      break;
+    case IForwardTsnChunk::kType:
+      HandleIForwardTsn(header, descriptor);
+      break;
+    default:
+      return HandleUnrecognizedChunk(descriptor);
+  }
+  return true;
+}
+
+bool DcSctpSocket::HandleUnrecognizedChunk(
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  bool report_as_error = (descriptor.type & 0x40) != 0;
+  bool continue_processing = (descriptor.type & 0x80) != 0;
+  RTC_DLOG(LS_VERBOSE) << log_prefix() << "Received unknown chunk: "
+                       << static_cast<int>(descriptor.type);
+  if (report_as_error) {
+    rtc::StringBuilder sb;
+    sb << "Received unknown chunk of type: "
+       << static_cast<int>(descriptor.type) << " with report-error bit set";
+    callbacks_.OnError(ErrorKind::kParseFailed, sb.str());
+    RTC_DLOG(LS_VERBOSE)
+        << log_prefix()
+        << "Unknown chunk, with type indicating it should be reported.";
+
+    // https://tools.ietf.org/html/rfc4960#section-3.2
+    // "... report in an ERROR chunk using the 'Unrecognized Chunk Type'
+    // cause."
+    if (tcb_ != nullptr) {
+      // Need TCB - this chunk must be sent with a correct verification tag.
+      packet_sender_.Send(tcb_->PacketBuilder().Add(
+          ErrorChunk(Parameters::Builder()
+                         .Add(UnrecognizedChunkTypeCause(std::vector<uint8_t>(
+                             descriptor.data.begin(), descriptor.data.end())))
+                         .Build())));
+    }
+  }
+  if (!continue_processing) {
+    // https://tools.ietf.org/html/rfc4960#section-3.2
+    // "Stop processing this SCTP packet and discard it, do not process any
+    // further chunks within it."
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Unknown chunk, with type indicating not to "
+                            "process any further chunks";
+  }
+
+  return continue_processing;
+}
+
+absl::optional<DurationMs> DcSctpSocket::OnInitTimerExpiry() {
+  RTC_DLOG(LS_VERBOSE) << log_prefix() << "Timer " << t1_init_->name()
+                       << " has expired: " << t1_init_->expiration_count()
+                       << "/" << t1_init_->options().max_restarts.value_or(-1);
+  RTC_DCHECK(state_ == State::kCookieWait);
+
+  if (t1_init_->is_running()) {
+    SendInit();
+  } else {
+    InternalClose(ErrorKind::kTooManyRetries, "No INIT_ACK received");
+  }
+  RTC_DCHECK(IsConsistent());
+  return absl::nullopt;
+}
+
+absl::optional<DurationMs> DcSctpSocket::OnCookieTimerExpiry() {
+  // https://tools.ietf.org/html/rfc4960#section-4
+  // "If the T1-cookie timer expires, the endpoint MUST retransmit COOKIE
+  // ECHO and restart the T1-cookie timer without changing state.  This MUST
+  // be repeated up to 'Max.Init.Retransmits' times. After that, the endpoint
+  // MUST abort the initialization process and report the error to the SCTP
+  // user."
+  RTC_DLOG(LS_VERBOSE) << log_prefix() << "Timer " << t1_cookie_->name()
+                       << " has expired: " << t1_cookie_->expiration_count()
+                       << "/"
+                       << t1_cookie_->options().max_restarts.value_or(-1);
+
+  RTC_DCHECK(state_ == State::kCookieEchoed);
+
+  if (t1_cookie_->is_running()) {
+    tcb_->SendBufferedPackets(callbacks_.TimeMillis());
+  } else {
+    InternalClose(ErrorKind::kTooManyRetries, "No COOKIE_ACK received");
+  }
+
+  RTC_DCHECK(IsConsistent());
+  return absl::nullopt;
+}
+
+absl::optional<DurationMs> DcSctpSocket::OnShutdownTimerExpiry() {
+  RTC_DLOG(LS_VERBOSE) << log_prefix() << "Timer " << t2_shutdown_->name()
+                       << " has expired: " << t2_shutdown_->expiration_count()
+                       << "/"
+                       << t2_shutdown_->options().max_restarts.value_or(-1);
+
+  if (!t2_shutdown_->is_running()) {
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "An endpoint should limit the number of retransmissions of the SHUTDOWN
+    // chunk to the protocol parameter 'Association.Max.Retrans'. If this
+    // threshold is exceeded, the endpoint should destroy the TCB..."
+
+    packet_sender_.Send(tcb_->PacketBuilder().Add(
+        AbortChunk(true, Parameters::Builder()
+                             .Add(UserInitiatedAbortCause(
+                                 "Too many retransmissions of SHUTDOWN"))
+                             .Build())));
+
+    InternalClose(ErrorKind::kTooManyRetries, "No SHUTDOWN_ACK received");
+    RTC_DCHECK(IsConsistent());
+    return absl::nullopt;
+  }
+
+  // https://tools.ietf.org/html/rfc4960#section-9.2
+  // "If the timer expires, the endpoint must resend the SHUTDOWN with the
+  // updated last sequential TSN received from its peer."
+  SendShutdown();
+  RTC_DCHECK(IsConsistent());
+  return tcb_->current_rto();
+}
+
+void DcSctpSocket::OnSentPacket(rtc::ArrayView<const uint8_t> packet,
+                                SendPacketStatus status) {
+  // The packet observer is invoked even if the packet was failed to be sent, to
+  // indicate an attempt was made.
+  if (packet_observer_ != nullptr) {
+    packet_observer_->OnSentPacket(callbacks_.TimeMillis(), packet);
+  }
+
+  if (status == SendPacketStatus::kSuccess) {
+    if (RTC_DLOG_IS_ON) {
+      DebugPrintOutgoing(packet);
+    }
+
+    // The heartbeat interval timer is restarted for every sent packet, to
+    // fire when the outgoing channel is inactive.
+    if (tcb_ != nullptr) {
+      tcb_->heartbeat_handler().RestartTimer();
+    }
+
+    ++metrics_.tx_packets_count;
+  }
+}
+
+bool DcSctpSocket::ValidateHasTCB() {
+  if (tcb_ != nullptr) {
+    return true;
+  }
+
+  callbacks_.OnError(
+      ErrorKind::kNotConnected,
+      "Received unexpected commands on socket that is not connected");
+  return false;
+}
+
+void DcSctpSocket::ReportFailedToParseChunk(int chunk_type) {
+  rtc::StringBuilder sb;
+  sb << "Failed to parse chunk of type: " << chunk_type;
+  callbacks_.OnError(ErrorKind::kParseFailed, sb.str());
+}
+
+void DcSctpSocket::HandleData(const CommonHeader& header,
+                              const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<DataChunk> chunk = DataChunk::Parse(descriptor.data);
+  if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
+    HandleDataCommon(*chunk);
+  }
+}
+
+void DcSctpSocket::HandleIData(const CommonHeader& header,
+                               const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<IDataChunk> chunk = IDataChunk::Parse(descriptor.data);
+  if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
+    HandleDataCommon(*chunk);
+  }
+}
+
+void DcSctpSocket::HandleDataCommon(AnyDataChunk& chunk) {
+  TSN tsn = chunk.tsn();
+  AnyDataChunk::ImmediateAckFlag immediate_ack = chunk.options().immediate_ack;
+  Data data = std::move(chunk).extract();
+
+  if (data.payload.empty()) {
+    // Empty DATA chunks are illegal.
+    packet_sender_.Send(tcb_->PacketBuilder().Add(
+        ErrorChunk(Parameters::Builder().Add(NoUserDataCause(tsn)).Build())));
+    callbacks_.OnError(ErrorKind::kProtocolViolation,
+                       "Received DATA chunk with no user data");
+    return;
+  }
+
+  RTC_DLOG(LS_VERBOSE) << log_prefix() << "Handle DATA, queue_size="
+                       << tcb_->reassembly_queue().queued_bytes()
+                       << ", water_mark="
+                       << tcb_->reassembly_queue().watermark_bytes()
+                       << ", full=" << tcb_->reassembly_queue().is_full()
+                       << ", above="
+                       << tcb_->reassembly_queue().is_above_watermark();
+
+  if (tcb_->reassembly_queue().is_full()) {
+    // If the reassembly queue is full, there is nothing that can be done. The
+    // specification only allows dropping gap-ack-blocks, and that's not
+    // likely to help as the socket has been trying to fill gaps since the
+    // watermark was reached.
+    packet_sender_.Send(tcb_->PacketBuilder().Add(AbortChunk(
+        true, Parameters::Builder().Add(OutOfResourceErrorCause()).Build())));
+    InternalClose(ErrorKind::kResourceExhaustion,
+                  "Reassembly Queue is exhausted");
+    return;
+  }
+
+  if (tcb_->reassembly_queue().is_above_watermark()) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix() << "Is above high watermark";
+    // If the reassembly queue is above its high watermark, only accept data
+    // chunks that increase its cumulative ack tsn in an attempt to fill gaps
+    // to deliver messages.
+    if (!tcb_->data_tracker().will_increase_cum_ack_tsn(tsn)) {
+      RTC_DLOG(LS_VERBOSE) << log_prefix()
+                           << "Rejected data because of exceeding watermark";
+      tcb_->data_tracker().ForceImmediateSack();
+      return;
+    }
+  }
+
+  if (!tcb_->data_tracker().IsTSNValid(tsn)) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Rejected data because of failing TSN validity";
+    return;
+  }
+
+  if (tcb_->data_tracker().Observe(tsn, immediate_ack)) {
+    tcb_->reassembly_queue().Add(tsn, std::move(data));
+    tcb_->reassembly_queue().MaybeResetStreamsDeferred(
+        tcb_->data_tracker().last_cumulative_acked_tsn());
+    DeliverReassembledMessages();
+  }
+}
+
+void DcSctpSocket::HandleInit(const CommonHeader& header,
+                              const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<InitChunk> chunk = InitChunk::Parse(descriptor.data);
+  if (!ValidateParseSuccess(chunk)) {
+    return;
+  }
+
+  if (chunk->initiate_tag() == VerificationTag(0) ||
+      chunk->nbr_outbound_streams() == 0 || chunk->nbr_inbound_streams() == 0) {
+    // https://tools.ietf.org/html/rfc4960#section-3.3.2
+    // "If the value of the Initiate Tag in a received INIT chunk is found
+    // to be 0, the receiver MUST treat it as an error and close the
+    // association by transmitting an ABORT."
+
+    // "A receiver of an INIT with the OS value set to 0 SHOULD abort the
+    // association."
+
+    // "A receiver of an INIT with the MIS value of 0 SHOULD abort the
+    // association."
+
+    packet_sender_.Send(
+        SctpPacket::Builder(VerificationTag(0), options_)
+            .Add(AbortChunk(
+                /*filled_in_verification_tag=*/false,
+                Parameters::Builder()
+                    .Add(ProtocolViolationCause("INIT malformed"))
+                    .Build())));
+    InternalClose(ErrorKind::kProtocolViolation, "Received invalid INIT");
+    return;
+  }
+
+  if (state_ == State::kShutdownAckSent) {
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "If an endpoint is in the SHUTDOWN-ACK-SENT state and receives an
+    // INIT chunk (e.g., if the SHUTDOWN COMPLETE was lost) with source and
+    // destination transport addresses (either in the IP addresses or in the
+    // INIT chunk) that belong to this association, it should discard the INIT
+    // chunk and retransmit the SHUTDOWN ACK chunk."
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Received Init indicating lost ShutdownComplete";
+    SendShutdownAck();
+    return;
+  }
+
+  TieTag tie_tag(0);
+  if (state_ == State::kClosed) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Received Init in closed state (normal)";
+
+    MakeConnectionParameters();
+  } else if (state_ == State::kCookieWait || state_ == State::kCookieEchoed) {
+    // https://tools.ietf.org/html/rfc4960#section-5.2.1
+    // "This usually indicates an initialization collision, i.e., each
+    // endpoint is attempting, at about the same time, to establish an
+    // association with the other endpoint. Upon receipt of an INIT in the
+    // COOKIE-WAIT state, an endpoint MUST respond with an INIT ACK using the
+    // same parameters it sent in its original INIT chunk (including its
+    // Initiate Tag, unchanged).  When responding, the endpoint MUST send the
+    // INIT ACK back to the same address that the original INIT (sent by this
+    // endpoint) was sent."
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Received Init indicating simultaneous connections";
+  } else {
+    RTC_DCHECK(tcb_ != nullptr);
+    // https://tools.ietf.org/html/rfc4960#section-5.2.2
+    // "The outbound SCTP packet containing this INIT ACK MUST carry a
+    // Verification Tag value equal to the Initiate Tag found in the
+    // unexpected INIT.  And the INIT ACK MUST contain a new Initiate Tag
+    // (randomly generated; see Section 5.3.1).  Other parameters for the
+    // endpoint SHOULD be copied from the existing parameters of the
+    // association (e.g., number of outbound streams) into the INIT ACK and
+    // cookie."
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Received Init indicating restarted connection";
+    // Create a new verification tag - different from the previous one.
+    for (int tries = 0; tries < 10; ++tries) {
+      connect_params_.verification_tag = VerificationTag(
+          callbacks_.GetRandomInt(kMinVerificationTag, kMaxVerificationTag));
+      if (connect_params_.verification_tag != tcb_->my_verification_tag()) {
+        break;
+      }
+    }
+
+    // Make the initial TSN make a large jump, so that there is no overlap
+    // with the old and new association.
+    connect_params_.initial_tsn =
+        TSN(*tcb_->retransmission_queue().next_tsn() + 1000000);
+    tie_tag = tcb_->tie_tag();
+  }
+
+  RTC_DLOG(LS_VERBOSE)
+      << log_prefix()
+      << rtc::StringFormat(
+             "Proceeding with connection. my_verification_tag=%08x, "
+             "my_initial_tsn=%u, peer_verification_tag=%08x, "
+             "peer_initial_tsn=%u",
+             *connect_params_.verification_tag, *connect_params_.initial_tsn,
+             *chunk->initiate_tag(), *chunk->initial_tsn());
+
+  Capabilities capabilities =
+      ComputeCapabilities(options_, chunk->nbr_outbound_streams(),
+                          chunk->nbr_inbound_streams(), chunk->parameters());
+
+  SctpPacket::Builder b(chunk->initiate_tag(), options_);
+  Parameters::Builder params_builder =
+      Parameters::Builder().Add(StateCookieParameter(
+          StateCookie(chunk->initiate_tag(), chunk->initial_tsn(),
+                      chunk->a_rwnd(), tie_tag, capabilities)
+              .Serialize()));
+  AddCapabilityParameters(options_, params_builder);
+
+  InitAckChunk init_ack(/*initiate_tag=*/connect_params_.verification_tag,
+                        options_.max_receiver_window_buffer_size,
+                        options_.announced_maximum_outgoing_streams,
+                        options_.announced_maximum_incoming_streams,
+                        connect_params_.initial_tsn, params_builder.Build());
+  b.Add(init_ack);
+  packet_sender_.Send(b);
+}
+
+void DcSctpSocket::HandleInitAck(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<InitAckChunk> chunk = InitAckChunk::Parse(descriptor.data);
+  if (!ValidateParseSuccess(chunk)) {
+    return;
+  }
+
+  if (state_ != State::kCookieWait) {
+    // https://tools.ietf.org/html/rfc4960#section-5.2.3
+    // "If an INIT ACK is received by an endpoint in any state other than
+    // the COOKIE-WAIT state, the endpoint should discard the INIT ACK chunk."
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Received INIT_ACK in unexpected state";
+    return;
+  }
+
+  auto cookie = chunk->parameters().get<StateCookieParameter>();
+  if (!cookie.has_value()) {
+    packet_sender_.Send(
+        SctpPacket::Builder(connect_params_.verification_tag, options_)
+            .Add(AbortChunk(
+                /*filled_in_verification_tag=*/false,
+                Parameters::Builder()
+                    .Add(ProtocolViolationCause("INIT-ACK malformed"))
+                    .Build())));
+    InternalClose(ErrorKind::kProtocolViolation,
+                  "InitAck chunk doesn't contain a cookie");
+    return;
+  }
+  Capabilities capabilities =
+      ComputeCapabilities(options_, chunk->nbr_outbound_streams(),
+                          chunk->nbr_inbound_streams(), chunk->parameters());
+  t1_init_->Stop();
+
+  metrics_.peer_implementation = DeterminePeerImplementation(cookie->data());
+
+  // If the connection is re-established (peer restarted, but re-used old
+  // connection), make sure that all message identifiers are reset and any
+  // partly sent message is re-sent in full. The same is true when the socket
+  // is closed and later re-opened, which never happens in WebRTC, but is a
+  // valid operation on the SCTP level. Note that in case of handover, the
+  // send queue is already re-configured, and shouldn't be reset.
+  send_queue_.Reset();
+
+  CreateTransmissionControlBlock(capabilities, connect_params_.verification_tag,
+                                 connect_params_.initial_tsn,
+                                 chunk->initiate_tag(), chunk->initial_tsn(),
+                                 chunk->a_rwnd(), MakeTieTag(callbacks_));
+
+  SetState(State::kCookieEchoed, "INIT_ACK received");
+
+  // The connection isn't fully established just yet.
+  tcb_->SetCookieEchoChunk(CookieEchoChunk(cookie->data()));
+  tcb_->SendBufferedPackets(callbacks_.TimeMillis());
+  t1_cookie_->Start();
+}
+
+void DcSctpSocket::HandleCookieEcho(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<CookieEchoChunk> chunk =
+      CookieEchoChunk::Parse(descriptor.data);
+  if (!ValidateParseSuccess(chunk)) {
+    return;
+  }
+
+  absl::optional<StateCookie> cookie =
+      StateCookie::Deserialize(chunk->cookie());
+  if (!cookie.has_value()) {
+    callbacks_.OnError(ErrorKind::kParseFailed, "Failed to parse state cookie");
+    return;
+  }
+
+  if (tcb_ != nullptr) {
+    if (!HandleCookieEchoWithTCB(header, *cookie)) {
+      return;
+    }
+  } else {
+    if (header.verification_tag != connect_params_.verification_tag) {
+      callbacks_.OnError(
+          ErrorKind::kParseFailed,
+          rtc::StringFormat(
+              "Received CookieEcho with invalid verification tag: %08x, "
+              "expected %08x",
+              *header.verification_tag, *connect_params_.verification_tag));
+      return;
+    }
+  }
+
+  // The init timer can be running on simultaneous connections.
+  t1_init_->Stop();
+  t1_cookie_->Stop();
+  if (state_ != State::kEstablished) {
+    if (tcb_ != nullptr) {
+      tcb_->ClearCookieEchoChunk();
+    }
+    SetState(State::kEstablished, "COOKIE_ECHO received");
+    callbacks_.OnConnected();
+  }
+
+  if (tcb_ == nullptr) {
+    // If the connection is re-established (peer restarted, but re-used old
+    // connection), make sure that all message identifiers are reset and any
+    // partly sent message is re-sent in full. The same is true when the socket
+    // is closed and later re-opened, which never happens in WebRTC, but is a
+    // valid operation on the SCTP level. Note that in case of handover, the
+    // send queue is already re-configured, and shouldn't be reset.
+    send_queue_.Reset();
+
+    CreateTransmissionControlBlock(
+        cookie->capabilities(), connect_params_.verification_tag,
+        connect_params_.initial_tsn, cookie->initiate_tag(),
+        cookie->initial_tsn(), cookie->a_rwnd(), MakeTieTag(callbacks_));
+  }
+
+  SctpPacket::Builder b = tcb_->PacketBuilder();
+  b.Add(CookieAckChunk());
+
+  // https://tools.ietf.org/html/rfc4960#section-5.1
+  // "A COOKIE ACK chunk may be bundled with any pending DATA chunks (and/or
+  // SACK chunks), but the COOKIE ACK chunk MUST be the first chunk in the
+  // packet."
+  tcb_->SendBufferedPackets(b, callbacks_.TimeMillis());
+}
+
+bool DcSctpSocket::HandleCookieEchoWithTCB(const CommonHeader& header,
+                                           const StateCookie& cookie) {
+  RTC_DLOG(LS_VERBOSE) << log_prefix()
+                       << "Handling CookieEchoChunk with TCB. local_tag="
+                       << *tcb_->my_verification_tag()
+                       << ", peer_tag=" << *header.verification_tag
+                       << ", tcb_tag=" << *tcb_->peer_verification_tag()
+                       << ", cookie_tag=" << *cookie.initiate_tag()
+                       << ", local_tie_tag=" << *tcb_->tie_tag()
+                       << ", peer_tie_tag=" << *cookie.tie_tag();
+  // https://tools.ietf.org/html/rfc4960#section-5.2.4
+  // "Handle a COOKIE ECHO when a TCB Exists"
+  if (header.verification_tag != tcb_->my_verification_tag() &&
+      tcb_->peer_verification_tag() != cookie.initiate_tag() &&
+      cookie.tie_tag() == tcb_->tie_tag()) {
+    // "A) In this case, the peer may have restarted."
+    if (state_ == State::kShutdownAckSent) {
+      // "If the endpoint is in the SHUTDOWN-ACK-SENT state and recognizes
+      // that the peer has restarted ...  it MUST NOT set up a new association
+      // but instead resend the SHUTDOWN ACK and send an ERROR chunk with a
+      // "Cookie Received While Shutting Down" error cause to its peer."
+      SctpPacket::Builder b(cookie.initiate_tag(), options_);
+      b.Add(ShutdownAckChunk());
+      b.Add(ErrorChunk(Parameters::Builder()
+                           .Add(CookieReceivedWhileShuttingDownCause())
+                           .Build()));
+      packet_sender_.Send(b);
+      callbacks_.OnError(ErrorKind::kWrongSequence,
+                         "Received COOKIE-ECHO while shutting down");
+      return false;
+    }
+
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Received COOKIE-ECHO indicating a restarted peer";
+
+    tcb_ = nullptr;
+    callbacks_.OnConnectionRestarted();
+  } else if (header.verification_tag == tcb_->my_verification_tag() &&
+             tcb_->peer_verification_tag() != cookie.initiate_tag()) {
+    // TODO(boivie): Handle the peer_tag == 0?
+    // "B) In this case, both sides may be attempting to start an
+    // association at about the same time, but the peer endpoint started its
+    // INIT after responding to the local endpoint's INIT."
+    RTC_DLOG(LS_VERBOSE)
+        << log_prefix()
+        << "Received COOKIE-ECHO indicating simultaneous connections";
+    tcb_ = nullptr;
+  } else if (header.verification_tag != tcb_->my_verification_tag() &&
+             tcb_->peer_verification_tag() == cookie.initiate_tag() &&
+             cookie.tie_tag() == TieTag(0)) {
+    // "C) In this case, the local endpoint's cookie has arrived late.
+    // Before it arrived, the local endpoint sent an INIT and received an
+    // INIT ACK and finally sent a COOKIE ECHO with the peer's same tag but
+    // a new tag of its own. The cookie should be silently discarded. The
+    // endpoint SHOULD NOT change states and should leave any timers
+    // running."
+    RTC_DLOG(LS_VERBOSE)
+        << log_prefix()
+        << "Received COOKIE-ECHO indicating a late COOKIE-ECHO. Discarding";
+    return false;
+  } else if (header.verification_tag == tcb_->my_verification_tag() &&
+             tcb_->peer_verification_tag() == cookie.initiate_tag()) {
+    // "D) When both local and remote tags match, the endpoint should enter
+    // the ESTABLISHED state, if it is in the COOKIE-ECHOED state.  It
+    // should stop any cookie timer that may be running and send a COOKIE
+    // ACK."
+    RTC_DLOG(LS_VERBOSE)
+        << log_prefix()
+        << "Received duplicate COOKIE-ECHO, probably because of peer not "
+           "receiving COOKIE-ACK and retransmitting COOKIE-ECHO. Continuing.";
+  }
+  return true;
+}
+
+void DcSctpSocket::HandleCookieAck(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<CookieAckChunk> chunk = CookieAckChunk::Parse(descriptor.data);
+  if (!ValidateParseSuccess(chunk)) {
+    return;
+  }
+
+  if (state_ != State::kCookieEchoed) {
+    // https://tools.ietf.org/html/rfc4960#section-5.2.5
+    // "At any state other than COOKIE-ECHOED, an endpoint should silently
+    // discard a received COOKIE ACK chunk."
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Received COOKIE_ACK not in COOKIE_ECHOED state";
+    return;
+  }
+
+  // RFC 4960, Errata ID: 4400
+  t1_cookie_->Stop();
+  tcb_->ClearCookieEchoChunk();
+  SetState(State::kEstablished, "COOKIE_ACK received");
+  tcb_->SendBufferedPackets(callbacks_.TimeMillis());
+  callbacks_.OnConnected();
+}
+
+void DcSctpSocket::DeliverReassembledMessages() {
+  if (tcb_->reassembly_queue().HasMessages()) {
+    for (auto& message : tcb_->reassembly_queue().FlushMessages()) {
+      ++metrics_.rx_messages_count;
+      callbacks_.OnMessageReceived(std::move(message));
+    }
+  }
+}
+
+void DcSctpSocket::HandleSack(const CommonHeader& header,
+                              const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<SackChunk> chunk = SackChunk::Parse(descriptor.data);
+
+  if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
+    TimeMs now = callbacks_.TimeMillis();
+    SackChunk sack = ChunkValidators::Clean(*std::move(chunk));
+
+    if (tcb_->retransmission_queue().HandleSack(now, sack)) {
+      MaybeSendShutdownOrAck();
+      // Receiving an ACK may make the socket go into fast recovery mode.
+      // https://datatracker.ietf.org/doc/html/rfc4960#section-7.2.4
+      // "Determine how many of the earliest (i.e., lowest TSN) DATA chunks
+      // marked for retransmission will fit into a single packet, subject to
+      // constraint of the path MTU of the destination transport address to
+      // which the packet is being sent.  Call this value K. Retransmit those K
+      // DATA chunks in a single packet.  When a Fast Retransmit is being
+      // performed, the sender SHOULD ignore the value of cwnd and SHOULD NOT
+      // delay retransmission for this single packet."
+      tcb_->MaybeSendFastRetransmit();
+
+      // Receiving an ACK will decrease outstanding bytes (maybe now below
+      // cwnd?) or indicate packet loss that may result in sending FORWARD-TSN.
+      tcb_->SendBufferedPackets(now);
+    } else {
+      RTC_DLOG(LS_VERBOSE) << log_prefix()
+                           << "Dropping out-of-order SACK with TSN "
+                           << *sack.cumulative_tsn_ack();
+    }
+  }
+}
+
+void DcSctpSocket::HandleHeartbeatRequest(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<HeartbeatRequestChunk> chunk =
+      HeartbeatRequestChunk::Parse(descriptor.data);
+
+  if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
+    tcb_->heartbeat_handler().HandleHeartbeatRequest(*std::move(chunk));
+  }
+}
+
+void DcSctpSocket::HandleHeartbeatAck(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<HeartbeatAckChunk> chunk =
+      HeartbeatAckChunk::Parse(descriptor.data);
+
+  if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
+    tcb_->heartbeat_handler().HandleHeartbeatAck(*std::move(chunk));
+  }
+}
+
+void DcSctpSocket::HandleAbort(const CommonHeader& header,
+                               const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<AbortChunk> chunk = AbortChunk::Parse(descriptor.data);
+  if (ValidateParseSuccess(chunk)) {
+    std::string error_string = ErrorCausesToString(chunk->error_causes());
+    if (tcb_ == nullptr) {
+      // https://tools.ietf.org/html/rfc4960#section-3.3.7
+      // "If an endpoint receives an ABORT with a format error or no TCB is
+      // found, it MUST silently discard it."
+      RTC_DLOG(LS_VERBOSE) << log_prefix() << "Received ABORT (" << error_string
+                           << ") on a connection with no TCB. Ignoring";
+      return;
+    }
+
+    RTC_DLOG(LS_WARNING) << log_prefix() << "Received ABORT (" << error_string
+                         << ") - closing connection.";
+    InternalClose(ErrorKind::kPeerReported, error_string);
+  }
+}
+
+void DcSctpSocket::HandleError(const CommonHeader& header,
+                               const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<ErrorChunk> chunk = ErrorChunk::Parse(descriptor.data);
+  if (ValidateParseSuccess(chunk)) {
+    std::string error_string = ErrorCausesToString(chunk->error_causes());
+    if (tcb_ == nullptr) {
+      RTC_DLOG(LS_VERBOSE) << log_prefix() << "Received ERROR (" << error_string
+                           << ") on a connection with no TCB. Ignoring";
+      return;
+    }
+
+    RTC_DLOG(LS_WARNING) << log_prefix() << "Received ERROR: " << error_string;
+    callbacks_.OnError(ErrorKind::kPeerReported,
+                       "Peer reported error: " + error_string);
+  }
+}
+
+void DcSctpSocket::HandleReconfig(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  TimeMs now = callbacks_.TimeMillis();
+  absl::optional<ReConfigChunk> chunk = ReConfigChunk::Parse(descriptor.data);
+  if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
+    tcb_->stream_reset_handler().HandleReConfig(*std::move(chunk));
+    // Handling this response may result in outgoing stream resets finishing
+    // (either successfully or with failure). If there still are pending streams
+    // that were waiting for this request to finish, continue resetting them.
+    MaybeSendResetStreamsRequest();
+
+    // If a response was processed, pending to-be-reset streams may now have
+    // become unpaused. Try to send more DATA chunks.
+    tcb_->SendBufferedPackets(now);
+  }
+}
+
+void DcSctpSocket::HandleShutdown(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  if (!ValidateParseSuccess(ShutdownChunk::Parse(descriptor.data))) {
+    return;
+  }
+
+  if (state_ == State::kClosed) {
+    return;
+  } else if (state_ == State::kCookieWait || state_ == State::kCookieEchoed) {
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "If a SHUTDOWN is received in the COOKIE-WAIT or COOKIE ECHOED state,
+    // the SHUTDOWN chunk SHOULD be silently discarded."
+  } else if (state_ == State::kShutdownSent) {
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "If an endpoint is in the SHUTDOWN-SENT state and receives a
+    // SHUTDOWN chunk from its peer, the endpoint shall respond immediately
+    // with a SHUTDOWN ACK to its peer, and move into the SHUTDOWN-ACK-SENT
+    // state restarting its T2-shutdown timer."
+    SendShutdownAck();
+    SetState(State::kShutdownAckSent, "SHUTDOWN received");
+  } else if (state_ == State::kShutdownAckSent) {
+    // TODO(webrtc:12739): This condition should be removed and handled by the
+    // next (state_ != State::kShutdownReceived).
+    return;
+  } else if (state_ != State::kShutdownReceived) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix()
+                         << "Received SHUTDOWN - shutting down the socket";
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "Upon reception of the SHUTDOWN, the peer endpoint shall enter the
+    // SHUTDOWN-RECEIVED state, stop accepting new data from its SCTP user,
+    // and verify, by checking the Cumulative TSN Ack field of the chunk, that
+    // all its outstanding DATA chunks have been received by the SHUTDOWN
+    // sender."
+    SetState(State::kShutdownReceived, "SHUTDOWN received");
+    MaybeSendShutdownOrAck();
+  }
+}
+
+void DcSctpSocket::HandleShutdownAck(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  if (!ValidateParseSuccess(ShutdownAckChunk::Parse(descriptor.data))) {
+    return;
+  }
+
+  if (state_ == State::kShutdownSent || state_ == State::kShutdownAckSent) {
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "Upon the receipt of the SHUTDOWN ACK, the SHUTDOWN sender shall stop
+    // the T2-shutdown timer, send a SHUTDOWN COMPLETE chunk to its peer, and
+    // remove all record of the association."
+
+    // "If an endpoint is in the SHUTDOWN-ACK-SENT state and receives a
+    // SHUTDOWN ACK, it shall stop the T2-shutdown timer, send a SHUTDOWN
+    // COMPLETE chunk to its peer, and remove all record of the association."
+
+    SctpPacket::Builder b = tcb_->PacketBuilder();
+    b.Add(ShutdownCompleteChunk(/*tag_reflected=*/false));
+    packet_sender_.Send(b);
+    InternalClose(ErrorKind::kNoError, "");
+  } else {
+    // https://tools.ietf.org/html/rfc4960#section-8.5.1
+    // "If the receiver is in COOKIE-ECHOED or COOKIE-WAIT state
+    // the procedures in Section 8.4 SHOULD be followed; in other words, it
+    // should be treated as an Out Of The Blue packet."
+
+    // https://tools.ietf.org/html/rfc4960#section-8.4
+    // "If the packet contains a SHUTDOWN ACK chunk, the receiver
+    // should respond to the sender of the OOTB packet with a SHUTDOWN
+    // COMPLETE. When sending the SHUTDOWN COMPLETE, the receiver of the OOTB
+    // packet must fill in the Verification Tag field of the outbound packet
+    // with the Verification Tag received in the SHUTDOWN ACK and set the T
+    // bit in the Chunk Flags to indicate that the Verification Tag is
+    // reflected."
+
+    SctpPacket::Builder b(header.verification_tag, options_);
+    b.Add(ShutdownCompleteChunk(/*tag_reflected=*/true));
+    packet_sender_.Send(b);
+  }
+}
+
+void DcSctpSocket::HandleShutdownComplete(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  if (!ValidateParseSuccess(ShutdownCompleteChunk::Parse(descriptor.data))) {
+    return;
+  }
+
+  if (state_ == State::kShutdownAckSent) {
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "Upon reception of the SHUTDOWN COMPLETE chunk, the endpoint will
+    // verify that it is in the SHUTDOWN-ACK-SENT state; if it is not, the
+    // chunk should be discarded.  If the endpoint is in the SHUTDOWN-ACK-SENT
+    // state, the endpoint should stop the T2-shutdown timer and remove all
+    // knowledge of the association (and thus the association enters the
+    // CLOSED state)."
+    InternalClose(ErrorKind::kNoError, "");
+  }
+}
+
+void DcSctpSocket::HandleForwardTsn(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<ForwardTsnChunk> chunk =
+      ForwardTsnChunk::Parse(descriptor.data);
+  if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
+    HandleForwardTsnCommon(*chunk);
+  }
+}
+
+void DcSctpSocket::HandleIForwardTsn(
+    const CommonHeader& header,
+    const SctpPacket::ChunkDescriptor& descriptor) {
+  absl::optional<IForwardTsnChunk> chunk =
+      IForwardTsnChunk::Parse(descriptor.data);
+  if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
+    HandleForwardTsnCommon(*chunk);
+  }
+}
+
+void DcSctpSocket::HandleForwardTsnCommon(const AnyForwardTsnChunk& chunk) {
+  if (!tcb_->capabilities().partial_reliability) {
+    SctpPacket::Builder b = tcb_->PacketBuilder();
+    b.Add(AbortChunk(/*filled_in_verification_tag=*/true,
+                     Parameters::Builder()
+                         .Add(ProtocolViolationCause(
+                             "I-FORWARD-TSN received, but not indicated "
+                             "during connection establishment"))
+                         .Build()));
+    packet_sender_.Send(b);
+
+    callbacks_.OnError(ErrorKind::kProtocolViolation,
+                       "Received a FORWARD_TSN without announced peer support");
+    return;
+  }
+  tcb_->data_tracker().HandleForwardTsn(chunk.new_cumulative_tsn());
+  tcb_->reassembly_queue().Handle(chunk);
+  // A forward TSN - for ordered streams - may allow messages to be
+  // delivered.
+  DeliverReassembledMessages();
+
+  // Processing a FORWARD_TSN might result in sending a SACK.
+  tcb_->MaybeSendSack();
+}
+
+void DcSctpSocket::MaybeSendShutdownOrAck() {
+  if (tcb_->retransmission_queue().outstanding_bytes() != 0) {
+    return;
+  }
+
+  if (state_ == State::kShutdownPending) {
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "Once all its outstanding data has been acknowledged, the endpoint
+    // shall send a SHUTDOWN chunk to its peer including in the Cumulative TSN
+    // Ack field the last sequential TSN it has received from the peer. It
+    // shall then start the T2-shutdown timer and enter the SHUTDOWN-SENT
+    // state.""
+
+    SendShutdown();
+    t2_shutdown_->set_duration(tcb_->current_rto());
+    t2_shutdown_->Start();
+    SetState(State::kShutdownSent, "No more outstanding data");
+  } else if (state_ == State::kShutdownReceived) {
+    // https://tools.ietf.org/html/rfc4960#section-9.2
+    // "If the receiver of the SHUTDOWN has no more outstanding DATA
+    // chunks, the SHUTDOWN receiver MUST send a SHUTDOWN ACK and start a
+    // T2-shutdown timer of its own, entering the SHUTDOWN-ACK-SENT state.  If
+    // the timer expires, the endpoint must resend the SHUTDOWN ACK."
+
+    SendShutdownAck();
+    SetState(State::kShutdownAckSent, "No more outstanding data");
+  }
+}
+
+void DcSctpSocket::SendShutdown() {
+  SctpPacket::Builder b = tcb_->PacketBuilder();
+  b.Add(ShutdownChunk(tcb_->data_tracker().last_cumulative_acked_tsn()));
+  packet_sender_.Send(b);
+}
+
+void DcSctpSocket::SendShutdownAck() {
+  packet_sender_.Send(tcb_->PacketBuilder().Add(ShutdownAckChunk()));
+  t2_shutdown_->set_duration(tcb_->current_rto());
+  t2_shutdown_->Start();
+}
+
+HandoverReadinessStatus DcSctpSocket::GetHandoverReadiness() const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  HandoverReadinessStatus status;
+  if (state_ != State::kClosed && state_ != State::kEstablished) {
+    status.Add(HandoverUnreadinessReason::kWrongConnectionState);
+  }
+  status.Add(send_queue_.GetHandoverReadiness());
+  if (tcb_) {
+    status.Add(tcb_->GetHandoverReadiness());
+  }
+  return status;
+}
+
+absl::optional<DcSctpSocketHandoverState>
+DcSctpSocket::GetHandoverStateAndClose() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
+
+  if (!GetHandoverReadiness().IsReady()) {
+    return absl::nullopt;
+  }
+
+  DcSctpSocketHandoverState state;
+
+  if (state_ == State::kClosed) {
+    state.socket_state = DcSctpSocketHandoverState::SocketState::kClosed;
+  } else if (state_ == State::kEstablished) {
+    state.socket_state = DcSctpSocketHandoverState::SocketState::kConnected;
+    tcb_->AddHandoverState(state);
+    send_queue_.AddHandoverState(state);
+    InternalClose(ErrorKind::kNoError, "handover");
+  }
+
+  return std::move(state);
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket.h b/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket.h
new file mode 100644
index 0000000000..157c515d65
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket.h
@@ -0,0 +1,298 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_DCSCTP_SOCKET_H_
+#define NET_DCSCTP_SOCKET_DCSCTP_SOCKET_H_
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/sequence_checker.h"
+#include "net/dcsctp/packet/chunk/abort_chunk.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/data_common.h"
+#include "net/dcsctp/packet/chunk/error_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/init_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/init_chunk.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_complete_chunk.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/packet_observer.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/callback_deferrer.h"
+#include "net/dcsctp/socket/packet_sender.h"
+#include "net/dcsctp/socket/state_cookie.h"
+#include "net/dcsctp/socket/transmission_control_block.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_error_counter.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "net/dcsctp/tx/retransmission_timeout.h"
+#include "net/dcsctp/tx/rr_send_queue.h"
+
+namespace dcsctp {
+
+// DcSctpSocket represents a single SCTP socket, to be used over DTLS.
+//
+// Every dcSCTP is completely isolated from any other socket.
+//
+// This class manages all packet and chunk dispatching and mainly handles the
+// connection sequences (connect, close, shutdown, etc) as well as managing
+// the Transmission Control Block (tcb).
+//
+// This class is thread-compatible.
+class DcSctpSocket : public DcSctpSocketInterface {
+ public:
+  // Instantiates a DcSctpSocket, which interacts with the world through the
+  // `callbacks` interface and is configured using `options`.
+  //
+  // For debugging, `log_prefix` will prefix all debug logs, and a
+  // `packet_observer` can be attached to e.g. dump sent and received packets.
+  DcSctpSocket(absl::string_view log_prefix,
+               DcSctpSocketCallbacks& callbacks,
+               std::unique_ptr<PacketObserver> packet_observer,
+               const DcSctpOptions& options);
+
+  DcSctpSocket(const DcSctpSocket&) = delete;
+  DcSctpSocket& operator=(const DcSctpSocket&) = delete;
+
+  // Implementation of `DcSctpSocketInterface`.
+  void ReceivePacket(rtc::ArrayView<const uint8_t> data) override;
+  void HandleTimeout(TimeoutID timeout_id) override;
+  void Connect() override;
+  void RestoreFromState(const DcSctpSocketHandoverState& state) override;
+  void Shutdown() override;
+  void Close() override;
+  SendStatus Send(DcSctpMessage message,
+                  const SendOptions& send_options) override;
+  ResetStreamsStatus ResetStreams(
+      rtc::ArrayView<const StreamID> outgoing_streams) override;
+  SocketState state() const override;
+  const DcSctpOptions& options() const override { return options_; }
+  void SetMaxMessageSize(size_t max_message_size) override;
+  void SetStreamPriority(StreamID stream_id, StreamPriority priority) override;
+  StreamPriority GetStreamPriority(StreamID stream_id) const override;
+  size_t buffered_amount(StreamID stream_id) const override;
+  size_t buffered_amount_low_threshold(StreamID stream_id) const override;
+  void SetBufferedAmountLowThreshold(StreamID stream_id, size_t bytes) override;
+  absl::optional<Metrics> GetMetrics() const override;
+  HandoverReadinessStatus GetHandoverReadiness() const override;
+  absl::optional<DcSctpSocketHandoverState> GetHandoverStateAndClose() override;
+  SctpImplementation peer_implementation() const override {
+    return metrics_.peer_implementation;
+  }
+  // Returns this socket's verification tag, or zero if not yet connected.
+  VerificationTag verification_tag() const {
+    return tcb_ != nullptr ? tcb_->my_verification_tag() : VerificationTag(0);
+  }
+
+ private:
+  // Parameter proposals valid during the connect phase.
+  struct ConnectParameters {
+    TSN initial_tsn = TSN(0);
+    VerificationTag verification_tag = VerificationTag(0);
+  };
+
+  // Detailed state (separate from SocketState, which is the public state).
+  enum class State {
+    kClosed,
+    kCookieWait,
+    // TCB valid in these:
+    kCookieEchoed,
+    kEstablished,
+    kShutdownPending,
+    kShutdownSent,
+    kShutdownReceived,
+    kShutdownAckSent,
+  };
+
+  // Returns the log prefix used for debug logging.
+  std::string log_prefix() const;
+
+  bool IsConsistent() const;
+  static constexpr absl::string_view ToString(DcSctpSocket::State state);
+
+  void CreateTransmissionControlBlock(const Capabilities& capabilities,
+                                      VerificationTag my_verification_tag,
+                                      TSN my_initial_tsn,
+                                      VerificationTag peer_verification_tag,
+                                      TSN peer_initial_tsn,
+                                      size_t a_rwnd,
+                                      TieTag tie_tag);
+
+  // Changes the socket state, given a `reason` (for debugging/logging).
+  void SetState(State state, absl::string_view reason);
+  // Fills in `connect_params` with random verification tag and initial TSN.
+  void MakeConnectionParameters();
+  // Closes the association. Note that the TCB will not be valid past this call.
+  void InternalClose(ErrorKind error, absl::string_view message);
+  // Closes the association, because of too many retransmission errors.
+  void CloseConnectionBecauseOfTooManyTransmissionErrors();
+  // Timer expiration handlers
+  absl::optional<DurationMs> OnInitTimerExpiry();
+  absl::optional<DurationMs> OnCookieTimerExpiry();
+  absl::optional<DurationMs> OnShutdownTimerExpiry();
+  void OnSentPacket(rtc::ArrayView<const uint8_t> packet,
+                    SendPacketStatus status);
+  // Sends SHUTDOWN or SHUTDOWN-ACK if the socket is shutting down and if all
+  // outstanding data has been acknowledged.
+  void MaybeSendShutdownOrAck();
+  // If the socket is shutting down, responds SHUTDOWN to any incoming DATA.
+  void MaybeSendShutdownOnPacketReceived(const SctpPacket& packet);
+  // If there are streams pending to be reset, send a request to reset them.
+  void MaybeSendResetStreamsRequest();
+  // Sends a INIT chunk.
+  void SendInit();
+  // Sends a SHUTDOWN chunk.
+  void SendShutdown();
+  // Sends a SHUTDOWN-ACK chunk.
+  void SendShutdownAck();
+  // Validates the SCTP packet, as a whole - not the validity of individual
+  // chunks within it, as that's done in the different chunk handlers.
+  bool ValidatePacket(const SctpPacket& packet);
+  // Parses `payload`, which is a serialized packet that is just going to be
+  // sent and prints all chunks.
+  void DebugPrintOutgoing(rtc::ArrayView<const uint8_t> payload);
+  // Called whenever there may be reassembled messages, and delivers those.
+  void DeliverReassembledMessages();
+  // Returns true if there is a TCB, and false otherwise (and reports an error).
+  bool ValidateHasTCB();
+
+  // Returns true if the parsing of a chunk of type `T` succeeded. If it didn't,
+  // it reports an error and returns false.
+  template <class T>
+  bool ValidateParseSuccess(const absl::optional<T>& c) {
+    if (c.has_value()) {
+      return true;
+    }
+
+    ReportFailedToParseChunk(T::kType);
+    return false;
+  }
+
+  // Reports failing to have parsed a chunk with the provided `chunk_type`.
+  void ReportFailedToParseChunk(int chunk_type);
+  // Called when unknown chunks are received. May report an error.
+  bool HandleUnrecognizedChunk(const SctpPacket::ChunkDescriptor& descriptor);
+
+  // Will dispatch more specific chunk handlers.
+  bool Dispatch(const CommonHeader& header,
+                const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming DATA chunks.
+  void HandleData(const CommonHeader& header,
+                  const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming I-DATA chunks.
+  void HandleIData(const CommonHeader& header,
+                   const SctpPacket::ChunkDescriptor& descriptor);
+  // Common handler for DATA and I-DATA chunks.
+  void HandleDataCommon(AnyDataChunk& chunk);
+  // Handles incoming INIT chunks.
+  void HandleInit(const CommonHeader& header,
+                  const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming INIT-ACK chunks.
+  void HandleInitAck(const CommonHeader& header,
+                     const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming SACK chunks.
+  void HandleSack(const CommonHeader& header,
+                  const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming HEARTBEAT chunks.
+  void HandleHeartbeatRequest(const CommonHeader& header,
+                              const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming HEARTBEAT-ACK chunks.
+  void HandleHeartbeatAck(const CommonHeader& header,
+                          const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming ABORT chunks.
+  void HandleAbort(const CommonHeader& header,
+                   const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming ERROR chunks.
+  void HandleError(const CommonHeader& header,
+                   const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming COOKIE-ECHO chunks.
+  void HandleCookieEcho(const CommonHeader& header,
+                        const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles receiving COOKIE-ECHO when there already is a TCB. The return value
+  // indicates if the processing should continue.
+  bool HandleCookieEchoWithTCB(const CommonHeader& header,
+                               const StateCookie& cookie);
+  // Handles incoming COOKIE-ACK chunks.
+  void HandleCookieAck(const CommonHeader& header,
+                       const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming SHUTDOWN chunks.
+  void HandleShutdown(const CommonHeader& header,
+                      const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming SHUTDOWN-ACK chunks.
+  void HandleShutdownAck(const CommonHeader& header,
+                         const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming FORWARD-TSN chunks.
+  void HandleForwardTsn(const CommonHeader& header,
+                        const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming I-FORWARD-TSN chunks.
+  void HandleIForwardTsn(const CommonHeader& header,
+                         const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming RE-CONFIG chunks.
+  void HandleReconfig(const CommonHeader& header,
+                      const SctpPacket::ChunkDescriptor& descriptor);
+  // Common handled for FORWARD-TSN/I-FORWARD-TSN.
+  void HandleForwardTsnCommon(const AnyForwardTsnChunk& chunk);
+  // Handles incoming SHUTDOWN-COMPLETE chunks
+  void HandleShutdownComplete(const CommonHeader& header,
+                              const SctpPacket::ChunkDescriptor& descriptor);
+
+  const std::string log_prefix_;
+  const std::unique_ptr<PacketObserver> packet_observer_;
+  RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
+  Metrics metrics_;
+  DcSctpOptions options_;
+
+  // Enqueues callbacks and dispatches them just before returning to the caller.
+  CallbackDeferrer callbacks_;
+
+  TimerManager timer_manager_;
+  const std::unique_ptr<Timer> t1_init_;
+  const std::unique_ptr<Timer> t1_cookie_;
+  const std::unique_ptr<Timer> t2_shutdown_;
+
+  // Packets that failed to be sent, but should be retried.
+  PacketSender packet_sender_;
+
+  // The actual SendQueue implementation. As data can be sent on a socket before
+  // the connection is established, this component is not in the TCB.
+  RRSendQueue send_queue_;
+
+  // Contains verification tag and initial TSN between having sent the INIT
+  // until the connection is established (there is no TCB at this point).
+  ConnectParameters connect_params_;
+  // The socket state.
+  State state_ = State::kClosed;
+  // If the connection is established, contains a transmission control block.
+  std::unique_ptr<TransmissionControlBlock> tcb_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_DCSCTP_SOCKET_H_
diff --git a/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket_network_test.cc b/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket_network_test.cc
new file mode 100644
index 0000000000..f097bfa095
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket_network_test.cc
@@ -0,0 +1,518 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <cstdint>
+#include <deque>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/test/create_network_emulation_manager.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/units/time_delta.h"
+#include "call/simulated_network.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/socket/dcsctp_socket.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "net/dcsctp/timer/task_queue_timeout.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/strings/string_format.h"
+#include "rtc_base/time_utils.h"
+#include "test/gmock.h"
+
+#if !defined(WEBRTC_ANDROID) && defined(NDEBUG) && \
+    !defined(THREAD_SANITIZER) && !defined(MEMORY_SANITIZER)
+#define DCSCTP_NDEBUG_TEST(t) t
+#else
+// In debug mode, and when MSAN or TSAN sanitizers are enabled, these tests are
+// too expensive to run due to extensive consistency checks that iterate on all
+// outstanding chunks. Same with low-end Android devices, which have
+// difficulties with these tests.
+#define DCSCTP_NDEBUG_TEST(t) DISABLED_##t
+#endif
+
+namespace dcsctp {
+namespace {
+using ::testing::AllOf;
+using ::testing::Ge;
+using ::testing::Le;
+using ::testing::SizeIs;
+
+constexpr StreamID kStreamId(1);
+constexpr PPID kPpid(53);
+constexpr size_t kSmallPayloadSize = 10;
+constexpr size_t kLargePayloadSize = 10000;
+constexpr size_t kHugePayloadSize = 262144;
+constexpr size_t kBufferedAmountLowThreshold = kLargePayloadSize * 2;
+constexpr webrtc::TimeDelta kPrintBandwidthDuration =
+    webrtc::TimeDelta::Seconds(1);
+constexpr webrtc::TimeDelta kBenchmarkRuntime(webrtc::TimeDelta::Seconds(10));
+constexpr webrtc::TimeDelta kAWhile(webrtc::TimeDelta::Seconds(1));
+
+inline int GetUniqueSeed() {
+  static int seed = 0;
+  return ++seed;
+}
+
+DcSctpOptions MakeOptionsForTest() {
+  DcSctpOptions options;
+
+  // Throughput numbers are affected by the MTU. Ensure it's constant.
+  options.mtu = 1200;
+
+  // By disabling the heartbeat interval, there will no timers at all running
+  // when the socket is idle, which makes it easy to just continue the test
+  // until there are no more scheduled tasks. Note that it _will_ run for longer
+  // than necessary as timers aren't cancelled when they are stopped (as that's
+  // not supported), but it's still simulated time and passes quickly.
+  options.heartbeat_interval = DurationMs(0);
+  return options;
+}
+
+// When doing throughput tests, knowing what each actor should do.
+enum class ActorMode {
+  kAtRest,
+  kThroughputSender,
+  kThroughputReceiver,
+  kLimitedRetransmissionSender,
+};
+
+// An abstraction around EmulatedEndpoint, representing a bound socket that
+// will send its packet to a given destination.
+class BoundSocket : public webrtc::EmulatedNetworkReceiverInterface {
+ public:
+  void Bind(webrtc::EmulatedEndpoint* endpoint) {
+    endpoint_ = endpoint;
+    uint16_t port = endpoint->BindReceiver(0, this).value();
+    source_address_ =
+        rtc::SocketAddress(endpoint_->GetPeerLocalAddress(), port);
+  }
+
+  void SetDestination(const BoundSocket& socket) {
+    dest_address_ = socket.source_address_;
+  }
+
+  void SetReceiver(std::function<void(rtc::CopyOnWriteBuffer)> receiver) {
+    receiver_ = std::move(receiver);
+  }
+
+  void SendPacket(rtc::ArrayView<const uint8_t> data) {
+    endpoint_->SendPacket(source_address_, dest_address_,
+                          rtc::CopyOnWriteBuffer(data.data(), data.size()));
+  }
+
+ private:
+  // Implementation of `webrtc::EmulatedNetworkReceiverInterface`.
+  void OnPacketReceived(webrtc::EmulatedIpPacket packet) override {
+    receiver_(std::move(packet.data));
+  }
+
+  std::function<void(rtc::CopyOnWriteBuffer)> receiver_;
+  webrtc::EmulatedEndpoint* endpoint_ = nullptr;
+  rtc::SocketAddress source_address_;
+  rtc::SocketAddress dest_address_;
+};
+
+// Sends at a constant rate but with random packet sizes.
+class SctpActor : public DcSctpSocketCallbacks {
+ public:
+  SctpActor(absl::string_view name,
+            BoundSocket& emulated_socket,
+            const DcSctpOptions& sctp_options)
+      : log_prefix_(std::string(name) + ": "),
+        thread_(rtc::Thread::Current()),
+        emulated_socket_(emulated_socket),
+        timeout_factory_(
+            *thread_,
+            [this]() { return TimeMillis(); },
+            [this](dcsctp::TimeoutID timeout_id) {
+              sctp_socket_.HandleTimeout(timeout_id);
+            }),
+        random_(GetUniqueSeed()),
+        sctp_socket_(name, *this, nullptr, sctp_options),
+        last_bandwidth_printout_(TimeMs(TimeMillis())) {
+    emulated_socket.SetReceiver([this](rtc::CopyOnWriteBuffer buf) {
+      // The receiver will be executed on the NetworkEmulation task queue, but
+      // the dcSCTP socket is owned by `thread_` and is not thread-safe.
+      thread_->PostTask([this, buf] { this->sctp_socket_.ReceivePacket(buf); });
+    });
+  }
+
+  void PrintBandwidth() {
+    TimeMs now = TimeMillis();
+    DurationMs duration = now - last_bandwidth_printout_;
+
+    double bitrate_mbps =
+        static_cast<double>(received_bytes_ * 8) / *duration / 1000;
+    RTC_LOG(LS_INFO) << log_prefix()
+                     << rtc::StringFormat("Received %0.2f Mbps", bitrate_mbps);
+
+    received_bitrate_mbps_.push_back(bitrate_mbps);
+    received_bytes_ = 0;
+    last_bandwidth_printout_ = now;
+    // Print again in a second.
+    if (mode_ == ActorMode::kThroughputReceiver) {
+      thread_->PostDelayedTask(
+          SafeTask(safety_.flag(), [this] { PrintBandwidth(); }),
+          kPrintBandwidthDuration);
+    }
+  }
+
+  void SendPacket(rtc::ArrayView<const uint8_t> data) override {
+    emulated_socket_.SendPacket(data);
+  }
+
+  std::unique_ptr<Timeout> CreateTimeout(
+      webrtc::TaskQueueBase::DelayPrecision precision) override {
+    return timeout_factory_.CreateTimeout(precision);
+  }
+
+  TimeMs TimeMillis() override { return TimeMs(rtc::TimeMillis()); }
+
+  uint32_t GetRandomInt(uint32_t low, uint32_t high) override {
+    return random_.Rand(low, high);
+  }
+
+  void OnMessageReceived(DcSctpMessage message) override {
+    received_bytes_ += message.payload().size();
+    last_received_message_ = std::move(message);
+  }
+
+  void OnError(ErrorKind error, absl::string_view message) override {
+    RTC_LOG(LS_WARNING) << log_prefix() << "Socket error: " << ToString(error)
+                        << "; " << message;
+  }
+
+  void OnAborted(ErrorKind error, absl::string_view message) override {
+    RTC_LOG(LS_ERROR) << log_prefix() << "Socket abort: " << ToString(error)
+                      << "; " << message;
+  }
+
+  void OnConnected() override {}
+
+  void OnClosed() override {}
+
+  void OnConnectionRestarted() override {}
+
+  void OnStreamsResetFailed(rtc::ArrayView<const StreamID> outgoing_streams,
+                            absl::string_view reason) override {}
+
+  void OnStreamsResetPerformed(
+      rtc::ArrayView<const StreamID> outgoing_streams) override {}
+
+  void OnIncomingStreamsReset(
+      rtc::ArrayView<const StreamID> incoming_streams) override {}
+
+  void NotifyOutgoingMessageBufferEmpty() override {}
+
+  void OnBufferedAmountLow(StreamID stream_id) override {
+    if (mode_ == ActorMode::kThroughputSender) {
+      std::vector<uint8_t> payload(kHugePayloadSize);
+      sctp_socket_.Send(DcSctpMessage(kStreamId, kPpid, std::move(payload)),
+                        SendOptions());
+
+    } else if (mode_ == ActorMode::kLimitedRetransmissionSender) {
+      while (sctp_socket_.buffered_amount(kStreamId) <
+             kBufferedAmountLowThreshold * 2) {
+        SendOptions send_options;
+        send_options.max_retransmissions = 0;
+        sctp_socket_.Send(
+            DcSctpMessage(kStreamId, kPpid,
+                          std::vector<uint8_t>(kLargePayloadSize)),
+            send_options);
+
+        send_options.max_retransmissions = absl::nullopt;
+        sctp_socket_.Send(
+            DcSctpMessage(kStreamId, kPpid,
+                          std::vector<uint8_t>(kSmallPayloadSize)),
+            send_options);
+      }
+    }
+  }
+
+  absl::optional<DcSctpMessage> ConsumeReceivedMessage() {
+    if (!last_received_message_.has_value()) {
+      return absl::nullopt;
+    }
+    DcSctpMessage ret = *std::move(last_received_message_);
+    last_received_message_ = absl::nullopt;
+    return ret;
+  }
+
+  DcSctpSocket& sctp_socket() { return sctp_socket_; }
+
+  void SetActorMode(ActorMode mode) {
+    mode_ = mode;
+    if (mode_ == ActorMode::kThroughputSender) {
+      sctp_socket_.SetBufferedAmountLowThreshold(kStreamId,
+                                                 kBufferedAmountLowThreshold);
+      std::vector<uint8_t> payload(kHugePayloadSize);
+      sctp_socket_.Send(DcSctpMessage(kStreamId, kPpid, std::move(payload)),
+                        SendOptions());
+
+    } else if (mode_ == ActorMode::kLimitedRetransmissionSender) {
+      sctp_socket_.SetBufferedAmountLowThreshold(kStreamId,
+                                                 kBufferedAmountLowThreshold);
+      std::vector<uint8_t> payload(kHugePayloadSize);
+      sctp_socket_.Send(DcSctpMessage(kStreamId, kPpid, std::move(payload)),
+                        SendOptions());
+
+    } else if (mode == ActorMode::kThroughputReceiver) {
+      thread_->PostDelayedTask(
+          SafeTask(safety_.flag(), [this] { PrintBandwidth(); }),
+          kPrintBandwidthDuration);
+    }
+  }
+
+  // Returns the average bitrate, stripping the first `remove_first_n` that
+  // represent the time it took to ramp up the congestion control algorithm.
+  double avg_received_bitrate_mbps(size_t remove_first_n = 3) const {
+    std::vector<double> bitrates = received_bitrate_mbps_;
+    bitrates.erase(bitrates.begin(), bitrates.begin() + remove_first_n);
+
+    double sum = 0;
+    for (double bitrate : bitrates) {
+      sum += bitrate;
+    }
+
+    return sum / bitrates.size();
+  }
+
+ private:
+  std::string log_prefix() const {
+    rtc::StringBuilder sb;
+    sb << log_prefix_;
+    sb << rtc::TimeMillis();
+    sb << ": ";
+    return sb.Release();
+  }
+
+  ActorMode mode_ = ActorMode::kAtRest;
+  const std::string log_prefix_;
+  rtc::Thread* thread_;
+  BoundSocket& emulated_socket_;
+  TaskQueueTimeoutFactory timeout_factory_;
+  webrtc::Random random_;
+  DcSctpSocket sctp_socket_;
+  size_t received_bytes_ = 0;
+  absl::optional<DcSctpMessage> last_received_message_;
+  TimeMs last_bandwidth_printout_;
+  // Per-second received bitrates, in Mbps
+  std::vector<double> received_bitrate_mbps_;
+  webrtc::ScopedTaskSafety safety_;
+};
+
+class DcSctpSocketNetworkTest : public testing::Test {
+ protected:
+  DcSctpSocketNetworkTest()
+      : options_(MakeOptionsForTest()),
+        emulation_(webrtc::CreateNetworkEmulationManager(
+            webrtc::TimeMode::kSimulated)) {}
+
+  void MakeNetwork(const webrtc::BuiltInNetworkBehaviorConfig& config) {
+    webrtc::EmulatedEndpoint* endpoint_a =
+        emulation_->CreateEndpoint(webrtc::EmulatedEndpointConfig());
+    webrtc::EmulatedEndpoint* endpoint_z =
+        emulation_->CreateEndpoint(webrtc::EmulatedEndpointConfig());
+
+    webrtc::EmulatedNetworkNode* node1 = emulation_->CreateEmulatedNode(config);
+    webrtc::EmulatedNetworkNode* node2 = emulation_->CreateEmulatedNode(config);
+
+    emulation_->CreateRoute(endpoint_a, {node1}, endpoint_z);
+    emulation_->CreateRoute(endpoint_z, {node2}, endpoint_a);
+
+    emulated_socket_a_.Bind(endpoint_a);
+    emulated_socket_z_.Bind(endpoint_z);
+
+    emulated_socket_a_.SetDestination(emulated_socket_z_);
+    emulated_socket_z_.SetDestination(emulated_socket_a_);
+  }
+
+  void Sleep(webrtc::TimeDelta duration) {
+    // Sleep in one-millisecond increments, to let timers expire when expected.
+    for (int i = 0; i < duration.ms(); ++i) {
+      emulation_->time_controller()->AdvanceTime(webrtc::TimeDelta::Millis(1));
+    }
+  }
+
+  DcSctpOptions options_;
+  std::unique_ptr<webrtc::NetworkEmulationManager> emulation_;
+  BoundSocket emulated_socket_a_;
+  BoundSocket emulated_socket_z_;
+};
+
+TEST_F(DcSctpSocketNetworkTest, CanConnectAndShutdown) {
+  webrtc::BuiltInNetworkBehaviorConfig pipe_config;
+  MakeNetwork(pipe_config);
+
+  SctpActor sender("A", emulated_socket_a_, options_);
+  SctpActor receiver("Z", emulated_socket_z_, options_);
+  EXPECT_THAT(sender.sctp_socket().state(), SocketState::kClosed);
+
+  sender.sctp_socket().Connect();
+  Sleep(kAWhile);
+  EXPECT_THAT(sender.sctp_socket().state(), SocketState::kConnected);
+
+  sender.sctp_socket().Shutdown();
+  Sleep(kAWhile);
+  EXPECT_THAT(sender.sctp_socket().state(), SocketState::kClosed);
+}
+
+TEST_F(DcSctpSocketNetworkTest, CanSendLargeMessage) {
+  webrtc::BuiltInNetworkBehaviorConfig pipe_config;
+  pipe_config.queue_delay_ms = 30;
+  MakeNetwork(pipe_config);
+
+  SctpActor sender("A", emulated_socket_a_, options_);
+  SctpActor receiver("Z", emulated_socket_z_, options_);
+  sender.sctp_socket().Connect();
+
+  constexpr size_t kPayloadSize = 100 * 1024;
+
+  std::vector<uint8_t> payload(kPayloadSize);
+  sender.sctp_socket().Send(DcSctpMessage(kStreamId, kPpid, payload),
+                            SendOptions());
+
+  Sleep(kAWhile);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(DcSctpMessage message,
+                              receiver.ConsumeReceivedMessage());
+
+  EXPECT_THAT(message.payload(), SizeIs(kPayloadSize));
+
+  sender.sctp_socket().Shutdown();
+  Sleep(kAWhile);
+}
+
+TEST_F(DcSctpSocketNetworkTest, CanSendMessagesReliablyWithLowBandwidth) {
+  webrtc::BuiltInNetworkBehaviorConfig pipe_config;
+  pipe_config.queue_delay_ms = 30;
+  pipe_config.link_capacity_kbps = 1000;
+  MakeNetwork(pipe_config);
+
+  SctpActor sender("A", emulated_socket_a_, options_);
+  SctpActor receiver("Z", emulated_socket_z_, options_);
+  sender.sctp_socket().Connect();
+
+  sender.SetActorMode(ActorMode::kThroughputSender);
+  receiver.SetActorMode(ActorMode::kThroughputReceiver);
+
+  Sleep(kBenchmarkRuntime);
+  sender.SetActorMode(ActorMode::kAtRest);
+  receiver.SetActorMode(ActorMode::kAtRest);
+
+  Sleep(kAWhile);
+
+  sender.sctp_socket().Shutdown();
+
+  Sleep(kAWhile);
+
+  // Verify that the bitrates are in the range of 0.5-1.0 Mbps.
+  double bitrate = receiver.avg_received_bitrate_mbps();
+  EXPECT_THAT(bitrate, AllOf(Ge(0.5), Le(1.0)));
+}
+
+TEST_F(DcSctpSocketNetworkTest,
+       DCSCTP_NDEBUG_TEST(CanSendMessagesReliablyWithMediumBandwidth)) {
+  webrtc::BuiltInNetworkBehaviorConfig pipe_config;
+  pipe_config.queue_delay_ms = 30;
+  pipe_config.link_capacity_kbps = 18000;
+  MakeNetwork(pipe_config);
+
+  SctpActor sender("A", emulated_socket_a_, options_);
+  SctpActor receiver("Z", emulated_socket_z_, options_);
+  sender.sctp_socket().Connect();
+
+  sender.SetActorMode(ActorMode::kThroughputSender);
+  receiver.SetActorMode(ActorMode::kThroughputReceiver);
+
+  Sleep(kBenchmarkRuntime);
+  sender.SetActorMode(ActorMode::kAtRest);
+  receiver.SetActorMode(ActorMode::kAtRest);
+
+  Sleep(kAWhile);
+
+  sender.sctp_socket().Shutdown();
+
+  Sleep(kAWhile);
+
+  // Verify that the bitrates are in the range of 16-18 Mbps.
+  double bitrate = receiver.avg_received_bitrate_mbps();
+  EXPECT_THAT(bitrate, AllOf(Ge(16), Le(18)));
+}
+
+TEST_F(DcSctpSocketNetworkTest, CanSendMessagesReliablyWithMuchPacketLoss) {
+  webrtc::BuiltInNetworkBehaviorConfig config;
+  config.queue_delay_ms = 30;
+  config.loss_percent = 1;
+  MakeNetwork(config);
+
+  SctpActor sender("A", emulated_socket_a_, options_);
+  SctpActor receiver("Z", emulated_socket_z_, options_);
+  sender.sctp_socket().Connect();
+
+  sender.SetActorMode(ActorMode::kThroughputSender);
+  receiver.SetActorMode(ActorMode::kThroughputReceiver);
+
+  Sleep(kBenchmarkRuntime);
+  sender.SetActorMode(ActorMode::kAtRest);
+  receiver.SetActorMode(ActorMode::kAtRest);
+
+  Sleep(kAWhile);
+
+  sender.sctp_socket().Shutdown();
+
+  Sleep(kAWhile);
+
+  // TCP calculator gives: 1200 MTU, 60ms RTT and 1% packet loss -> 1.6Mbps.
+  // This test is doing slightly better (doesn't have any additional header
+  // overhead etc). Verify that the bitrates are in the range of 1.5-2.5 Mbps.
+  double bitrate = receiver.avg_received_bitrate_mbps();
+  EXPECT_THAT(bitrate, AllOf(Ge(1.5), Le(2.5)));
+}
+
+TEST_F(DcSctpSocketNetworkTest, DCSCTP_NDEBUG_TEST(HasHighBandwidth)) {
+  webrtc::BuiltInNetworkBehaviorConfig pipe_config;
+  pipe_config.queue_delay_ms = 30;
+  MakeNetwork(pipe_config);
+
+  SctpActor sender("A", emulated_socket_a_, options_);
+  SctpActor receiver("Z", emulated_socket_z_, options_);
+  sender.sctp_socket().Connect();
+
+  sender.SetActorMode(ActorMode::kThroughputSender);
+  receiver.SetActorMode(ActorMode::kThroughputReceiver);
+
+  Sleep(kBenchmarkRuntime);
+
+  sender.SetActorMode(ActorMode::kAtRest);
+  receiver.SetActorMode(ActorMode::kAtRest);
+  Sleep(kAWhile);
+
+  sender.sctp_socket().Shutdown();
+  Sleep(kAWhile);
+
+  // Verify that the bitrate is in the range of 540-640 Mbps
+  double bitrate = receiver.avg_received_bitrate_mbps();
+  EXPECT_THAT(bitrate, AllOf(Ge(520), Le(640)));
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket_test.cc b/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket_test.cc
new file mode 100644
index 0000000000..f38624d606
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/dcsctp_socket_test.cc
@@ -0,0 +1,2853 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/dcsctp_socket.h"
+
+#include <cstdint>
+#include <deque>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/handover_testing.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/data_common.h"
+#include "net/dcsctp/packet/chunk/error_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+#include "net/dcsctp/packet/chunk/init_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_chunk.h"
+#include "net/dcsctp/packet/error_cause/error_cause.h"
+#include "net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h"
+#include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/text_pcap_packet_observer.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/mock_dcsctp_socket_callbacks.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+ABSL_FLAG(bool, dcsctp_capture_packets, false, "Print packet capture.");
+
+namespace dcsctp {
+namespace {
+using ::testing::_;
+using ::testing::AllOf;
+using ::testing::ElementsAre;
+using ::testing::HasSubstr;
+using ::testing::IsEmpty;
+using ::testing::SizeIs;
+using ::testing::UnorderedElementsAre;
+
+constexpr SendOptions kSendOptions;
+constexpr size_t kLargeMessageSize = DcSctpOptions::kMaxSafeMTUSize * 20;
+constexpr size_t kSmallMessageSize = 10;
+constexpr int kMaxBurstPackets = 4;
+
+MATCHER_P(HasDataChunkWithStreamId, stream_id, "") {
+  absl::optional<SctpPacket> packet = SctpPacket::Parse(arg);
+  if (!packet.has_value()) {
+    *result_listener << "data didn't parse as an SctpPacket";
+    return false;
+  }
+
+  if (packet->descriptors()[0].type != DataChunk::kType) {
+    *result_listener << "the first chunk in the packet is not a data chunk";
+    return false;
+  }
+
+  absl::optional<DataChunk> dc =
+      DataChunk::Parse(packet->descriptors()[0].data);
+  if (!dc.has_value()) {
+    *result_listener << "The first chunk didn't parse as a data chunk";
+    return false;
+  }
+
+  if (dc->stream_id() != stream_id) {
+    *result_listener << "the stream_id is " << *dc->stream_id();
+    return false;
+  }
+
+  return true;
+}
+
+MATCHER_P(HasDataChunkWithPPID, ppid, "") {
+  absl::optional<SctpPacket> packet = SctpPacket::Parse(arg);
+  if (!packet.has_value()) {
+    *result_listener << "data didn't parse as an SctpPacket";
+    return false;
+  }
+
+  if (packet->descriptors()[0].type != DataChunk::kType) {
+    *result_listener << "the first chunk in the packet is not a data chunk";
+    return false;
+  }
+
+  absl::optional<DataChunk> dc =
+      DataChunk::Parse(packet->descriptors()[0].data);
+  if (!dc.has_value()) {
+    *result_listener << "The first chunk didn't parse as a data chunk";
+    return false;
+  }
+
+  if (dc->ppid() != ppid) {
+    *result_listener << "the ppid is " << *dc->ppid();
+    return false;
+  }
+
+  return true;
+}
+
+MATCHER_P(HasDataChunkWithSsn, ssn, "") {
+  absl::optional<SctpPacket> packet = SctpPacket::Parse(arg);
+  if (!packet.has_value()) {
+    *result_listener << "data didn't parse as an SctpPacket";
+    return false;
+  }
+
+  if (packet->descriptors()[0].type != DataChunk::kType) {
+    *result_listener << "the first chunk in the packet is not a data chunk";
+    return false;
+  }
+
+  absl::optional<DataChunk> dc =
+      DataChunk::Parse(packet->descriptors()[0].data);
+  if (!dc.has_value()) {
+    *result_listener << "The first chunk didn't parse as a data chunk";
+    return false;
+  }
+
+  if (dc->ssn() != ssn) {
+    *result_listener << "the ssn is " << *dc->ssn();
+    return false;
+  }
+
+  return true;
+}
+
+MATCHER_P(HasDataChunkWithMid, mid, "") {
+  absl::optional<SctpPacket> packet = SctpPacket::Parse(arg);
+  if (!packet.has_value()) {
+    *result_listener << "data didn't parse as an SctpPacket";
+    return false;
+  }
+
+  if (packet->descriptors()[0].type != IDataChunk::kType) {
+    *result_listener << "the first chunk in the packet is not an i-data chunk";
+    return false;
+  }
+
+  absl::optional<IDataChunk> dc =
+      IDataChunk::Parse(packet->descriptors()[0].data);
+  if (!dc.has_value()) {
+    *result_listener << "The first chunk didn't parse as an i-data chunk";
+    return false;
+  }
+
+  if (dc->message_id() != mid) {
+    *result_listener << "the mid is " << *dc->message_id();
+    return false;
+  }
+
+  return true;
+}
+
+MATCHER_P(HasSackWithCumAckTsn, tsn, "") {
+  absl::optional<SctpPacket> packet = SctpPacket::Parse(arg);
+  if (!packet.has_value()) {
+    *result_listener << "data didn't parse as an SctpPacket";
+    return false;
+  }
+
+  if (packet->descriptors()[0].type != SackChunk::kType) {
+    *result_listener << "the first chunk in the packet is not a data chunk";
+    return false;
+  }
+
+  absl::optional<SackChunk> sc =
+      SackChunk::Parse(packet->descriptors()[0].data);
+  if (!sc.has_value()) {
+    *result_listener << "The first chunk didn't parse as a data chunk";
+    return false;
+  }
+
+  if (sc->cumulative_tsn_ack() != tsn) {
+    *result_listener << "the cum_ack_tsn is " << *sc->cumulative_tsn_ack();
+    return false;
+  }
+
+  return true;
+}
+
+MATCHER(HasSackWithNoGapAckBlocks, "") {
+  absl::optional<SctpPacket> packet = SctpPacket::Parse(arg);
+  if (!packet.has_value()) {
+    *result_listener << "data didn't parse as an SctpPacket";
+    return false;
+  }
+
+  if (packet->descriptors()[0].type != SackChunk::kType) {
+    *result_listener << "the first chunk in the packet is not a data chunk";
+    return false;
+  }
+
+  absl::optional<SackChunk> sc =
+      SackChunk::Parse(packet->descriptors()[0].data);
+  if (!sc.has_value()) {
+    *result_listener << "The first chunk didn't parse as a data chunk";
+    return false;
+  }
+
+  if (!sc->gap_ack_blocks().empty()) {
+    *result_listener << "there are gap ack blocks";
+    return false;
+  }
+
+  return true;
+}
+
+MATCHER_P(HasReconfigWithStreams, streams_matcher, "") {
+  absl::optional<SctpPacket> packet = SctpPacket::Parse(arg);
+  if (!packet.has_value()) {
+    *result_listener << "data didn't parse as an SctpPacket";
+    return false;
+  }
+
+  if (packet->descriptors()[0].type != ReConfigChunk::kType) {
+    *result_listener << "the first chunk in the packet is not a data chunk";
+    return false;
+  }
+
+  absl::optional<ReConfigChunk> reconfig =
+      ReConfigChunk::Parse(packet->descriptors()[0].data);
+  if (!reconfig.has_value()) {
+    *result_listener << "The first chunk didn't parse as a data chunk";
+    return false;
+  }
+
+  const Parameters& parameters = reconfig->parameters();
+  if (parameters.descriptors().size() != 1 ||
+      parameters.descriptors()[0].type !=
+          OutgoingSSNResetRequestParameter::kType) {
+    *result_listener << "Expected the reconfig chunk to have an outgoing SSN "
+                        "reset request parameter";
+    return false;
+  }
+
+  absl::optional<OutgoingSSNResetRequestParameter> p =
+      OutgoingSSNResetRequestParameter::Parse(parameters.descriptors()[0].data);
+  testing::Matcher<rtc::ArrayView<const StreamID>> matcher = streams_matcher;
+  if (!matcher.MatchAndExplain(p->stream_ids(), result_listener)) {
+    return false;
+  }
+
+  return true;
+}
+
+MATCHER_P(HasReconfigWithResponse, result, "") {
+  absl::optional<SctpPacket> packet = SctpPacket::Parse(arg);
+  if (!packet.has_value()) {
+    *result_listener << "data didn't parse as an SctpPacket";
+    return false;
+  }
+
+  if (packet->descriptors()[0].type != ReConfigChunk::kType) {
+    *result_listener << "the first chunk in the packet is not a reconfig chunk";
+    return false;
+  }
+
+  absl::optional<ReConfigChunk> reconfig =
+      ReConfigChunk::Parse(packet->descriptors()[0].data);
+  if (!reconfig.has_value()) {
+    *result_listener << "The first chunk didn't parse as a reconfig chunk";
+    return false;
+  }
+
+  const Parameters& parameters = reconfig->parameters();
+  if (parameters.descriptors().size() != 1 ||
+      parameters.descriptors()[0].type !=
+          ReconfigurationResponseParameter::kType) {
+    *result_listener << "Expected the reconfig chunk to have a "
+                        "ReconfigurationResponse Parameter";
+    return false;
+  }
+
+  absl::optional<ReconfigurationResponseParameter> p =
+      ReconfigurationResponseParameter::Parse(parameters.descriptors()[0].data);
+  if (p->result() != result) {
+    *result_listener << "ReconfigurationResponse Parameter doesn't contain the "
+                        "expected result";
+    return false;
+  }
+
+  return true;
+}
+
+TSN AddTo(TSN tsn, int delta) {
+  return TSN(*tsn + delta);
+}
+
+DcSctpOptions FixupOptions(DcSctpOptions options = {}) {
+  DcSctpOptions fixup = options;
+  // To make the interval more predictable in tests.
+  fixup.heartbeat_interval_include_rtt = false;
+  fixup.max_burst = kMaxBurstPackets;
+  return fixup;
+}
+
+std::unique_ptr<PacketObserver> GetPacketObserver(absl::string_view name) {
+  if (absl::GetFlag(FLAGS_dcsctp_capture_packets)) {
+    return std::make_unique<TextPcapPacketObserver>(name);
+  }
+  return nullptr;
+}
+
+struct SocketUnderTest {
+  explicit SocketUnderTest(absl::string_view name,
+                           const DcSctpOptions& opts = {})
+      : options(FixupOptions(opts)),
+        cb(name),
+        socket(name, cb, GetPacketObserver(name), options) {}
+
+  const DcSctpOptions options;
+  testing::NiceMock<MockDcSctpSocketCallbacks> cb;
+  DcSctpSocket socket;
+};
+
+void ExchangeMessages(SocketUnderTest& a, SocketUnderTest& z) {
+  bool delivered_packet = false;
+  do {
+    delivered_packet = false;
+    std::vector<uint8_t> packet_from_a = a.cb.ConsumeSentPacket();
+    if (!packet_from_a.empty()) {
+      delivered_packet = true;
+      z.socket.ReceivePacket(std::move(packet_from_a));
+    }
+    std::vector<uint8_t> packet_from_z = z.cb.ConsumeSentPacket();
+    if (!packet_from_z.empty()) {
+      delivered_packet = true;
+      a.socket.ReceivePacket(std::move(packet_from_z));
+    }
+  } while (delivered_packet);
+}
+
+void RunTimers(SocketUnderTest& s) {
+  for (;;) {
+    absl::optional<TimeoutID> timeout_id = s.cb.GetNextExpiredTimeout();
+    if (!timeout_id.has_value()) {
+      break;
+    }
+    s.socket.HandleTimeout(*timeout_id);
+  }
+}
+
+void AdvanceTime(SocketUnderTest& a, SocketUnderTest& z, DurationMs duration) {
+  a.cb.AdvanceTime(duration);
+  z.cb.AdvanceTime(duration);
+
+  RunTimers(a);
+  RunTimers(z);
+}
+
+// Calls Connect() on `sock_a_` and make the connection established.
+void ConnectSockets(SocketUnderTest& a, SocketUnderTest& z) {
+  EXPECT_CALL(a.cb, OnConnected).Times(1);
+  EXPECT_CALL(z.cb, OnConnected).Times(1);
+
+  a.socket.Connect();
+  // Z reads INIT, INIT_ACK, COOKIE_ECHO, COOKIE_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnected);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+}
+
+std::unique_ptr<SocketUnderTest> HandoverSocket(
+    std::unique_ptr<SocketUnderTest> sut) {
+  EXPECT_EQ(sut->socket.GetHandoverReadiness(), HandoverReadinessStatus());
+
+  bool is_closed = sut->socket.state() == SocketState::kClosed;
+  if (!is_closed) {
+    EXPECT_CALL(sut->cb, OnClosed).Times(1);
+  }
+  absl::optional<DcSctpSocketHandoverState> handover_state =
+      sut->socket.GetHandoverStateAndClose();
+  EXPECT_TRUE(handover_state.has_value());
+  g_handover_state_transformer_for_test(&*handover_state);
+
+  auto handover_socket = std::make_unique<SocketUnderTest>("H", sut->options);
+  if (!is_closed) {
+    EXPECT_CALL(handover_socket->cb, OnConnected).Times(1);
+  }
+  handover_socket->socket.RestoreFromState(*handover_state);
+  return handover_socket;
+}
+
+std::vector<uint32_t> GetReceivedMessagePpids(SocketUnderTest& z) {
+  std::vector<uint32_t> ppids;
+  for (;;) {
+    absl::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
+    if (!msg.has_value()) {
+      break;
+    }
+    ppids.push_back(*msg->ppid());
+  }
+  return ppids;
+}
+
+// Test parameter that controls whether to perform handovers during the test. A
+// test can have multiple points where it conditionally hands over socket Z.
+// Either socket Z will be handed over at all those points or handed over never.
+enum class HandoverMode {
+  kNoHandover,
+  kPerformHandovers,
+};
+
+class DcSctpSocketParametrizedTest
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<HandoverMode> {
+ protected:
+  // Trigger handover for `sut` depending on the current test param.
+  std::unique_ptr<SocketUnderTest> MaybeHandoverSocket(
+      std::unique_ptr<SocketUnderTest> sut) {
+    if (GetParam() == HandoverMode::kPerformHandovers) {
+      return HandoverSocket(std::move(sut));
+    }
+    return sut;
+  }
+
+  // Trigger handover for socket Z depending on the current test param.
+  // Then checks message passing to verify the handed over socket is functional.
+  void MaybeHandoverSocketAndSendMessage(SocketUnderTest& a,
+                                         std::unique_ptr<SocketUnderTest> z) {
+    if (GetParam() == HandoverMode::kPerformHandovers) {
+      z = HandoverSocket(std::move(z));
+    }
+
+    ExchangeMessages(a, *z);
+    a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions);
+    ExchangeMessages(a, *z);
+
+    absl::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
+    ASSERT_TRUE(msg.has_value());
+    EXPECT_EQ(msg->stream_id(), StreamID(1));
+  }
+};
+
+INSTANTIATE_TEST_SUITE_P(Handovers,
+                         DcSctpSocketParametrizedTest,
+                         testing::Values(HandoverMode::kNoHandover,
+                                         HandoverMode::kPerformHandovers),
+                         [](const auto& test_info) {
+                           return test_info.param ==
+                                          HandoverMode::kPerformHandovers
+                                      ? "WithHandovers"
+                                      : "NoHandover";
+                         });
+
+TEST(DcSctpSocketTest, EstablishConnection) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  EXPECT_CALL(a.cb, OnConnected).Times(1);
+  EXPECT_CALL(z.cb, OnConnected).Times(1);
+  EXPECT_CALL(a.cb, OnConnectionRestarted).Times(0);
+  EXPECT_CALL(z.cb, OnConnectionRestarted).Times(0);
+
+  a.socket.Connect();
+  // Z reads INIT, produces INIT_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads INIT_ACK, produces COOKIE_ECHO
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+  // Z reads COOKIE_ECHO, produces COOKIE_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads COOKIE_ACK.
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnected);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+}
+
+TEST(DcSctpSocketTest, EstablishConnectionWithSetupCollision) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  EXPECT_CALL(a.cb, OnConnected).Times(1);
+  EXPECT_CALL(z.cb, OnConnected).Times(1);
+  EXPECT_CALL(a.cb, OnConnectionRestarted).Times(0);
+  EXPECT_CALL(z.cb, OnConnectionRestarted).Times(0);
+  a.socket.Connect();
+  z.socket.Connect();
+
+  ExchangeMessages(a, z);
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnected);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+}
+
+TEST(DcSctpSocketTest, ShuttingDownWhileEstablishingConnection) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  EXPECT_CALL(a.cb, OnConnected).Times(0);
+  EXPECT_CALL(z.cb, OnConnected).Times(1);
+  a.socket.Connect();
+
+  // Z reads INIT, produces INIT_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads INIT_ACK, produces COOKIE_ECHO
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+  // Z reads COOKIE_ECHO, produces COOKIE_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // Drop COOKIE_ACK, just to more easily verify shutdown protocol.
+  z.cb.ConsumeSentPacket();
+
+  // As Socket A has received INIT_ACK, it has a TCB and is connected, while
+  // Socket Z needs to receive COOKIE_ECHO to get there. Socket A still has
+  // timers running at this point.
+  EXPECT_EQ(a.socket.state(), SocketState::kConnecting);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+
+  // Socket A is now shut down, which should make it stop those timers.
+  a.socket.Shutdown();
+
+  EXPECT_CALL(a.cb, OnClosed).Times(1);
+  EXPECT_CALL(z.cb, OnClosed).Times(1);
+
+  // Z reads SHUTDOWN, produces SHUTDOWN_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads SHUTDOWN_ACK, produces SHUTDOWN_COMPLETE
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+  // Z reads SHUTDOWN_COMPLETE.
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  EXPECT_TRUE(a.cb.ConsumeSentPacket().empty());
+  EXPECT_TRUE(z.cb.ConsumeSentPacket().empty());
+
+  EXPECT_EQ(a.socket.state(), SocketState::kClosed);
+  EXPECT_EQ(z.socket.state(), SocketState::kClosed);
+}
+
+TEST(DcSctpSocketTest, EstablishSimultaneousConnection) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  EXPECT_CALL(a.cb, OnConnected).Times(1);
+  EXPECT_CALL(z.cb, OnConnected).Times(1);
+  EXPECT_CALL(a.cb, OnConnectionRestarted).Times(0);
+  EXPECT_CALL(z.cb, OnConnectionRestarted).Times(0);
+  a.socket.Connect();
+
+  // INIT isn't received by Z, as it wasn't ready yet.
+  a.cb.ConsumeSentPacket();
+
+  z.socket.Connect();
+
+  // A reads INIT, produces INIT_ACK
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  // Z reads INIT_ACK, sends COOKIE_ECHO
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  // A reads COOKIE_ECHO - establishes connection.
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnected);
+
+  // Proceed with the remaining packets.
+  ExchangeMessages(a, z);
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnected);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+}
+
+TEST(DcSctpSocketTest, EstablishConnectionLostCookieAck) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  EXPECT_CALL(a.cb, OnConnected).Times(1);
+  EXPECT_CALL(z.cb, OnConnected).Times(1);
+  EXPECT_CALL(a.cb, OnConnectionRestarted).Times(0);
+  EXPECT_CALL(z.cb, OnConnectionRestarted).Times(0);
+
+  a.socket.Connect();
+  // Z reads INIT, produces INIT_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads INIT_ACK, produces COOKIE_ECHO
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+  // Z reads COOKIE_ECHO, produces COOKIE_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // COOKIE_ACK is lost.
+  z.cb.ConsumeSentPacket();
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnecting);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+
+  // This will make A re-send the COOKIE_ECHO
+  AdvanceTime(a, z, DurationMs(a.options.t1_cookie_timeout));
+
+  // Z reads COOKIE_ECHO, produces COOKIE_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads COOKIE_ACK.
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnected);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+}
+
+TEST(DcSctpSocketTest, ResendInitAndEstablishConnection) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  a.socket.Connect();
+  // INIT is never received by Z.
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket init_packet,
+                              SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+  EXPECT_EQ(init_packet.descriptors()[0].type, InitChunk::kType);
+
+  AdvanceTime(a, z, a.options.t1_init_timeout);
+
+  // Z reads INIT, produces INIT_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads INIT_ACK, produces COOKIE_ECHO
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+  // Z reads COOKIE_ECHO, produces COOKIE_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads COOKIE_ACK.
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnected);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+}
+
+TEST(DcSctpSocketTest, ResendingInitTooManyTimesAborts) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  a.socket.Connect();
+
+  // INIT is never received by Z.
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket init_packet,
+                              SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+  EXPECT_EQ(init_packet.descriptors()[0].type, InitChunk::kType);
+
+  for (int i = 0; i < *a.options.max_init_retransmits; ++i) {
+    AdvanceTime(a, z, a.options.t1_init_timeout * (1 << i));
+
+    // INIT is resent
+    ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket resent_init_packet,
+                                SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+    EXPECT_EQ(resent_init_packet.descriptors()[0].type, InitChunk::kType);
+  }
+
+  // Another timeout, after the max init retransmits.
+  EXPECT_CALL(a.cb, OnAborted).Times(1);
+  AdvanceTime(
+      a, z, a.options.t1_init_timeout * (1 << *a.options.max_init_retransmits));
+
+  EXPECT_EQ(a.socket.state(), SocketState::kClosed);
+}
+
+TEST(DcSctpSocketTest, ResendCookieEchoAndEstablishConnection) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  a.socket.Connect();
+
+  // Z reads INIT, produces INIT_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads INIT_ACK, produces COOKIE_ECHO
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  // COOKIE_ECHO is never received by Z.
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket init_packet,
+                              SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+  EXPECT_EQ(init_packet.descriptors()[0].type, CookieEchoChunk::kType);
+
+  AdvanceTime(a, z, a.options.t1_init_timeout);
+
+  // Z reads COOKIE_ECHO, produces COOKIE_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads COOKIE_ACK.
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnected);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+}
+
+TEST(DcSctpSocketTest, ResendingCookieEchoTooManyTimesAborts) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  a.socket.Connect();
+
+  // Z reads INIT, produces INIT_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads INIT_ACK, produces COOKIE_ECHO
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  // COOKIE_ECHO is never received by Z.
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket init_packet,
+                              SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+  EXPECT_EQ(init_packet.descriptors()[0].type, CookieEchoChunk::kType);
+
+  for (int i = 0; i < *a.options.max_init_retransmits; ++i) {
+    AdvanceTime(a, z, a.options.t1_cookie_timeout * (1 << i));
+
+    // COOKIE_ECHO is resent
+    ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket resent_init_packet,
+                                SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+    EXPECT_EQ(resent_init_packet.descriptors()[0].type, CookieEchoChunk::kType);
+  }
+
+  // Another timeout, after the max init retransmits.
+  EXPECT_CALL(a.cb, OnAborted).Times(1);
+  AdvanceTime(
+      a, z,
+      a.options.t1_cookie_timeout * (1 << *a.options.max_init_retransmits));
+
+  EXPECT_EQ(a.socket.state(), SocketState::kClosed);
+}
+
+TEST(DcSctpSocketTest, DoesntSendMorePacketsUntilCookieAckHasBeenReceived) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+  a.socket.Connect();
+
+  // Z reads INIT, produces INIT_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads INIT_ACK, produces COOKIE_ECHO
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  // COOKIE_ECHO is never received by Z.
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket cookie_echo_packet1,
+                              SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+  EXPECT_THAT(cookie_echo_packet1.descriptors(), SizeIs(2));
+  EXPECT_EQ(cookie_echo_packet1.descriptors()[0].type, CookieEchoChunk::kType);
+  EXPECT_EQ(cookie_echo_packet1.descriptors()[1].type, DataChunk::kType);
+
+  EXPECT_THAT(a.cb.ConsumeSentPacket(), IsEmpty());
+
+  // There are DATA chunks in the sent packet (that was lost), which means that
+  // the T3-RTX timer is running, but as the socket is in kCookieEcho state, it
+  // will be T1-COOKIE that drives retransmissions, so when the T3-RTX expires,
+  // nothing should be retransmitted.
+  ASSERT_TRUE(a.options.rto_initial < a.options.t1_cookie_timeout);
+  AdvanceTime(a, z, a.options.rto_initial);
+  EXPECT_THAT(a.cb.ConsumeSentPacket(), IsEmpty());
+
+  // When T1-COOKIE expires, both the COOKIE-ECHO and DATA should be present.
+  AdvanceTime(a, z, a.options.t1_cookie_timeout - a.options.rto_initial);
+
+  // And this COOKIE-ECHO and DATA is also lost - never received by Z.
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket cookie_echo_packet2,
+                              SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+  EXPECT_THAT(cookie_echo_packet2.descriptors(), SizeIs(2));
+  EXPECT_EQ(cookie_echo_packet2.descriptors()[0].type, CookieEchoChunk::kType);
+  EXPECT_EQ(cookie_echo_packet2.descriptors()[1].type, DataChunk::kType);
+
+  EXPECT_THAT(a.cb.ConsumeSentPacket(), IsEmpty());
+
+  // COOKIE_ECHO has exponential backoff.
+  AdvanceTime(a, z, a.options.t1_cookie_timeout * 2);
+
+  // Z reads COOKIE_ECHO, produces COOKIE_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads COOKIE_ACK.
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnected);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+
+  ExchangeMessages(a, z);
+  EXPECT_THAT(z.cb.ConsumeReceivedMessage()->payload(),
+              SizeIs(kLargeMessageSize));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, ShutdownConnection) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  RTC_LOG(LS_INFO) << "Shutting down";
+
+  EXPECT_CALL(z->cb, OnClosed).Times(1);
+  a.socket.Shutdown();
+  // Z reads SHUTDOWN, produces SHUTDOWN_ACK
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // A reads SHUTDOWN_ACK, produces SHUTDOWN_COMPLETE
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+  // Z reads SHUTDOWN_COMPLETE.
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  EXPECT_EQ(a.socket.state(), SocketState::kClosed);
+  EXPECT_EQ(z->socket.state(), SocketState::kClosed);
+
+  z = MaybeHandoverSocket(std::move(z));
+  EXPECT_EQ(z->socket.state(), SocketState::kClosed);
+}
+
+TEST(DcSctpSocketTest, ShutdownTimerExpiresTooManyTimeClosesConnection) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  ConnectSockets(a, z);
+
+  a.socket.Shutdown();
+  // Drop first SHUTDOWN packet.
+  a.cb.ConsumeSentPacket();
+
+  EXPECT_EQ(a.socket.state(), SocketState::kShuttingDown);
+
+  for (int i = 0; i < *a.options.max_retransmissions; ++i) {
+    AdvanceTime(a, z, DurationMs(a.options.rto_initial * (1 << i)));
+
+    // Dropping every shutdown chunk.
+    ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet,
+                                SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+    EXPECT_EQ(packet.descriptors()[0].type, ShutdownChunk::kType);
+    EXPECT_TRUE(a.cb.ConsumeSentPacket().empty());
+  }
+  // The last expiry, makes it abort the connection.
+  EXPECT_CALL(a.cb, OnAborted).Times(1);
+  AdvanceTime(a, z,
+              a.options.rto_initial * (1 << *a.options.max_retransmissions));
+
+  EXPECT_EQ(a.socket.state(), SocketState::kClosed);
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet,
+                              SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+  EXPECT_EQ(packet.descriptors()[0].type, AbortChunk::kType);
+  EXPECT_TRUE(a.cb.ConsumeSentPacket().empty());
+}
+
+TEST(DcSctpSocketTest, EstablishConnectionWhileSendingData) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  a.socket.Connect();
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions);
+
+  // Z reads INIT, produces INIT_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // // A reads INIT_ACK, produces COOKIE_ECHO
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+  // // Z reads COOKIE_ECHO, produces COOKIE_ACK
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // // A reads COOKIE_ACK.
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  EXPECT_EQ(a.socket.state(), SocketState::kConnected);
+  EXPECT_EQ(z.socket.state(), SocketState::kConnected);
+
+  absl::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg.has_value());
+  EXPECT_EQ(msg->stream_id(), StreamID(1));
+}
+
+TEST(DcSctpSocketTest, SendMessageAfterEstablished) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  ConnectSockets(a, z);
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions);
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  absl::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg.has_value());
+  EXPECT_EQ(msg->stream_id(), StreamID(1));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, TimeoutResendsPacket) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions);
+  a.cb.ConsumeSentPacket();
+
+  RTC_LOG(LS_INFO) << "Advancing time";
+  AdvanceTime(a, *z, a.options.rto_initial);
+
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  absl::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg.has_value());
+  EXPECT_EQ(msg->stream_id(), StreamID(1));
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, SendALotOfBytesMissedSecondPacket) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  std::vector<uint8_t> payload(kLargeMessageSize);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), kSendOptions);
+
+  // First DATA
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // Second DATA (lost)
+  a.cb.ConsumeSentPacket();
+
+  // Retransmit and handle the rest
+  ExchangeMessages(a, *z);
+
+  absl::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg.has_value());
+  EXPECT_EQ(msg->stream_id(), StreamID(1));
+  EXPECT_THAT(msg->payload(), testing::ElementsAreArray(payload));
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, SendingHeartbeatAnswersWithAck) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  // Inject a HEARTBEAT chunk
+  SctpPacket::Builder b(a.socket.verification_tag(), DcSctpOptions());
+  uint8_t info[] = {1, 2, 3, 4};
+  Parameters::Builder params_builder;
+  params_builder.Add(HeartbeatInfoParameter(info));
+  b.Add(HeartbeatRequestChunk(params_builder.Build()));
+  a.socket.ReceivePacket(b.Build());
+
+  // HEARTBEAT_ACK is sent as a reply. Capture it.
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket ack_packet,
+                              SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+  ASSERT_THAT(ack_packet.descriptors(), SizeIs(1));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      HeartbeatAckChunk ack,
+      HeartbeatAckChunk::Parse(ack_packet.descriptors()[0].data));
+  ASSERT_HAS_VALUE_AND_ASSIGN(HeartbeatInfoParameter info_param, ack.info());
+  EXPECT_THAT(info_param.info(), ElementsAre(1, 2, 3, 4));
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, ExpectHeartbeatToBeSent) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_THAT(a.cb.ConsumeSentPacket(), IsEmpty());
+
+  AdvanceTime(a, *z, a.options.heartbeat_interval);
+
+  std::vector<uint8_t> hb_packet_raw = a.cb.ConsumeSentPacket();
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket hb_packet,
+                              SctpPacket::Parse(hb_packet_raw));
+  ASSERT_THAT(hb_packet.descriptors(), SizeIs(1));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      HeartbeatRequestChunk hb,
+      HeartbeatRequestChunk::Parse(hb_packet.descriptors()[0].data));
+  ASSERT_HAS_VALUE_AND_ASSIGN(HeartbeatInfoParameter info_param, hb.info());
+
+  // The info is a single 64-bit number.
+  EXPECT_THAT(hb.info()->info(), SizeIs(8));
+
+  // Feed it to Sock-z and expect a HEARTBEAT_ACK that will be propagated back.
+  z->socket.ReceivePacket(hb_packet_raw);
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest,
+       CloseConnectionAfterTooManyLostHeartbeats) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_CALL(z->cb, OnClosed).Times(1);
+  EXPECT_THAT(a.cb.ConsumeSentPacket(), testing::IsEmpty());
+  // Force-close socket Z so that it doesn't interfere from now on.
+  z->socket.Close();
+
+  DurationMs time_to_next_hearbeat = a.options.heartbeat_interval;
+
+  for (int i = 0; i < *a.options.max_retransmissions; ++i) {
+    RTC_LOG(LS_INFO) << "Letting HEARTBEAT interval timer expire - sending...";
+    AdvanceTime(a, *z, time_to_next_hearbeat);
+
+    // Dropping every heartbeat.
+    ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket hb_packet,
+                                SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+    EXPECT_EQ(hb_packet.descriptors()[0].type, HeartbeatRequestChunk::kType);
+
+    RTC_LOG(LS_INFO) << "Letting the heartbeat expire.";
+    AdvanceTime(a, *z, DurationMs(1000));
+
+    time_to_next_hearbeat = a.options.heartbeat_interval - DurationMs(1000);
+  }
+
+  RTC_LOG(LS_INFO) << "Letting HEARTBEAT interval timer expire - sending...";
+  AdvanceTime(a, *z, time_to_next_hearbeat);
+
+  // Last heartbeat
+  EXPECT_THAT(a.cb.ConsumeSentPacket(), Not(IsEmpty()));
+
+  EXPECT_CALL(a.cb, OnAborted).Times(1);
+  // Should suffice as exceeding RTO
+  AdvanceTime(a, *z, DurationMs(1000));
+
+  z = MaybeHandoverSocket(std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, RecoversAfterASuccessfulAck) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_THAT(a.cb.ConsumeSentPacket(), testing::IsEmpty());
+  EXPECT_CALL(z->cb, OnClosed).Times(1);
+  // Force-close socket Z so that it doesn't interfere from now on.
+  z->socket.Close();
+
+  DurationMs time_to_next_hearbeat = a.options.heartbeat_interval;
+
+  for (int i = 0; i < *a.options.max_retransmissions; ++i) {
+    AdvanceTime(a, *z, time_to_next_hearbeat);
+
+    // Dropping every heartbeat.
+    a.cb.ConsumeSentPacket();
+
+    RTC_LOG(LS_INFO) << "Letting the heartbeat expire.";
+    AdvanceTime(a, *z, DurationMs(1000));
+
+    time_to_next_hearbeat = a.options.heartbeat_interval - DurationMs(1000);
+  }
+
+  RTC_LOG(LS_INFO) << "Getting the last heartbeat - and acking it";
+  AdvanceTime(a, *z, time_to_next_hearbeat);
+
+  std::vector<uint8_t> hb_packet_raw = a.cb.ConsumeSentPacket();
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket hb_packet,
+                              SctpPacket::Parse(hb_packet_raw));
+  ASSERT_THAT(hb_packet.descriptors(), SizeIs(1));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      HeartbeatRequestChunk hb,
+      HeartbeatRequestChunk::Parse(hb_packet.descriptors()[0].data));
+
+  SctpPacket::Builder b(a.socket.verification_tag(), a.options);
+  b.Add(HeartbeatAckChunk(std::move(hb).extract_parameters()));
+  a.socket.ReceivePacket(b.Build());
+
+  // Should suffice as exceeding RTO - which will not fire.
+  EXPECT_CALL(a.cb, OnAborted).Times(0);
+  AdvanceTime(a, *z, DurationMs(1000));
+
+  EXPECT_THAT(a.cb.ConsumeSentPacket(), IsEmpty());
+
+  // Verify that we get new heartbeats again.
+  RTC_LOG(LS_INFO) << "Expecting a new heartbeat";
+  AdvanceTime(a, *z, time_to_next_hearbeat);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket another_packet,
+                              SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+  EXPECT_EQ(another_packet.descriptors()[0].type, HeartbeatRequestChunk::kType);
+}
+
+TEST_P(DcSctpSocketParametrizedTest, ResetStream) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), {});
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  absl::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg.has_value());
+  EXPECT_EQ(msg->stream_id(), StreamID(1));
+
+  // Handle SACK
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  // Reset the outgoing stream. This will directly send a RE-CONFIG.
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(1)}));
+
+  // Receiving the packet will trigger a callback, indicating that A has
+  // reset its stream. It will also send a RE-CONFIG with a response.
+  EXPECT_CALL(z->cb, OnIncomingStreamsReset).Times(1);
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  // Receiving a response will trigger a callback. Streams are now reset.
+  EXPECT_CALL(a.cb, OnStreamsResetPerformed).Times(1);
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, ResetStreamWillMakeChunksStartAtZeroSsn) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  std::vector<uint8_t> payload(a.options.mtu - 100);
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), {});
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), {});
+
+  auto packet1 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet1, HasDataChunkWithSsn(SSN(0)));
+  z->socket.ReceivePacket(packet1);
+
+  auto packet2 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet2, HasDataChunkWithSsn(SSN(1)));
+  z->socket.ReceivePacket(packet2);
+
+  // Handle SACK
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  absl::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg1.has_value());
+  EXPECT_EQ(msg1->stream_id(), StreamID(1));
+
+  absl::optional<DcSctpMessage> msg2 = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg2.has_value());
+  EXPECT_EQ(msg2->stream_id(), StreamID(1));
+
+  // Reset the outgoing stream. This will directly send a RE-CONFIG.
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(1)}));
+  // RE-CONFIG, req
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // RE-CONFIG, resp
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), {});
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), {});
+
+  auto packet3 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet3, HasDataChunkWithSsn(SSN(0)));
+  z->socket.ReceivePacket(packet3);
+
+  auto packet4 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet4, HasDataChunkWithSsn(SSN(1)));
+  z->socket.ReceivePacket(packet4);
+
+  // Handle SACK
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest,
+       ResetStreamWillOnlyResetTheRequestedStreams) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  std::vector<uint8_t> payload(a.options.mtu - 100);
+
+  // Send two ordered messages on SID 1
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), {});
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), {});
+
+  auto packet1 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet1, HasDataChunkWithStreamId(StreamID(1)));
+  EXPECT_THAT(packet1, HasDataChunkWithSsn(SSN(0)));
+  z->socket.ReceivePacket(packet1);
+
+  auto packet2 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet1, HasDataChunkWithStreamId(StreamID(1)));
+  EXPECT_THAT(packet2, HasDataChunkWithSsn(SSN(1)));
+  z->socket.ReceivePacket(packet2);
+
+  // Handle SACK
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  // Do the same, for SID 3
+  a.socket.Send(DcSctpMessage(StreamID(3), PPID(53), payload), {});
+  a.socket.Send(DcSctpMessage(StreamID(3), PPID(53), payload), {});
+  auto packet3 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet3, HasDataChunkWithStreamId(StreamID(3)));
+  EXPECT_THAT(packet3, HasDataChunkWithSsn(SSN(0)));
+  z->socket.ReceivePacket(packet3);
+  auto packet4 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet4, HasDataChunkWithStreamId(StreamID(3)));
+  EXPECT_THAT(packet4, HasDataChunkWithSsn(SSN(1)));
+  z->socket.ReceivePacket(packet4);
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  // Receive all messages.
+  absl::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg1.has_value());
+  EXPECT_EQ(msg1->stream_id(), StreamID(1));
+
+  absl::optional<DcSctpMessage> msg2 = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg2.has_value());
+  EXPECT_EQ(msg2->stream_id(), StreamID(1));
+
+  absl::optional<DcSctpMessage> msg3 = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg3.has_value());
+  EXPECT_EQ(msg3->stream_id(), StreamID(3));
+
+  absl::optional<DcSctpMessage> msg4 = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg4.has_value());
+  EXPECT_EQ(msg4->stream_id(), StreamID(3));
+
+  // Reset SID 1. This will directly send a RE-CONFIG.
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(3)}));
+  // RE-CONFIG, req
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // RE-CONFIG, resp
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  // Send a message on SID 1 and 3 - SID 1 should not be reset, but 3 should.
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), {});
+
+  a.socket.Send(DcSctpMessage(StreamID(3), PPID(53), payload), {});
+
+  auto packet5 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet5, HasDataChunkWithStreamId(StreamID(1)));
+  EXPECT_THAT(packet5, HasDataChunkWithSsn(SSN(2)));  // Unchanged.
+  z->socket.ReceivePacket(packet5);
+
+  auto packet6 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet6, HasDataChunkWithStreamId(StreamID(3)));
+  EXPECT_THAT(packet6, HasDataChunkWithSsn(SSN(0)));  // Reset.
+  z->socket.ReceivePacket(packet6);
+
+  // Handle SACK
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, OnePeerReconnects) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_CALL(a.cb, OnConnectionRestarted).Times(1);
+  // Let's be evil here - reconnect while a fragmented packet was about to be
+  // sent. The receiving side should get it in full.
+  std::vector<uint8_t> payload(kLargeMessageSize);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), kSendOptions);
+
+  // First DATA
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  // Create a new association, z2 - and don't use z anymore.
+  SocketUnderTest z2("Z2");
+  z2.socket.Connect();
+
+  // Retransmit and handle the rest. As there will be some chunks in-flight that
+  // have the wrong verification tag, those will yield errors.
+  ExchangeMessages(a, z2);
+
+  absl::optional<DcSctpMessage> msg = z2.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg.has_value());
+  EXPECT_EQ(msg->stream_id(), StreamID(1));
+  EXPECT_THAT(msg->payload(), testing::ElementsAreArray(payload));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, SendMessageWithLimitedRtx) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  SendOptions send_options;
+  send_options.max_retransmissions = 0;
+  std::vector<uint8_t> payload(a.options.mtu - 100);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(51), payload), send_options);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(52), payload), send_options);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), send_options);
+
+  // First DATA
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // Second DATA (lost)
+  a.cb.ConsumeSentPacket();
+  // Third DATA
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  // Handle SACK for first DATA
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  // Handle delayed SACK for third DATA
+  AdvanceTime(a, *z, a.options.delayed_ack_max_timeout);
+
+  // Handle SACK for second DATA
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  // Now the missing data chunk will be marked as nacked, but it might still be
+  // in-flight and the reported gap could be due to out-of-order delivery. So
+  // the RetransmissionQueue will not mark it as "to be retransmitted" until
+  // after the t3-rtx timer has expired.
+  AdvanceTime(a, *z, a.options.rto_initial);
+
+  // The chunk will be marked as retransmitted, and then as abandoned, which
+  // will trigger a FORWARD-TSN to be sent.
+
+  // FORWARD-TSN (third)
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  // Which will trigger a SACK
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
+
+  absl::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg1.has_value());
+  EXPECT_EQ(msg1->ppid(), PPID(51));
+
+  absl::optional<DcSctpMessage> msg2 = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg2.has_value());
+  EXPECT_EQ(msg2->ppid(), PPID(53));
+
+  absl::optional<DcSctpMessage> msg3 = z->cb.ConsumeReceivedMessage();
+  EXPECT_FALSE(msg3.has_value());
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, SendManyFragmentedMessagesWithLimitedRtx) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  SendOptions send_options;
+  send_options.unordered = IsUnordered(true);
+  send_options.max_retransmissions = 0;
+  std::vector<uint8_t> payload(a.options.mtu * 2 - 100 /* margin */);
+  // Sending first message
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(51), payload), send_options);
+  // Sending second message
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(52), payload), send_options);
+  // Sending third message
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), send_options);
+  // Sending fourth message
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(54), payload), send_options);
+
+  // First DATA, first fragment
+  std::vector<uint8_t> packet = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet, HasDataChunkWithPPID(PPID(51)));
+  z->socket.ReceivePacket(std::move(packet));
+
+  // First DATA, second fragment (lost)
+  packet = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet, HasDataChunkWithPPID(PPID(51)));
+
+  // Second DATA, first fragment
+  packet = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet, HasDataChunkWithPPID(PPID(52)));
+  z->socket.ReceivePacket(std::move(packet));
+
+  // Second DATA, second fragment (lost)
+  packet = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet, HasDataChunkWithPPID(PPID(52)));
+  EXPECT_THAT(packet, HasDataChunkWithSsn(SSN(0)));
+
+  // Third DATA, first fragment
+  packet = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet, HasDataChunkWithPPID(PPID(53)));
+  EXPECT_THAT(packet, HasDataChunkWithSsn(SSN(0)));
+  z->socket.ReceivePacket(std::move(packet));
+
+  // Third DATA, second fragment (lost)
+  packet = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet, HasDataChunkWithPPID(PPID(53)));
+  EXPECT_THAT(packet, HasDataChunkWithSsn(SSN(0)));
+
+  // Fourth DATA, first fragment
+  packet = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet, HasDataChunkWithPPID(PPID(54)));
+  EXPECT_THAT(packet, HasDataChunkWithSsn(SSN(0)));
+  z->socket.ReceivePacket(std::move(packet));
+
+  // Fourth DATA, second fragment
+  packet = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet, HasDataChunkWithPPID(PPID(54)));
+  EXPECT_THAT(packet, HasDataChunkWithSsn(SSN(0)));
+  z->socket.ReceivePacket(std::move(packet));
+
+  ExchangeMessages(a, *z);
+
+  // Let the RTX timer expire, and exchange FORWARD-TSN/SACKs
+  AdvanceTime(a, *z, a.options.rto_initial);
+
+  ExchangeMessages(a, *z);
+
+  absl::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg1.has_value());
+  EXPECT_EQ(msg1->ppid(), PPID(54));
+
+  ASSERT_FALSE(z->cb.ConsumeReceivedMessage().has_value());
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+struct FakeChunkConfig : ChunkConfig {
+  static constexpr int kType = 0x49;
+  static constexpr size_t kHeaderSize = 4;
+  static constexpr int kVariableLengthAlignment = 0;
+};
+
+class FakeChunk : public Chunk, public TLVTrait<FakeChunkConfig> {
+ public:
+  FakeChunk() {}
+
+  FakeChunk(FakeChunk&& other) = default;
+  FakeChunk& operator=(FakeChunk&& other) = default;
+
+  void SerializeTo(std::vector<uint8_t>& out) const override {
+    AllocateTLV(out);
+  }
+  std::string ToString() const override { return "FAKE"; }
+};
+
+TEST_P(DcSctpSocketParametrizedTest, ReceivingUnknownChunkRespondsWithError) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  // Inject a FAKE chunk
+  SctpPacket::Builder b(a.socket.verification_tag(), DcSctpOptions());
+  b.Add(FakeChunk());
+  a.socket.ReceivePacket(b.Build());
+
+  // ERROR is sent as a reply. Capture it.
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket reply_packet,
+                              SctpPacket::Parse(a.cb.ConsumeSentPacket()));
+  ASSERT_THAT(reply_packet.descriptors(), SizeIs(1));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      ErrorChunk error, ErrorChunk::Parse(reply_packet.descriptors()[0].data));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      UnrecognizedChunkTypeCause cause,
+      error.error_causes().get<UnrecognizedChunkTypeCause>());
+  EXPECT_THAT(cause.unrecognized_chunk(), ElementsAre(0x49, 0x00, 0x00, 0x04));
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, ReceivingErrorChunkReportsAsCallback) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  // Inject a ERROR chunk
+  SctpPacket::Builder b(a.socket.verification_tag(), DcSctpOptions());
+  b.Add(
+      ErrorChunk(Parameters::Builder()
+                     .Add(UnrecognizedChunkTypeCause({0x49, 0x00, 0x00, 0x04}))
+                     .Build()));
+
+  EXPECT_CALL(a.cb, OnError(ErrorKind::kPeerReported,
+                            HasSubstr("Unrecognized Chunk Type")));
+  a.socket.ReceivePacket(b.Build());
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST(DcSctpSocketTest, PassingHighWatermarkWillOnlyAcceptCumAckTsn) {
+  SocketUnderTest a("A");
+
+  constexpr size_t kReceiveWindowBufferSize = 2000;
+  SocketUnderTest z(
+      "Z", {.mtu = 3000,
+            .max_receiver_window_buffer_size = kReceiveWindowBufferSize});
+
+  EXPECT_CALL(z.cb, OnClosed).Times(0);
+  EXPECT_CALL(z.cb, OnAborted).Times(0);
+
+  a.socket.Connect();
+  std::vector<uint8_t> init_data = a.cb.ConsumeSentPacket();
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket init_packet,
+                              SctpPacket::Parse(init_data));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      InitChunk init_chunk,
+      InitChunk::Parse(init_packet.descriptors()[0].data));
+  z.socket.ReceivePacket(init_data);
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  // Fill up Z2 to the high watermark limit.
+  constexpr size_t kWatermarkLimit =
+      kReceiveWindowBufferSize * ReassemblyQueue::kHighWatermarkLimit;
+  constexpr size_t kRemainingSize = kReceiveWindowBufferSize - kWatermarkLimit;
+
+  TSN tsn = init_chunk.initial_tsn();
+  AnyDataChunk::Options opts;
+  opts.is_beginning = Data::IsBeginning(true);
+  z.socket.ReceivePacket(
+      SctpPacket::Builder(z.socket.verification_tag(), z.options)
+          .Add(DataChunk(tsn, StreamID(1), SSN(0), PPID(53),
+                         std::vector<uint8_t>(kWatermarkLimit + 1), opts))
+          .Build());
+
+  // First DATA will always trigger a SACK. It's not interesting.
+  EXPECT_THAT(z.cb.ConsumeSentPacket(),
+              AllOf(HasSackWithCumAckTsn(tsn), HasSackWithNoGapAckBlocks()));
+
+  // This DATA should be accepted - it's advancing cum ack tsn.
+  z.socket.ReceivePacket(
+      SctpPacket::Builder(z.socket.verification_tag(), z.options)
+          .Add(DataChunk(AddTo(tsn, 1), StreamID(1), SSN(0), PPID(53),
+                         std::vector<uint8_t>(1),
+                         /*options=*/{}))
+          .Build());
+
+  // The receiver might have moved into delayed ack mode.
+  AdvanceTime(a, z, z.options.rto_initial);
+
+  EXPECT_THAT(
+      z.cb.ConsumeSentPacket(),
+      AllOf(HasSackWithCumAckTsn(AddTo(tsn, 1)), HasSackWithNoGapAckBlocks()));
+
+  // This DATA will not be accepted - it's not advancing cum ack tsn.
+  z.socket.ReceivePacket(
+      SctpPacket::Builder(z.socket.verification_tag(), z.options)
+          .Add(DataChunk(AddTo(tsn, 3), StreamID(1), SSN(0), PPID(53),
+                         std::vector<uint8_t>(1),
+                         /*options=*/{}))
+          .Build());
+
+  // Sack will be sent in IMMEDIATE mode when this is happening.
+  EXPECT_THAT(
+      z.cb.ConsumeSentPacket(),
+      AllOf(HasSackWithCumAckTsn(AddTo(tsn, 1)), HasSackWithNoGapAckBlocks()));
+
+  // This DATA will not be accepted either.
+  z.socket.ReceivePacket(
+      SctpPacket::Builder(z.socket.verification_tag(), z.options)
+          .Add(DataChunk(AddTo(tsn, 4), StreamID(1), SSN(0), PPID(53),
+                         std::vector<uint8_t>(1),
+                         /*options=*/{}))
+          .Build());
+
+  // Sack will be sent in IMMEDIATE mode when this is happening.
+  EXPECT_THAT(
+      z.cb.ConsumeSentPacket(),
+      AllOf(HasSackWithCumAckTsn(AddTo(tsn, 1)), HasSackWithNoGapAckBlocks()));
+
+  // This DATA should be accepted, and it fills the reassembly queue.
+  z.socket.ReceivePacket(
+      SctpPacket::Builder(z.socket.verification_tag(), z.options)
+          .Add(DataChunk(AddTo(tsn, 2), StreamID(1), SSN(0), PPID(53),
+                         std::vector<uint8_t>(kRemainingSize),
+                         /*options=*/{}))
+          .Build());
+
+  // The receiver might have moved into delayed ack mode.
+  AdvanceTime(a, z, z.options.rto_initial);
+
+  EXPECT_THAT(
+      z.cb.ConsumeSentPacket(),
+      AllOf(HasSackWithCumAckTsn(AddTo(tsn, 2)), HasSackWithNoGapAckBlocks()));
+
+  EXPECT_CALL(z.cb, OnAborted(ErrorKind::kResourceExhaustion, _));
+  EXPECT_CALL(z.cb, OnClosed).Times(0);
+
+  // This DATA will make the connection close. It's too full now.
+  z.socket.ReceivePacket(
+      SctpPacket::Builder(z.socket.verification_tag(), z.options)
+          .Add(DataChunk(AddTo(tsn, 3), StreamID(1), SSN(0), PPID(53),
+                         std::vector<uint8_t>(kSmallMessageSize),
+                         /*options=*/{}))
+          .Build());
+}
+
+TEST(DcSctpSocketTest, SetMaxMessageSize) {
+  SocketUnderTest a("A");
+
+  a.socket.SetMaxMessageSize(42u);
+  EXPECT_EQ(a.socket.options().max_message_size, 42u);
+}
+
+TEST_P(DcSctpSocketParametrizedTest, SendsMessagesWithLowLifetime) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  // Mock that the time always goes forward.
+  TimeMs now(0);
+  EXPECT_CALL(a.cb, TimeMillis).WillRepeatedly([&]() {
+    now += DurationMs(3);
+    return now;
+  });
+  EXPECT_CALL(z->cb, TimeMillis).WillRepeatedly([&]() {
+    now += DurationMs(3);
+    return now;
+  });
+
+  // Queue a few small messages with low lifetime, both ordered and unordered,
+  // and validate that all are delivered.
+  static constexpr int kIterations = 100;
+  for (int i = 0; i < kIterations; ++i) {
+    SendOptions send_options;
+    send_options.unordered = IsUnordered((i % 2) == 0);
+    send_options.lifetime = DurationMs(i % 3);  // 0, 1, 2 ms
+
+    a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), send_options);
+  }
+
+  ExchangeMessages(a, *z);
+
+  for (int i = 0; i < kIterations; ++i) {
+    EXPECT_TRUE(z->cb.ConsumeReceivedMessage().has_value());
+  }
+
+  EXPECT_FALSE(z->cb.ConsumeReceivedMessage().has_value());
+
+  // Validate that the sockets really make the time move forward.
+  EXPECT_GE(*now, kIterations * 2);
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest,
+       DiscardsMessagesWithLowLifetimeIfMustBuffer) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  SendOptions lifetime_0;
+  lifetime_0.unordered = IsUnordered(true);
+  lifetime_0.lifetime = DurationMs(0);
+
+  SendOptions lifetime_1;
+  lifetime_1.unordered = IsUnordered(true);
+  lifetime_1.lifetime = DurationMs(1);
+
+  // Mock that the time always goes forward.
+  TimeMs now(0);
+  EXPECT_CALL(a.cb, TimeMillis).WillRepeatedly([&]() {
+    now += DurationMs(3);
+    return now;
+  });
+  EXPECT_CALL(z->cb, TimeMillis).WillRepeatedly([&]() {
+    now += DurationMs(3);
+    return now;
+  });
+
+  // Fill up the send buffer with a large message.
+  std::vector<uint8_t> payload(kLargeMessageSize);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload), kSendOptions);
+
+  // And queue a few small messages with lifetime=0 or 1 ms - can't be sent.
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2, 3}), lifetime_0);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {4, 5, 6}), lifetime_1);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {7, 8, 9}), lifetime_0);
+
+  // Handle all that was sent until congestion window got full.
+  for (;;) {
+    std::vector<uint8_t> packet_from_a = a.cb.ConsumeSentPacket();
+    if (packet_from_a.empty()) {
+      break;
+    }
+    z->socket.ReceivePacket(std::move(packet_from_a));
+  }
+
+  // Shouldn't be enough to send that large message.
+  EXPECT_FALSE(z->cb.ConsumeReceivedMessage().has_value());
+
+  // Exchange the rest of the messages, with the time ever increasing.
+  ExchangeMessages(a, *z);
+
+  // The large message should be delivered. It was sent reliably.
+  ASSERT_HAS_VALUE_AND_ASSIGN(DcSctpMessage m1, z->cb.ConsumeReceivedMessage());
+  EXPECT_EQ(m1.stream_id(), StreamID(1));
+  EXPECT_THAT(m1.payload(), SizeIs(kLargeMessageSize));
+
+  // But none of the smaller messages.
+  EXPECT_FALSE(z->cb.ConsumeReceivedMessage().has_value());
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, HasReasonableBufferedAmountValues) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_EQ(a.socket.buffered_amount(StreamID(1)), 0u);
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                kSendOptions);
+  // Sending a small message will directly send it as a single packet, so
+  // nothing is left in the queue.
+  EXPECT_EQ(a.socket.buffered_amount(StreamID(1)), 0u);
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+
+  // Sending a message will directly start sending a few packets, so the
+  // buffered amount is not the full message size.
+  EXPECT_GT(a.socket.buffered_amount(StreamID(1)), 0u);
+  EXPECT_LT(a.socket.buffered_amount(StreamID(1)), kLargeMessageSize);
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST(DcSctpSocketTest, HasDefaultOnBufferedAmountLowValueZero) {
+  SocketUnderTest a("A");
+  EXPECT_EQ(a.socket.buffered_amount_low_threshold(StreamID(1)), 0u);
+}
+
+TEST_P(DcSctpSocketParametrizedTest,
+       TriggersOnBufferedAmountLowWithDefaultValueZero) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  EXPECT_CALL(a.cb, OnBufferedAmountLow).Times(0);
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_CALL(a.cb, OnBufferedAmountLow(StreamID(1)));
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                kSendOptions);
+  ExchangeMessages(a, *z);
+
+  EXPECT_CALL(a.cb, OnBufferedAmountLow).WillRepeatedly(testing::Return());
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest,
+       DoesntTriggerOnBufferedAmountLowIfBelowThreshold) {
+  static constexpr size_t kMessageSize = 1000;
+  static constexpr size_t kBufferedAmountLowThreshold = kMessageSize * 10;
+
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  a.socket.SetBufferedAmountLowThreshold(StreamID(1),
+                                         kBufferedAmountLowThreshold);
+  EXPECT_CALL(a.cb, OnBufferedAmountLow).Times(0);
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_CALL(a.cb, OnBufferedAmountLow(StreamID(1))).Times(0);
+  a.socket.Send(
+      DcSctpMessage(StreamID(1), PPID(53), std::vector<uint8_t>(kMessageSize)),
+      kSendOptions);
+  ExchangeMessages(a, *z);
+
+  a.socket.Send(
+      DcSctpMessage(StreamID(1), PPID(53), std::vector<uint8_t>(kMessageSize)),
+      kSendOptions);
+  ExchangeMessages(a, *z);
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, TriggersOnBufferedAmountMultipleTimes) {
+  static constexpr size_t kMessageSize = 1000;
+  static constexpr size_t kBufferedAmountLowThreshold = kMessageSize / 2;
+
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  a.socket.SetBufferedAmountLowThreshold(StreamID(1),
+                                         kBufferedAmountLowThreshold);
+  EXPECT_CALL(a.cb, OnBufferedAmountLow).Times(0);
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_CALL(a.cb, OnBufferedAmountLow(StreamID(1))).Times(3);
+  EXPECT_CALL(a.cb, OnBufferedAmountLow(StreamID(2))).Times(2);
+  a.socket.Send(
+      DcSctpMessage(StreamID(1), PPID(53), std::vector<uint8_t>(kMessageSize)),
+      kSendOptions);
+  ExchangeMessages(a, *z);
+
+  a.socket.Send(
+      DcSctpMessage(StreamID(2), PPID(53), std::vector<uint8_t>(kMessageSize)),
+      kSendOptions);
+  ExchangeMessages(a, *z);
+
+  a.socket.Send(
+      DcSctpMessage(StreamID(1), PPID(53), std::vector<uint8_t>(kMessageSize)),
+      kSendOptions);
+  ExchangeMessages(a, *z);
+
+  a.socket.Send(
+      DcSctpMessage(StreamID(2), PPID(53), std::vector<uint8_t>(kMessageSize)),
+      kSendOptions);
+  ExchangeMessages(a, *z);
+
+  a.socket.Send(
+      DcSctpMessage(StreamID(1), PPID(53), std::vector<uint8_t>(kMessageSize)),
+      kSendOptions);
+  ExchangeMessages(a, *z);
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest,
+       TriggersOnBufferedAmountLowOnlyWhenCrossingThreshold) {
+  static constexpr size_t kMessageSize = 1000;
+  static constexpr size_t kBufferedAmountLowThreshold = kMessageSize * 1.5;
+
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  a.socket.SetBufferedAmountLowThreshold(StreamID(1),
+                                         kBufferedAmountLowThreshold);
+  EXPECT_CALL(a.cb, OnBufferedAmountLow).Times(0);
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_CALL(a.cb, OnBufferedAmountLow).Times(0);
+
+  // Add a few messages to fill up the congestion window. When that is full,
+  // messages will start to be fully buffered.
+  while (a.socket.buffered_amount(StreamID(1)) <= kBufferedAmountLowThreshold) {
+    a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                                std::vector<uint8_t>(kMessageSize)),
+                  kSendOptions);
+  }
+  size_t initial_buffered = a.socket.buffered_amount(StreamID(1));
+  ASSERT_GT(initial_buffered, kBufferedAmountLowThreshold);
+
+  // Start ACKing packets, which will empty the send queue, and trigger the
+  // callback.
+  EXPECT_CALL(a.cb, OnBufferedAmountLow(StreamID(1))).Times(1);
+  ExchangeMessages(a, *z);
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest,
+       DoesntTriggerOnTotalBufferAmountLowWhenBelow) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_CALL(a.cb, OnTotalBufferedAmountLow).Times(0);
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+
+  ExchangeMessages(a, *z);
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest,
+       TriggersOnTotalBufferAmountLowWhenCrossingThreshold) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  EXPECT_CALL(a.cb, OnTotalBufferedAmountLow).Times(0);
+
+  // Fill up the send queue completely.
+  for (;;) {
+    if (a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                                    std::vector<uint8_t>(kLargeMessageSize)),
+                      kSendOptions) == SendStatus::kErrorResourceExhaustion) {
+      break;
+    }
+  }
+
+  EXPECT_CALL(a.cb, OnTotalBufferedAmountLow).Times(1);
+  ExchangeMessages(a, *z);
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST(DcSctpSocketTest, InitialMetricsAreUnset) {
+  SocketUnderTest a("A");
+
+  EXPECT_FALSE(a.socket.GetMetrics().has_value());
+}
+
+TEST(DcSctpSocketTest, MessageInterleavingMetricsAreSet) {
+  std::vector<std::pair<bool, bool>> combinations = {
+      {false, false}, {false, true}, {true, false}, {true, true}};
+  for (const auto& [a_enable, z_enable] : combinations) {
+    DcSctpOptions a_options = {.enable_message_interleaving = a_enable};
+    DcSctpOptions z_options = {.enable_message_interleaving = z_enable};
+
+    SocketUnderTest a("A", a_options);
+    SocketUnderTest z("Z", z_options);
+    ConnectSockets(a, z);
+
+    EXPECT_EQ(a.socket.GetMetrics()->uses_message_interleaving,
+              a_enable && z_enable);
+  }
+}
+
+TEST(DcSctpSocketTest, RxAndTxPacketMetricsIncrease) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  ConnectSockets(a, z);
+
+  const size_t initial_a_rwnd = a.options.max_receiver_window_buffer_size *
+                                ReassemblyQueue::kHighWatermarkLimit;
+
+  EXPECT_EQ(a.socket.GetMetrics()->tx_packets_count, 2u);
+  EXPECT_EQ(a.socket.GetMetrics()->rx_packets_count, 2u);
+  EXPECT_EQ(a.socket.GetMetrics()->tx_messages_count, 0u);
+  EXPECT_EQ(a.socket.GetMetrics()->cwnd_bytes,
+            a.options.cwnd_mtus_initial * a.options.mtu);
+  EXPECT_EQ(a.socket.GetMetrics()->unack_data_count, 0u);
+
+  EXPECT_EQ(z.socket.GetMetrics()->rx_packets_count, 2u);
+  EXPECT_EQ(z.socket.GetMetrics()->rx_messages_count, 0u);
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions);
+  EXPECT_EQ(a.socket.GetMetrics()->unack_data_count, 1u);
+
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());  // DATA
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());  // SACK
+  EXPECT_EQ(a.socket.GetMetrics()->peer_rwnd_bytes, initial_a_rwnd);
+  EXPECT_EQ(a.socket.GetMetrics()->unack_data_count, 0u);
+
+  EXPECT_TRUE(z.cb.ConsumeReceivedMessage().has_value());
+
+  EXPECT_EQ(a.socket.GetMetrics()->tx_packets_count, 3u);
+  EXPECT_EQ(a.socket.GetMetrics()->rx_packets_count, 3u);
+  EXPECT_EQ(a.socket.GetMetrics()->tx_messages_count, 1u);
+
+  EXPECT_EQ(z.socket.GetMetrics()->rx_packets_count, 3u);
+  EXPECT_EQ(z.socket.GetMetrics()->rx_messages_count, 1u);
+
+  // Send one more (large - fragmented), and receive the delayed SACK.
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                              std::vector<uint8_t>(a.options.mtu * 2 + 1)),
+                kSendOptions);
+  EXPECT_EQ(a.socket.GetMetrics()->unack_data_count, 3u);
+
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());  // DATA
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());  // DATA
+
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());  // SACK
+  EXPECT_EQ(a.socket.GetMetrics()->unack_data_count, 1u);
+  EXPECT_GT(a.socket.GetMetrics()->peer_rwnd_bytes, 0u);
+  EXPECT_LT(a.socket.GetMetrics()->peer_rwnd_bytes, initial_a_rwnd);
+
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());  // DATA
+
+  EXPECT_TRUE(z.cb.ConsumeReceivedMessage().has_value());
+
+  EXPECT_EQ(a.socket.GetMetrics()->tx_packets_count, 6u);
+  EXPECT_EQ(a.socket.GetMetrics()->rx_packets_count, 4u);
+  EXPECT_EQ(a.socket.GetMetrics()->tx_messages_count, 2u);
+
+  EXPECT_EQ(z.socket.GetMetrics()->rx_packets_count, 6u);
+  EXPECT_EQ(z.socket.GetMetrics()->rx_messages_count, 2u);
+
+  // Delayed sack
+  AdvanceTime(a, z, a.options.delayed_ack_max_timeout);
+
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());  // SACK
+  EXPECT_EQ(a.socket.GetMetrics()->unack_data_count, 0u);
+  EXPECT_EQ(a.socket.GetMetrics()->rx_packets_count, 5u);
+  EXPECT_EQ(a.socket.GetMetrics()->peer_rwnd_bytes, initial_a_rwnd);
+}
+
+TEST_P(DcSctpSocketParametrizedTest, UnackDataAlsoIncludesSendQueue) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+  size_t payload_bytes =
+      a.options.mtu - SctpPacket::kHeaderSize - DataChunk::kHeaderSize;
+
+  size_t expected_sent_packets = a.options.cwnd_mtus_initial;
+
+  size_t expected_queued_bytes =
+      kLargeMessageSize - expected_sent_packets * payload_bytes;
+
+  size_t expected_queued_packets = expected_queued_bytes / payload_bytes;
+
+  // Due to alignment, padding etc, it's hard to calculate the exact number, but
+  // it should be in this range.
+  EXPECT_GE(a.socket.GetMetrics()->unack_data_count,
+            expected_sent_packets + expected_queued_packets);
+
+  EXPECT_LE(a.socket.GetMetrics()->unack_data_count,
+            expected_sent_packets + expected_queued_packets + 2);
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, DoesntSendMoreThanMaxBurstPackets) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+
+  for (int i = 0; i < kMaxBurstPackets; ++i) {
+    std::vector<uint8_t> packet = a.cb.ConsumeSentPacket();
+    EXPECT_THAT(packet, Not(IsEmpty()));
+    z->socket.ReceivePacket(std::move(packet));  // DATA
+  }
+
+  EXPECT_THAT(a.cb.ConsumeSentPacket(), IsEmpty());
+
+  ExchangeMessages(a, *z);
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, SendsOnlyLargePackets) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  // A really large message, to ensure that the congestion window is often full.
+  constexpr size_t kMessageSize = 100000;
+  a.socket.Send(
+      DcSctpMessage(StreamID(1), PPID(53), std::vector<uint8_t>(kMessageSize)),
+      kSendOptions);
+
+  bool delivered_packet = false;
+  std::vector<size_t> data_packet_sizes;
+  do {
+    delivered_packet = false;
+    std::vector<uint8_t> packet_from_a = a.cb.ConsumeSentPacket();
+    if (!packet_from_a.empty()) {
+      data_packet_sizes.push_back(packet_from_a.size());
+      delivered_packet = true;
+      z->socket.ReceivePacket(std::move(packet_from_a));
+    }
+    std::vector<uint8_t> packet_from_z = z->cb.ConsumeSentPacket();
+    if (!packet_from_z.empty()) {
+      delivered_packet = true;
+      a.socket.ReceivePacket(std::move(packet_from_z));
+    }
+  } while (delivered_packet);
+
+  size_t packet_payload_bytes =
+      a.options.mtu - SctpPacket::kHeaderSize - DataChunk::kHeaderSize;
+  // +1 accounts for padding, and rounding up.
+  size_t expected_packets =
+      (kMessageSize + packet_payload_bytes - 1) / packet_payload_bytes + 1;
+  EXPECT_THAT(data_packet_sizes, SizeIs(expected_packets));
+
+  // Remove the last size - it will be the remainder. But all other sizes should
+  // be large.
+  data_packet_sizes.pop_back();
+
+  for (size_t size : data_packet_sizes) {
+    // The 4 is for padding/alignment.
+    EXPECT_GE(size, a.options.mtu - 4);
+  }
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST(DcSctpSocketTest, SendMessagesAfterHandover) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+
+  // Send message before handover to move socket to a not initial state
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions);
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  z->cb.ConsumeReceivedMessage();
+
+  z = HandoverSocket(std::move(z));
+
+  absl::optional<DcSctpMessage> msg;
+
+  RTC_LOG(LS_INFO) << "Sending A #1";
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {3, 4}), kSendOptions);
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  msg = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg.has_value());
+  EXPECT_EQ(msg->stream_id(), StreamID(1));
+  EXPECT_THAT(msg->payload(), testing::ElementsAre(3, 4));
+
+  RTC_LOG(LS_INFO) << "Sending A #2";
+
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(53), {5, 6}), kSendOptions);
+  z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
+
+  msg = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg.has_value());
+  EXPECT_EQ(msg->stream_id(), StreamID(2));
+  EXPECT_THAT(msg->payload(), testing::ElementsAre(5, 6));
+
+  RTC_LOG(LS_INFO) << "Sending Z #1";
+
+  z->socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2, 3}), kSendOptions);
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());  // ack
+  a.socket.ReceivePacket(z->cb.ConsumeSentPacket());  // data
+
+  msg = a.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg.has_value());
+  EXPECT_EQ(msg->stream_id(), StreamID(1));
+  EXPECT_THAT(msg->payload(), testing::ElementsAre(1, 2, 3));
+}
+
+TEST(DcSctpSocketTest, CanDetectDcsctpImplementation) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  ConnectSockets(a, z);
+
+  EXPECT_EQ(a.socket.peer_implementation(), SctpImplementation::kDcsctp);
+
+  // As A initiated the connection establishment, Z will not receive enough
+  // information to know about A's implementation
+  EXPECT_EQ(z.socket.peer_implementation(), SctpImplementation::kUnknown);
+}
+
+TEST(DcSctpSocketTest, BothCanDetectDcsctpImplementation) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  EXPECT_CALL(a.cb, OnConnected).Times(1);
+  EXPECT_CALL(z.cb, OnConnected).Times(1);
+  a.socket.Connect();
+  z.socket.Connect();
+
+  ExchangeMessages(a, z);
+
+  EXPECT_EQ(a.socket.peer_implementation(), SctpImplementation::kDcsctp);
+  EXPECT_EQ(z.socket.peer_implementation(), SctpImplementation::kDcsctp);
+}
+
+TEST_P(DcSctpSocketParametrizedTest, CanLoseFirstOrderedMessage) {
+  SocketUnderTest a("A");
+  auto z = std::make_unique<SocketUnderTest>("Z");
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  SendOptions send_options;
+  send_options.unordered = IsUnordered(false);
+  send_options.max_retransmissions = 0;
+  std::vector<uint8_t> payload(a.options.mtu - 100);
+
+  // Send a first message (SID=1, SSN=0)
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(51), payload), send_options);
+
+  // First DATA is lost, and retransmission timer will delete it.
+  a.cb.ConsumeSentPacket();
+  AdvanceTime(a, *z, a.options.rto_initial);
+  ExchangeMessages(a, *z);
+
+  // Send a second message (SID=0, SSN=1).
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(52), payload), send_options);
+  ExchangeMessages(a, *z);
+
+  // The Z socket should receive the second message, but not the first.
+  absl::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg.has_value());
+  EXPECT_EQ(msg->ppid(), PPID(52));
+
+  EXPECT_FALSE(z->cb.ConsumeReceivedMessage().has_value());
+
+  MaybeHandoverSocketAndSendMessage(a, std::move(z));
+}
+
+TEST(DcSctpSocketTest, ReceiveBothUnorderedAndOrderedWithSameTSN) {
+  /* This issue was found by fuzzing. */
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  a.socket.Connect();
+  std::vector<uint8_t> init_data = a.cb.ConsumeSentPacket();
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket init_packet,
+                              SctpPacket::Parse(init_data));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      InitChunk init_chunk,
+      InitChunk::Parse(init_packet.descriptors()[0].data));
+  z.socket.ReceivePacket(init_data);
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  a.socket.ReceivePacket(z.cb.ConsumeSentPacket());
+
+  // Receive a short unordered message with tsn=INITIAL_TSN+1
+  TSN tsn = init_chunk.initial_tsn();
+  AnyDataChunk::Options opts;
+  opts.is_beginning = Data::IsBeginning(true);
+  opts.is_end = Data::IsEnd(true);
+  opts.is_unordered = IsUnordered(true);
+  z.socket.ReceivePacket(
+      SctpPacket::Builder(z.socket.verification_tag(), z.options)
+          .Add(DataChunk(TSN(*tsn + 1), StreamID(1), SSN(0), PPID(53),
+                         std::vector<uint8_t>(10), opts))
+          .Build());
+
+  // Now receive a longer _ordered_ message with [INITIAL_TSN, INITIAL_TSN+1].
+  // This isn't allowed as it reuses TSN=53 with different properties, but it
+  // shouldn't cause any issues.
+  opts.is_unordered = IsUnordered(false);
+  opts.is_end = Data::IsEnd(false);
+  z.socket.ReceivePacket(
+      SctpPacket::Builder(z.socket.verification_tag(), z.options)
+          .Add(DataChunk(tsn, StreamID(1), SSN(0), PPID(53),
+                         std::vector<uint8_t>(10), opts))
+          .Build());
+
+  opts.is_beginning = Data::IsBeginning(false);
+  opts.is_end = Data::IsEnd(true);
+  z.socket.ReceivePacket(
+      SctpPacket::Builder(z.socket.verification_tag(), z.options)
+          .Add(DataChunk(TSN(*tsn + 1), StreamID(1), SSN(0), PPID(53),
+                         std::vector<uint8_t>(10), opts))
+          .Build());
+}
+
+TEST(DcSctpSocketTest, CloseTwoStreamsAtTheSameTime) {
+  // Reported as https://crbug.com/1312009.
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  EXPECT_CALL(z.cb, OnIncomingStreamsReset(ElementsAre(StreamID(1)))).Times(1);
+  EXPECT_CALL(z.cb, OnIncomingStreamsReset(ElementsAre(StreamID(2)))).Times(1);
+  EXPECT_CALL(a.cb, OnStreamsResetPerformed(ElementsAre(StreamID(1)))).Times(1);
+  EXPECT_CALL(a.cb, OnStreamsResetPerformed(ElementsAre(StreamID(2)))).Times(1);
+
+  ConnectSockets(a, z);
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(53), {1, 2}), kSendOptions);
+
+  ExchangeMessages(a, z);
+
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(1)}));
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(2)}));
+
+  ExchangeMessages(a, z);
+}
+
+TEST(DcSctpSocketTest, CloseThreeStreamsAtTheSameTime) {
+  // Similar to CloseTwoStreamsAtTheSameTime, but ensuring that the two
+  // remaining streams are reset at the same time in the second request.
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  EXPECT_CALL(z.cb, OnIncomingStreamsReset(ElementsAre(StreamID(1)))).Times(1);
+  EXPECT_CALL(z.cb, OnIncomingStreamsReset(
+                        UnorderedElementsAre(StreamID(2), StreamID(3))))
+      .Times(1);
+  EXPECT_CALL(a.cb, OnStreamsResetPerformed(ElementsAre(StreamID(1)))).Times(1);
+  EXPECT_CALL(a.cb, OnStreamsResetPerformed(
+                        UnorderedElementsAre(StreamID(2), StreamID(3))))
+      .Times(1);
+
+  ConnectSockets(a, z);
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(53), {1, 2}), kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(3), PPID(53), {1, 2}), kSendOptions);
+
+  ExchangeMessages(a, z);
+
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(1)}));
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(2)}));
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(3)}));
+
+  ExchangeMessages(a, z);
+}
+
+TEST(DcSctpSocketTest, CloseStreamsWithPendingRequest) {
+  // Checks that stream reset requests are properly paused when they can't be
+  // immediately reset - i.e. when there is already an ongoing stream reset
+  // request (and there can only be a single one in-flight).
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  EXPECT_CALL(z.cb, OnIncomingStreamsReset(ElementsAre(StreamID(1)))).Times(1);
+  EXPECT_CALL(z.cb, OnIncomingStreamsReset(
+                        UnorderedElementsAre(StreamID(2), StreamID(3))))
+      .Times(1);
+  EXPECT_CALL(a.cb, OnStreamsResetPerformed(ElementsAre(StreamID(1)))).Times(1);
+  EXPECT_CALL(a.cb, OnStreamsResetPerformed(
+                        UnorderedElementsAre(StreamID(2), StreamID(3))))
+      .Times(1);
+
+  ConnectSockets(a, z);
+
+  SendOptions send_options = {.unordered = IsUnordered(false)};
+
+  // Send a few ordered messages
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), send_options);
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(53), {1, 2}), send_options);
+  a.socket.Send(DcSctpMessage(StreamID(3), PPID(53), {1, 2}), send_options);
+
+  ExchangeMessages(a, z);
+
+  // Receive these messages
+  absl::optional<DcSctpMessage> msg1 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg1.has_value());
+  EXPECT_EQ(msg1->stream_id(), StreamID(1));
+  absl::optional<DcSctpMessage> msg2 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg2.has_value());
+  EXPECT_EQ(msg2->stream_id(), StreamID(2));
+  absl::optional<DcSctpMessage> msg3 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg3.has_value());
+  EXPECT_EQ(msg3->stream_id(), StreamID(3));
+
+  // Reset the streams - not all at once.
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(1)}));
+
+  std::vector<uint8_t> packet = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(packet, HasReconfigWithStreams(ElementsAre(StreamID(1))));
+  z.socket.ReceivePacket(std::move(packet));
+
+  // Sending more reset requests while this one is ongoing.
+
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(2)}));
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(3)}));
+
+  ExchangeMessages(a, z);
+
+  // Send a few more ordered messages
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), send_options);
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(53), {1, 2}), send_options);
+  a.socket.Send(DcSctpMessage(StreamID(3), PPID(53), {1, 2}), send_options);
+
+  ExchangeMessages(a, z);
+
+  // Receive these messages
+  absl::optional<DcSctpMessage> msg4 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg4.has_value());
+  EXPECT_EQ(msg4->stream_id(), StreamID(1));
+  absl::optional<DcSctpMessage> msg5 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg5.has_value());
+  EXPECT_EQ(msg5->stream_id(), StreamID(2));
+  absl::optional<DcSctpMessage> msg6 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg6.has_value());
+  EXPECT_EQ(msg6->stream_id(), StreamID(3));
+}
+
+TEST(DcSctpSocketTest, StreamsHaveInitialPriority) {
+  DcSctpOptions options = {.default_stream_priority = StreamPriority(42)};
+  SocketUnderTest a("A", options);
+
+  EXPECT_EQ(a.socket.GetStreamPriority(StreamID(1)),
+            options.default_stream_priority);
+
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(53), {1, 2}), kSendOptions);
+
+  EXPECT_EQ(a.socket.GetStreamPriority(StreamID(2)),
+            options.default_stream_priority);
+}
+
+TEST(DcSctpSocketTest, CanChangeStreamPriority) {
+  DcSctpOptions options = {.default_stream_priority = StreamPriority(42)};
+  SocketUnderTest a("A", options);
+
+  a.socket.SetStreamPriority(StreamID(1), StreamPriority(43));
+  EXPECT_EQ(a.socket.GetStreamPriority(StreamID(1)), StreamPriority(43));
+
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(53), {1, 2}), kSendOptions);
+
+  a.socket.SetStreamPriority(StreamID(2), StreamPriority(43));
+  EXPECT_EQ(a.socket.GetStreamPriority(StreamID(2)), StreamPriority(43));
+}
+
+TEST_P(DcSctpSocketParametrizedTest, WillHandoverPriority) {
+  DcSctpOptions options = {.default_stream_priority = StreamPriority(42)};
+  auto a = std::make_unique<SocketUnderTest>("A", options);
+  SocketUnderTest z("Z");
+
+  ConnectSockets(*a, z);
+
+  a->socket.SetStreamPriority(StreamID(1), StreamPriority(43));
+  a->socket.Send(DcSctpMessage(StreamID(2), PPID(53), {1, 2}), kSendOptions);
+  a->socket.SetStreamPriority(StreamID(2), StreamPriority(43));
+
+  ExchangeMessages(*a, z);
+
+  a = MaybeHandoverSocket(std::move(a));
+
+  EXPECT_EQ(a->socket.GetStreamPriority(StreamID(1)), StreamPriority(43));
+  EXPECT_EQ(a->socket.GetStreamPriority(StreamID(2)), StreamPriority(43));
+}
+
+TEST(DcSctpSocketTest, ReconnectSocketWithPendingStreamReset) {
+  // This is an issue found by fuzzing, and doesn't really make sense in WebRTC
+  // data channels as a SCTP connection is never ever closed and then
+  // reconnected. SCTP connections are closed when the peer connection is
+  // deleted, and then it doesn't do more with SCTP.
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  ConnectSockets(a, z);
+
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(1)}));
+
+  EXPECT_CALL(z.cb, OnAborted).Times(1);
+  a.socket.Close();
+
+  EXPECT_EQ(a.socket.state(), SocketState::kClosed);
+
+  EXPECT_CALL(a.cb, OnConnected).Times(1);
+  EXPECT_CALL(z.cb, OnConnected).Times(1);
+  a.socket.Connect();
+  ExchangeMessages(a, z);
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(2)}));
+}
+
+TEST(DcSctpSocketTest, SmallSentMessagesWithPrioWillArriveInSpecificOrder) {
+  DcSctpOptions options = {.enable_message_interleaving = true};
+  SocketUnderTest a("A", options);
+  SocketUnderTest z("A", options);
+
+  a.socket.SetStreamPriority(StreamID(1), StreamPriority(700));
+  a.socket.SetStreamPriority(StreamID(2), StreamPriority(200));
+  a.socket.SetStreamPriority(StreamID(3), StreamPriority(100));
+
+  // Enqueue messages before connecting the socket, to ensure they aren't send
+  // as soon as Send() is called.
+  a.socket.Send(DcSctpMessage(StreamID(3), PPID(301),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(101),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(201),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(102),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(103),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                kSendOptions);
+
+  ConnectSockets(a, z);
+  ExchangeMessages(a, z);
+
+  std::vector<uint32_t> received_ppids;
+  for (;;) {
+    absl::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
+    if (!msg.has_value()) {
+      break;
+    }
+    received_ppids.push_back(*msg->ppid());
+  }
+
+  EXPECT_THAT(received_ppids, ElementsAre(101, 102, 103, 201, 301));
+}
+
+TEST(DcSctpSocketTest, LargeSentMessagesWithPrioWillArriveInSpecificOrder) {
+  DcSctpOptions options = {.enable_message_interleaving = true};
+  SocketUnderTest a("A", options);
+  SocketUnderTest z("A", options);
+
+  a.socket.SetStreamPriority(StreamID(1), StreamPriority(700));
+  a.socket.SetStreamPriority(StreamID(2), StreamPriority(200));
+  a.socket.SetStreamPriority(StreamID(3), StreamPriority(100));
+
+  // Enqueue messages before connecting the socket, to ensure they aren't send
+  // as soon as Send() is called.
+  a.socket.Send(DcSctpMessage(StreamID(3), PPID(301),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(101),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(201),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(102),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+
+  ConnectSockets(a, z);
+  ExchangeMessages(a, z);
+
+  EXPECT_THAT(GetReceivedMessagePpids(z), ElementsAre(101, 102, 201, 301));
+}
+
+TEST(DcSctpSocketTest, MessageWithHigherPrioWillInterruptLowerPrioMessage) {
+  DcSctpOptions options = {.enable_message_interleaving = true};
+  SocketUnderTest a("A", options);
+  SocketUnderTest z("Z", options);
+
+  ConnectSockets(a, z);
+
+  a.socket.SetStreamPriority(StreamID(2), StreamPriority(128));
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(201),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+
+  // Due to a non-zero initial congestion window, the message will already start
+  // to send, but will not succeed to be sent completely before filling the
+  // congestion window or stopping due to reaching how many packets that can be
+  // sent at once (max burst). The important thing is that the entire message
+  // doesn't get sent in full.
+
+  // Now enqueue two messages; one small and one large higher priority message.
+  a.socket.SetStreamPriority(StreamID(1), StreamPriority(512));
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(101),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                kSendOptions);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(102),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+
+  ExchangeMessages(a, z);
+
+  EXPECT_THAT(GetReceivedMessagePpids(z), ElementsAre(101, 102, 201));
+}
+
+TEST(DcSctpSocketTest, LifecycleEventsAreGeneratedForAckedMessages) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+  ConnectSockets(a, z);
+
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(101),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                {.lifecycle_id = LifecycleId(41)});
+
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(102),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                kSendOptions);
+
+  a.socket.Send(DcSctpMessage(StreamID(2), PPID(103),
+                              std::vector<uint8_t>(kLargeMessageSize)),
+                {.lifecycle_id = LifecycleId(42)});
+
+  EXPECT_CALL(a.cb, OnLifecycleMessageDelivered(LifecycleId(41)));
+  EXPECT_CALL(a.cb, OnLifecycleEnd(LifecycleId(41)));
+  EXPECT_CALL(a.cb, OnLifecycleMessageDelivered(LifecycleId(42)));
+  EXPECT_CALL(a.cb, OnLifecycleEnd(LifecycleId(42)));
+  ExchangeMessages(a, z);
+  // In case of delayed ack.
+  AdvanceTime(a, z, a.options.delayed_ack_max_timeout);
+  ExchangeMessages(a, z);
+
+  EXPECT_THAT(GetReceivedMessagePpids(z), ElementsAre(101, 102, 103));
+}
+
+TEST(DcSctpSocketTest, LifecycleEventsForFailMaxRetransmissions) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+  ConnectSockets(a, z);
+
+  std::vector<uint8_t> payload(a.options.mtu - 100);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(51), payload),
+                {
+                    .max_retransmissions = 0,
+                    .lifecycle_id = LifecycleId(1),
+                });
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(52), payload),
+                {
+                    .max_retransmissions = 0,
+                    .lifecycle_id = LifecycleId(2),
+                });
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), payload),
+                {
+                    .max_retransmissions = 0,
+                    .lifecycle_id = LifecycleId(3),
+                });
+
+  // First DATA
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // Second DATA (lost)
+  a.cb.ConsumeSentPacket();
+
+  EXPECT_CALL(a.cb, OnLifecycleMessageDelivered(LifecycleId(1)));
+  EXPECT_CALL(a.cb, OnLifecycleEnd(LifecycleId(1)));
+  EXPECT_CALL(a.cb, OnLifecycleMessageExpired(LifecycleId(2),
+                                              /*maybe_delivered=*/true));
+  EXPECT_CALL(a.cb, OnLifecycleEnd(LifecycleId(2)));
+  EXPECT_CALL(a.cb, OnLifecycleMessageDelivered(LifecycleId(3)));
+  EXPECT_CALL(a.cb, OnLifecycleEnd(LifecycleId(3)));
+  ExchangeMessages(a, z);
+
+  // Handle delayed SACK.
+  AdvanceTime(a, z, a.options.delayed_ack_max_timeout);
+  ExchangeMessages(a, z);
+
+  // The chunk is now NACKed. Let the RTO expire, to discard the message.
+  AdvanceTime(a, z, a.options.rto_initial);
+  ExchangeMessages(a, z);
+
+  // Handle delayed SACK.
+  AdvanceTime(a, z, a.options.delayed_ack_max_timeout);
+  ExchangeMessages(a, z);
+
+  EXPECT_THAT(GetReceivedMessagePpids(z), ElementsAre(51, 53));
+}
+
+TEST(DcSctpSocketTest, LifecycleEventsForExpiredMessageWithRetransmitLimit) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+  ConnectSockets(a, z);
+
+  // Will not be able to send it in full within the congestion window, but will
+  // need to wait for SACKs to be received for more fragments to be sent.
+  std::vector<uint8_t> payload(kLargeMessageSize);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(51), payload),
+                {
+                    .max_retransmissions = 0,
+                    .lifecycle_id = LifecycleId(1),
+                });
+
+  // First DATA
+  z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
+  // Second DATA (lost)
+  a.cb.ConsumeSentPacket();
+
+  EXPECT_CALL(a.cb, OnLifecycleMessageExpired(LifecycleId(1),
+                                              /*maybe_delivered=*/false));
+  EXPECT_CALL(a.cb, OnLifecycleEnd(LifecycleId(1)));
+  ExchangeMessages(a, z);
+
+  EXPECT_THAT(GetReceivedMessagePpids(z), IsEmpty());
+}
+
+TEST(DcSctpSocketTest, LifecycleEventsForExpiredMessageWithLifetimeLimit) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  // Send it before the socket is connected, to prevent it from being sent too
+  // quickly. The idea is that it should be expired before even attempting to
+  // send it in full.
+  std::vector<uint8_t> payload(kSmallMessageSize);
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(51), payload),
+                {
+                    .lifetime = DurationMs(100),
+                    .lifecycle_id = LifecycleId(1),
+                });
+
+  AdvanceTime(a, z, DurationMs(200));
+
+  EXPECT_CALL(a.cb, OnLifecycleMessageExpired(LifecycleId(1),
+                                              /*maybe_delivered=*/false));
+  EXPECT_CALL(a.cb, OnLifecycleEnd(LifecycleId(1)));
+  ConnectSockets(a, z);
+  ExchangeMessages(a, z);
+
+  EXPECT_THAT(GetReceivedMessagePpids(z), IsEmpty());
+}
+
+TEST_P(DcSctpSocketParametrizedTest, ExposesTheNumberOfNegotiatedStreams) {
+  DcSctpOptions options_a = {
+      .announced_maximum_incoming_streams = 12,
+      .announced_maximum_outgoing_streams = 45,
+  };
+  SocketUnderTest a("A", options_a);
+
+  DcSctpOptions options_z = {
+      .announced_maximum_incoming_streams = 23,
+      .announced_maximum_outgoing_streams = 34,
+  };
+  auto z = std::make_unique<SocketUnderTest>("Z", options_z);
+
+  ConnectSockets(a, *z);
+  z = MaybeHandoverSocket(std::move(z));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(Metrics metrics_a, a.socket.GetMetrics());
+  EXPECT_EQ(metrics_a.negotiated_maximum_incoming_streams, 12);
+  EXPECT_EQ(metrics_a.negotiated_maximum_outgoing_streams, 23);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(Metrics metrics_z, z->socket.GetMetrics());
+  EXPECT_EQ(metrics_z.negotiated_maximum_incoming_streams, 23);
+  EXPECT_EQ(metrics_z.negotiated_maximum_outgoing_streams, 12);
+}
+
+TEST(DcSctpSocketTest, ResetStreamsDeferred) {
+  // Guaranteed to be fragmented into two fragments.
+  constexpr size_t kTwoFragmentsSize = DcSctpOptions::kMaxSafeMTUSize + 100;
+
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  ConnectSockets(a, z);
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(53),
+                              std::vector<uint8_t>(kTwoFragmentsSize)),
+                {});
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(54),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                {});
+
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(1)}));
+
+  auto data1 = a.cb.ConsumeSentPacket();
+  auto data2 = a.cb.ConsumeSentPacket();
+  auto data3 = a.cb.ConsumeSentPacket();
+  auto reconfig = a.cb.ConsumeSentPacket();
+
+  EXPECT_THAT(data1, HasDataChunkWithSsn(SSN(0)));
+  EXPECT_THAT(data2, HasDataChunkWithSsn(SSN(0)));
+  EXPECT_THAT(data3, HasDataChunkWithSsn(SSN(1)));
+  EXPECT_THAT(reconfig, HasReconfigWithStreams(ElementsAre(StreamID(1))));
+
+  // Receive them slightly out of order to make stream resetting deferred.
+  z.socket.ReceivePacket(reconfig);
+
+  z.socket.ReceivePacket(data1);
+  z.socket.ReceivePacket(data2);
+  z.socket.ReceivePacket(data3);
+
+  absl::optional<DcSctpMessage> msg1 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg1.has_value());
+  EXPECT_EQ(msg1->stream_id(), StreamID(1));
+  EXPECT_EQ(msg1->ppid(), PPID(53));
+  EXPECT_EQ(msg1->payload().size(), kTwoFragmentsSize);
+
+  absl::optional<DcSctpMessage> msg2 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg2.has_value());
+  EXPECT_EQ(msg2->stream_id(), StreamID(1));
+  EXPECT_EQ(msg2->ppid(), PPID(54));
+  EXPECT_EQ(msg2->payload().size(), kSmallMessageSize);
+
+  EXPECT_CALL(a.cb, OnStreamsResetPerformed(ElementsAre(StreamID(1))));
+  ExchangeMessages(a, z);
+
+  // Z sent "in progress", which will make A buffer packets until it's sure
+  // that the reconfiguration has been applied. A will retry - wait for that.
+  AdvanceTime(a, z, a.options.rto_initial);
+
+  auto reconfig2 = a.cb.ConsumeSentPacket();
+  EXPECT_THAT(reconfig2, HasReconfigWithStreams(ElementsAre(StreamID(1))));
+  EXPECT_CALL(z.cb, OnIncomingStreamsReset(ElementsAre(StreamID(1))));
+  z.socket.ReceivePacket(reconfig2);
+
+  auto reconfig3 = z.cb.ConsumeSentPacket();
+  EXPECT_THAT(reconfig3,
+              HasReconfigWithResponse(
+                  ReconfigurationResponseParameter::Result::kSuccessPerformed));
+  a.socket.ReceivePacket(reconfig3);
+
+  EXPECT_THAT(data1, HasDataChunkWithSsn(SSN(0)));
+  EXPECT_THAT(data2, HasDataChunkWithSsn(SSN(0)));
+  EXPECT_THAT(data3, HasDataChunkWithSsn(SSN(1)));
+  EXPECT_THAT(reconfig, HasReconfigWithStreams(ElementsAre(StreamID(1))));
+
+  // Send a new message after the stream has been reset.
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(55),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                {});
+  ExchangeMessages(a, z);
+
+  absl::optional<DcSctpMessage> msg3 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg3.has_value());
+  EXPECT_EQ(msg3->stream_id(), StreamID(1));
+  EXPECT_EQ(msg3->ppid(), PPID(55));
+  EXPECT_EQ(msg3->payload().size(), kSmallMessageSize);
+}
+
+TEST(DcSctpSocketTest, ResetStreamsWithPausedSenderResumesWhenPerformed) {
+  SocketUnderTest a("A");
+  SocketUnderTest z("Z");
+
+  ConnectSockets(a, z);
+
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(51),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                {});
+
+  a.socket.ResetStreams(std::vector<StreamID>({StreamID(1)}));
+
+  // Will be queued, as the stream has an outstanding reset operation.
+  a.socket.Send(DcSctpMessage(StreamID(1), PPID(52),
+                              std::vector<uint8_t>(kSmallMessageSize)),
+                {});
+
+  EXPECT_CALL(a.cb, OnStreamsResetPerformed(ElementsAre(StreamID(1))));
+  EXPECT_CALL(z.cb, OnIncomingStreamsReset(ElementsAre(StreamID(1))));
+  ExchangeMessages(a, z);
+
+  absl::optional<DcSctpMessage> msg1 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg1.has_value());
+  EXPECT_EQ(msg1->stream_id(), StreamID(1));
+  EXPECT_EQ(msg1->ppid(), PPID(51));
+  EXPECT_EQ(msg1->payload().size(), kSmallMessageSize);
+
+  absl::optional<DcSctpMessage> msg2 = z.cb.ConsumeReceivedMessage();
+  ASSERT_TRUE(msg2.has_value());
+  EXPECT_EQ(msg2->stream_id(), StreamID(1));
+  EXPECT_EQ(msg2->ppid(), PPID(52));
+  EXPECT_EQ(msg2->payload().size(), kSmallMessageSize);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/heartbeat_handler.cc b/third_party/libwebrtc/net/dcsctp/socket/heartbeat_handler.cc
new file mode 100644
index 0000000000..9588b85b59
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/heartbeat_handler.cc
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/heartbeat_handler.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/bind_front.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+#include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/timer/timer.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+
+// This is stored (in serialized form) as HeartbeatInfoParameter sent in
+// HeartbeatRequestChunk and received back in HeartbeatAckChunk. It should be
+// well understood that this data may be modified by the peer, so it can't
+// be trusted.
+//
+// It currently only stores a timestamp, in millisecond precision, to allow for
+// RTT measurements. If that would be manipulated by the peer, it would just
+// result in incorrect RTT measurements, which isn't an issue.
+class HeartbeatInfo {
+ public:
+  static constexpr size_t kBufferSize = sizeof(uint64_t);
+  static_assert(kBufferSize == 8, "Unexpected buffer size");
+
+  explicit HeartbeatInfo(TimeMs created_at) : created_at_(created_at) {}
+
+  std::vector<uint8_t> Serialize() {
+    uint32_t high_bits = static_cast<uint32_t>(*created_at_ >> 32);
+    uint32_t low_bits = static_cast<uint32_t>(*created_at_);
+
+    std::vector<uint8_t> data(kBufferSize);
+    BoundedByteWriter<kBufferSize> writer(data);
+    writer.Store32<0>(high_bits);
+    writer.Store32<4>(low_bits);
+    return data;
+  }
+
+  static absl::optional<HeartbeatInfo> Deserialize(
+      rtc::ArrayView<const uint8_t> data) {
+    if (data.size() != kBufferSize) {
+      RTC_LOG(LS_WARNING) << "Invalid heartbeat info: " << data.size()
+                          << " bytes";
+      return absl::nullopt;
+    }
+
+    BoundedByteReader<kBufferSize> reader(data);
+    uint32_t high_bits = reader.Load32<0>();
+    uint32_t low_bits = reader.Load32<4>();
+
+    uint64_t created_at = static_cast<uint64_t>(high_bits) << 32 | low_bits;
+    return HeartbeatInfo(TimeMs(created_at));
+  }
+
+  TimeMs created_at() const { return created_at_; }
+
+ private:
+  const TimeMs created_at_;
+};
+
+HeartbeatHandler::HeartbeatHandler(absl::string_view log_prefix,
+                                   const DcSctpOptions& options,
+                                   Context* context,
+                                   TimerManager* timer_manager)
+    : log_prefix_(std::string(log_prefix) + "heartbeat: "),
+      ctx_(context),
+      timer_manager_(timer_manager),
+      interval_duration_(options.heartbeat_interval),
+      interval_duration_should_include_rtt_(
+          options.heartbeat_interval_include_rtt),
+      interval_timer_(timer_manager_->CreateTimer(
+          "heartbeat-interval",
+          absl::bind_front(&HeartbeatHandler::OnIntervalTimerExpiry, this),
+          TimerOptions(interval_duration_, TimerBackoffAlgorithm::kFixed))),
+      timeout_timer_(timer_manager_->CreateTimer(
+          "heartbeat-timeout",
+          absl::bind_front(&HeartbeatHandler::OnTimeoutTimerExpiry, this),
+          TimerOptions(options.rto_initial,
+                       TimerBackoffAlgorithm::kExponential,
+                       /*max_restarts=*/0))) {
+  // The interval timer must always be running as long as the association is up.
+  RestartTimer();
+}
+
+void HeartbeatHandler::RestartTimer() {
+  if (interval_duration_ == DurationMs(0)) {
+    // Heartbeating has been disabled.
+    return;
+  }
+
+  if (interval_duration_should_include_rtt_) {
+    // The RTT should be used, but it's not easy accessible. The RTO will
+    // suffice.
+    interval_timer_->set_duration(interval_duration_ + ctx_->current_rto());
+  } else {
+    interval_timer_->set_duration(interval_duration_);
+  }
+
+  interval_timer_->Start();
+}
+
+void HeartbeatHandler::HandleHeartbeatRequest(HeartbeatRequestChunk chunk) {
+  // https://tools.ietf.org/html/rfc4960#section-8.3
+  // "The receiver of the HEARTBEAT should immediately respond with a
+  // HEARTBEAT ACK that contains the Heartbeat Information TLV, together with
+  // any other received TLVs, copied unchanged from the received HEARTBEAT
+  // chunk."
+  ctx_->Send(ctx_->PacketBuilder().Add(
+      HeartbeatAckChunk(std::move(chunk).extract_parameters())));
+}
+
+void HeartbeatHandler::HandleHeartbeatAck(HeartbeatAckChunk chunk) {
+  timeout_timer_->Stop();
+  absl::optional<HeartbeatInfoParameter> info_param = chunk.info();
+  if (!info_param.has_value()) {
+    ctx_->callbacks().OnError(
+        ErrorKind::kParseFailed,
+        "Failed to parse HEARTBEAT-ACK; No Heartbeat Info parameter");
+    return;
+  }
+  absl::optional<HeartbeatInfo> info =
+      HeartbeatInfo::Deserialize(info_param->info());
+  if (!info.has_value()) {
+    ctx_->callbacks().OnError(ErrorKind::kParseFailed,
+                              "Failed to parse HEARTBEAT-ACK; Failed to "
+                              "deserialized Heartbeat info parameter");
+    return;
+  }
+
+  TimeMs now = ctx_->callbacks().TimeMillis();
+  if (info->created_at() > TimeMs(0) && info->created_at() <= now) {
+    ctx_->ObserveRTT(now - info->created_at());
+  }
+
+  // https://tools.ietf.org/html/rfc4960#section-8.1
+  // "The counter shall be reset each time ... a HEARTBEAT ACK is received from
+  // the peer endpoint."
+  ctx_->ClearTxErrorCounter();
+}
+
+absl::optional<DurationMs> HeartbeatHandler::OnIntervalTimerExpiry() {
+  if (ctx_->is_connection_established()) {
+    HeartbeatInfo info(ctx_->callbacks().TimeMillis());
+    timeout_timer_->set_duration(ctx_->current_rto());
+    timeout_timer_->Start();
+    RTC_DLOG(LS_INFO) << log_prefix_ << "Sending HEARTBEAT with timeout "
+                      << *timeout_timer_->duration();
+
+    Parameters parameters = Parameters::Builder()
+                                .Add(HeartbeatInfoParameter(info.Serialize()))
+                                .Build();
+
+    ctx_->Send(ctx_->PacketBuilder().Add(
+        HeartbeatRequestChunk(std::move(parameters))));
+  } else {
+    RTC_DLOG(LS_VERBOSE)
+        << log_prefix_
+        << "Will not send HEARTBEAT when connection not established";
+  }
+  return absl::nullopt;
+}
+
+absl::optional<DurationMs> HeartbeatHandler::OnTimeoutTimerExpiry() {
+  // Note that the timeout timer is not restarted. It will be started again when
+  // the interval timer expires.
+  RTC_DCHECK(!timeout_timer_->is_running());
+  ctx_->IncrementTxErrorCounter("HEARTBEAT timeout");
+  return absl::nullopt;
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/heartbeat_handler.h b/third_party/libwebrtc/net/dcsctp/socket/heartbeat_handler.h
new file mode 100644
index 0000000000..14c3109534
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/heartbeat_handler.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_HEARTBEAT_HANDLER_H_
+#define NET_DCSCTP_SOCKET_HEARTBEAT_HANDLER_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/timer/timer.h"
+
+namespace dcsctp {
+
+// HeartbeatHandler handles all logic around sending heartbeats and receiving
+// the responses, as well as receiving incoming heartbeat requests.
+//
+// Heartbeats are sent on idle connections to ensure that the connection is
+// still healthy and to measure the RTT. If a number of heartbeats time out,
+// the connection will eventually be closed.
+class HeartbeatHandler {
+ public:
+  HeartbeatHandler(absl::string_view log_prefix,
+                   const DcSctpOptions& options,
+                   Context* context,
+                   TimerManager* timer_manager);
+
+  // Called when the heartbeat interval timer should be restarted. This is
+  // generally done every time data is sent, which makes the timer expire when
+  // the connection is idle.
+  void RestartTimer();
+
+  // Called on received HeartbeatRequestChunk chunks.
+  void HandleHeartbeatRequest(HeartbeatRequestChunk chunk);
+
+  // Called on received HeartbeatRequestChunk chunks.
+  void HandleHeartbeatAck(HeartbeatAckChunk chunk);
+
+ private:
+  absl::optional<DurationMs> OnIntervalTimerExpiry();
+  absl::optional<DurationMs> OnTimeoutTimerExpiry();
+
+  const std::string log_prefix_;
+  Context* ctx_;
+  TimerManager* timer_manager_;
+  // The time for a connection to be idle before a heartbeat is sent.
+  const DurationMs interval_duration_;
+  // Adding RTT to the duration will add some jitter, which is good in
+  // production, but less good in unit tests, which is why it can be disabled.
+  const bool interval_duration_should_include_rtt_;
+  const std::unique_ptr<Timer> interval_timer_;
+  const std::unique_ptr<Timer> timeout_timer_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_HEARTBEAT_HANDLER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/socket/heartbeat_handler_test.cc b/third_party/libwebrtc/net/dcsctp/socket/heartbeat_handler_test.cc
new file mode 100644
index 0000000000..faa0e3da06
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/heartbeat_handler_test.cc
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/heartbeat_handler.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+#include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/socket/mock_context.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::SizeIs;
+
+constexpr DurationMs kHeartbeatInterval = DurationMs(30'000);
+
+DcSctpOptions MakeOptions(DurationMs heartbeat_interval) {
+  DcSctpOptions options;
+  options.heartbeat_interval_include_rtt = false;
+  options.heartbeat_interval = heartbeat_interval;
+  return options;
+}
+
+class HeartbeatHandlerTestBase : public testing::Test {
+ protected:
+  explicit HeartbeatHandlerTestBase(DurationMs heartbeat_interval)
+      : options_(MakeOptions(heartbeat_interval)),
+        context_(&callbacks_),
+        timer_manager_([this](webrtc::TaskQueueBase::DelayPrecision precision) {
+          return callbacks_.CreateTimeout(precision);
+        }),
+        handler_("log: ", options_, &context_, &timer_manager_) {}
+
+  void AdvanceTime(DurationMs duration) {
+    callbacks_.AdvanceTime(duration);
+    for (;;) {
+      absl::optional<TimeoutID> timeout_id = callbacks_.GetNextExpiredTimeout();
+      if (!timeout_id.has_value()) {
+        break;
+      }
+      timer_manager_.HandleTimeout(*timeout_id);
+    }
+  }
+
+  const DcSctpOptions options_;
+  NiceMock<MockDcSctpSocketCallbacks> callbacks_;
+  NiceMock<MockContext> context_;
+  TimerManager timer_manager_;
+  HeartbeatHandler handler_;
+};
+
+class HeartbeatHandlerTest : public HeartbeatHandlerTestBase {
+ protected:
+  HeartbeatHandlerTest() : HeartbeatHandlerTestBase(kHeartbeatInterval) {}
+};
+
+class DisabledHeartbeatHandlerTest : public HeartbeatHandlerTestBase {
+ protected:
+  DisabledHeartbeatHandlerTest() : HeartbeatHandlerTestBase(DurationMs(0)) {}
+};
+
+TEST_F(HeartbeatHandlerTest, HasRunningHeartbeatIntervalTimer) {
+  AdvanceTime(options_.heartbeat_interval);
+
+  // Validate that a heartbeat request was sent.
+  std::vector<uint8_t> payload = callbacks_.ConsumeSentPacket();
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet, SctpPacket::Parse(payload));
+  ASSERT_THAT(packet.descriptors(), SizeIs(1));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      HeartbeatRequestChunk request,
+      HeartbeatRequestChunk::Parse(packet.descriptors()[0].data));
+
+  EXPECT_TRUE(request.info().has_value());
+}
+
+TEST_F(HeartbeatHandlerTest, RepliesToHeartbeatRequests) {
+  uint8_t info_data[] = {1, 2, 3, 4, 5};
+  HeartbeatRequestChunk request(
+      Parameters::Builder().Add(HeartbeatInfoParameter(info_data)).Build());
+
+  handler_.HandleHeartbeatRequest(std::move(request));
+
+  std::vector<uint8_t> payload = callbacks_.ConsumeSentPacket();
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet, SctpPacket::Parse(payload));
+  ASSERT_THAT(packet.descriptors(), SizeIs(1));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      HeartbeatAckChunk response,
+      HeartbeatAckChunk::Parse(packet.descriptors()[0].data));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      HeartbeatInfoParameter param,
+      response.parameters().get<HeartbeatInfoParameter>());
+
+  EXPECT_THAT(param.info(), ElementsAre(1, 2, 3, 4, 5));
+}
+
+TEST_F(HeartbeatHandlerTest, SendsHeartbeatRequestsOnIdleChannel) {
+  AdvanceTime(options_.heartbeat_interval);
+
+  // Grab the request, and make a response.
+  std::vector<uint8_t> payload = callbacks_.ConsumeSentPacket();
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet, SctpPacket::Parse(payload));
+  ASSERT_THAT(packet.descriptors(), SizeIs(1));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      HeartbeatRequestChunk req,
+      HeartbeatRequestChunk::Parse(packet.descriptors()[0].data));
+
+  HeartbeatAckChunk ack(std::move(req).extract_parameters());
+
+  // Respond a while later. This RTT will be measured by the handler
+  constexpr DurationMs rtt(313);
+
+  EXPECT_CALL(context_, ObserveRTT(rtt)).Times(1);
+
+  callbacks_.AdvanceTime(rtt);
+  handler_.HandleHeartbeatAck(std::move(ack));
+}
+
+TEST_F(HeartbeatHandlerTest, DoesntObserveInvalidHeartbeats) {
+  AdvanceTime(options_.heartbeat_interval);
+
+  // Grab the request, and make a response.
+  std::vector<uint8_t> payload = callbacks_.ConsumeSentPacket();
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet, SctpPacket::Parse(payload));
+  ASSERT_THAT(packet.descriptors(), SizeIs(1));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      HeartbeatRequestChunk req,
+      HeartbeatRequestChunk::Parse(packet.descriptors()[0].data));
+
+  HeartbeatAckChunk ack(std::move(req).extract_parameters());
+
+  EXPECT_CALL(context_, ObserveRTT).Times(0);
+
+  // Go backwards in time - which make the HEARTBEAT-ACK have an invalid
+  // timestamp in it, as it will be in the future.
+  callbacks_.AdvanceTime(DurationMs(-100));
+
+  handler_.HandleHeartbeatAck(std::move(ack));
+}
+
+TEST_F(HeartbeatHandlerTest, IncreasesErrorIfNotAckedInTime) {
+  DurationMs rto(105);
+  EXPECT_CALL(context_, current_rto).WillOnce(Return(rto));
+  AdvanceTime(options_.heartbeat_interval);
+
+  // Validate that a request was sent.
+  EXPECT_THAT(callbacks_.ConsumeSentPacket(), Not(IsEmpty()));
+
+  EXPECT_CALL(context_, IncrementTxErrorCounter).Times(1);
+  AdvanceTime(rto);
+}
+
+TEST_F(DisabledHeartbeatHandlerTest, IsReallyDisabled) {
+  AdvanceTime(options_.heartbeat_interval);
+
+  // Validate that a request was NOT sent.
+  EXPECT_THAT(callbacks_.ConsumeSentPacket(), IsEmpty());
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/mock_context.h b/third_party/libwebrtc/net/dcsctp/socket/mock_context.h
new file mode 100644
index 0000000000..88e71d1b35
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/mock_context.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_MOCK_CONTEXT_H_
+#define NET_DCSCTP_SOCKET_MOCK_CONTEXT_H_
+
+#include <cstdint>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/socket/mock_dcsctp_socket_callbacks.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+
+class MockContext : public Context {
+ public:
+  static constexpr TSN MyInitialTsn() { return TSN(990); }
+  static constexpr TSN PeerInitialTsn() { return TSN(10); }
+  static constexpr VerificationTag PeerVerificationTag() {
+    return VerificationTag(0x01234567);
+  }
+
+  explicit MockContext(MockDcSctpSocketCallbacks* callbacks)
+      : callbacks_(*callbacks) {
+    ON_CALL(*this, is_connection_established)
+        .WillByDefault(testing::Return(true));
+    ON_CALL(*this, my_initial_tsn)
+        .WillByDefault(testing::Return(MyInitialTsn()));
+    ON_CALL(*this, peer_initial_tsn)
+        .WillByDefault(testing::Return(PeerInitialTsn()));
+    ON_CALL(*this, callbacks).WillByDefault(testing::ReturnRef(callbacks_));
+    ON_CALL(*this, current_rto).WillByDefault(testing::Return(DurationMs(123)));
+    ON_CALL(*this, Send).WillByDefault([this](SctpPacket::Builder& builder) {
+      callbacks_.SendPacketWithStatus(builder.Build());
+    });
+  }
+
+  MOCK_METHOD(bool, is_connection_established, (), (const, override));
+  MOCK_METHOD(TSN, my_initial_tsn, (), (const, override));
+  MOCK_METHOD(TSN, peer_initial_tsn, (), (const, override));
+  MOCK_METHOD(DcSctpSocketCallbacks&, callbacks, (), (const, override));
+
+  MOCK_METHOD(void, ObserveRTT, (DurationMs rtt_ms), (override));
+  MOCK_METHOD(DurationMs, current_rto, (), (const, override));
+  MOCK_METHOD(bool,
+              IncrementTxErrorCounter,
+              (absl::string_view reason),
+              (override));
+  MOCK_METHOD(void, ClearTxErrorCounter, (), (override));
+  MOCK_METHOD(bool, HasTooManyTxErrors, (), (const, override));
+  SctpPacket::Builder PacketBuilder() const override {
+    return SctpPacket::Builder(PeerVerificationTag(), options_);
+  }
+  MOCK_METHOD(void, Send, (SctpPacket::Builder & builder), (override));
+
+  DcSctpOptions options_;
+  MockDcSctpSocketCallbacks& callbacks_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_MOCK_CONTEXT_H_
diff --git a/third_party/libwebrtc/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h b/third_party/libwebrtc/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h
new file mode 100644
index 0000000000..8b2a772fa3
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_MOCK_DCSCTP_SOCKET_CALLBACKS_H_
+#define NET_DCSCTP_SOCKET_MOCK_DCSCTP_SOCKET_CALLBACKS_H_
+
+#include <cstdint>
+#include <deque>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/timeout.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/timer/fake_timeout.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/random.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+
+namespace internal {
+// It can be argued if a mocked random number generator should be deterministic
+// or if it should be have as a "real" random number generator. In this
+// implementation, each instantiation of `MockDcSctpSocketCallbacks` will have
+// their `GetRandomInt` return different sequences, but each instantiation will
+// always generate the same sequence of random numbers. This to make it easier
+// to compare logs from tests, but still to let e.g. two different sockets (used
+// in the same test) get different random numbers, so that they don't start e.g.
+// on the same sequence number. While that isn't an issue in the protocol, it
+// just makes debugging harder as the two sockets would look exactly the same.
+//
+// In a real implementation of `DcSctpSocketCallbacks` the random number
+// generator backing `GetRandomInt` should be seeded externally and correctly.
+inline int GetUniqueSeed() {
+  static int seed = 0;
+  return ++seed;
+}
+}  // namespace internal
+
+class MockDcSctpSocketCallbacks : public DcSctpSocketCallbacks {
+ public:
+  explicit MockDcSctpSocketCallbacks(absl::string_view name = "")
+      : log_prefix_(name.empty() ? "" : std::string(name) + ": "),
+        random_(internal::GetUniqueSeed()),
+        timeout_manager_([this]() { return now_; }) {
+    ON_CALL(*this, SendPacketWithStatus)
+        .WillByDefault([this](rtc::ArrayView<const uint8_t> data) {
+          sent_packets_.emplace_back(
+              std::vector<uint8_t>(data.begin(), data.end()));
+          return SendPacketStatus::kSuccess;
+        });
+    ON_CALL(*this, OnMessageReceived)
+        .WillByDefault([this](DcSctpMessage message) {
+          received_messages_.emplace_back(std::move(message));
+        });
+
+    ON_CALL(*this, OnError)
+        .WillByDefault([this](ErrorKind error, absl::string_view message) {
+          RTC_LOG(LS_WARNING)
+              << log_prefix_ << "Socket error: " << ToString(error) << "; "
+              << message;
+        });
+    ON_CALL(*this, OnAborted)
+        .WillByDefault([this](ErrorKind error, absl::string_view message) {
+          RTC_LOG(LS_WARNING)
+              << log_prefix_ << "Socket abort: " << ToString(error) << "; "
+              << message;
+        });
+    ON_CALL(*this, TimeMillis).WillByDefault([this]() { return now_; });
+  }
+
+  MOCK_METHOD(SendPacketStatus,
+              SendPacketWithStatus,
+              (rtc::ArrayView<const uint8_t> data),
+              (override));
+
+  std::unique_ptr<Timeout> CreateTimeout(
+      webrtc::TaskQueueBase::DelayPrecision precision) override {
+    // The fake timeout manager does not implement |precision|.
+    return timeout_manager_.CreateTimeout();
+  }
+
+  MOCK_METHOD(TimeMs, TimeMillis, (), (override));
+  uint32_t GetRandomInt(uint32_t low, uint32_t high) override {
+    return random_.Rand(low, high);
+  }
+
+  MOCK_METHOD(void, OnMessageReceived, (DcSctpMessage message), (override));
+  MOCK_METHOD(void,
+              OnError,
+              (ErrorKind error, absl::string_view message),
+              (override));
+  MOCK_METHOD(void,
+              OnAborted,
+              (ErrorKind error, absl::string_view message),
+              (override));
+  MOCK_METHOD(void, OnConnected, (), (override));
+  MOCK_METHOD(void, OnClosed, (), (override));
+  MOCK_METHOD(void, OnConnectionRestarted, (), (override));
+  MOCK_METHOD(void,
+              OnStreamsResetFailed,
+              (rtc::ArrayView<const StreamID> outgoing_streams,
+               absl::string_view reason),
+              (override));
+  MOCK_METHOD(void,
+              OnStreamsResetPerformed,
+              (rtc::ArrayView<const StreamID> outgoing_streams),
+              (override));
+  MOCK_METHOD(void,
+              OnIncomingStreamsReset,
+              (rtc::ArrayView<const StreamID> incoming_streams),
+              (override));
+  MOCK_METHOD(void, OnBufferedAmountLow, (StreamID stream_id), (override));
+  MOCK_METHOD(void, OnTotalBufferedAmountLow, (), (override));
+  MOCK_METHOD(void,
+              OnLifecycleMessageExpired,
+              (LifecycleId lifecycle_id, bool maybe_delivered),
+              (override));
+  MOCK_METHOD(void,
+              OnLifecycleMessageFullySent,
+              (LifecycleId lifecycle_id),
+              (override));
+  MOCK_METHOD(void,
+              OnLifecycleMessageDelivered,
+              (LifecycleId lifecycle_id),
+              (override));
+  MOCK_METHOD(void, OnLifecycleEnd, (LifecycleId lifecycle_id), (override));
+
+  bool HasPacket() const { return !sent_packets_.empty(); }
+
+  std::vector<uint8_t> ConsumeSentPacket() {
+    if (sent_packets_.empty()) {
+      return {};
+    }
+    std::vector<uint8_t> ret = std::move(sent_packets_.front());
+    sent_packets_.pop_front();
+    return ret;
+  }
+  absl::optional<DcSctpMessage> ConsumeReceivedMessage() {
+    if (received_messages_.empty()) {
+      return absl::nullopt;
+    }
+    DcSctpMessage ret = std::move(received_messages_.front());
+    received_messages_.pop_front();
+    return ret;
+  }
+
+  void AdvanceTime(DurationMs duration_ms) { now_ = now_ + duration_ms; }
+  void SetTime(TimeMs now) { now_ = now; }
+
+  absl::optional<TimeoutID> GetNextExpiredTimeout() {
+    return timeout_manager_.GetNextExpiredTimeout();
+  }
+
+ private:
+  const std::string log_prefix_;
+  TimeMs now_ = TimeMs(0);
+  webrtc::Random random_;
+  FakeTimeoutManager timeout_manager_;
+  std::deque<std::vector<uint8_t>> sent_packets_;
+  std::deque<DcSctpMessage> received_messages_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_MOCK_DCSCTP_SOCKET_CALLBACKS_H_
diff --git a/third_party/libwebrtc/net/dcsctp/socket/packet_sender.cc b/third_party/libwebrtc/net/dcsctp/socket/packet_sender.cc
new file mode 100644
index 0000000000..85392e205d
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/packet_sender.cc
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/packet_sender.h"
+
+#include <utility>
+#include <vector>
+
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+PacketSender::PacketSender(DcSctpSocketCallbacks& callbacks,
+                           std::function<void(rtc::ArrayView<const uint8_t>,
+                                              SendPacketStatus)> on_sent_packet)
+    : callbacks_(callbacks), on_sent_packet_(std::move(on_sent_packet)) {}
+
+bool PacketSender::Send(SctpPacket::Builder& builder) {
+  if (builder.empty()) {
+    return false;
+  }
+
+  std::vector<uint8_t> payload = builder.Build();
+
+  SendPacketStatus status = callbacks_.SendPacketWithStatus(payload);
+  on_sent_packet_(payload, status);
+  switch (status) {
+    case SendPacketStatus::kSuccess: {
+      return true;
+    }
+    case SendPacketStatus::kTemporaryFailure: {
+      // TODO(boivie): Queue this packet to be retried to be sent later.
+      return false;
+    }
+
+    case SendPacketStatus::kError: {
+      // Nothing that can be done.
+      return false;
+    }
+  }
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/packet_sender.h b/third_party/libwebrtc/net/dcsctp/socket/packet_sender.h
new file mode 100644
index 0000000000..7af4d3c47b
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/packet_sender.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_PACKET_SENDER_H_
+#define NET_DCSCTP_SOCKET_PACKET_SENDER_H_
+
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+
+namespace dcsctp {
+
+// The PacketSender sends packets to the network using the provided callback
+// interface. When an attempt to send a packet is made, the `on_sent_packet`
+// callback will be triggered.
+class PacketSender {
+ public:
+  PacketSender(DcSctpSocketCallbacks& callbacks,
+               std::function<void(rtc::ArrayView<const uint8_t>,
+                                  SendPacketStatus)> on_sent_packet);
+
+  // Sends the packet, and returns true if it was sent successfully.
+  bool Send(SctpPacket::Builder& builder);
+
+ private:
+  DcSctpSocketCallbacks& callbacks_;
+
+  // Callback that will be triggered for every send attempt, indicating the
+  // status of the operation.
+  std::function<void(rtc::ArrayView<const uint8_t>, SendPacketStatus)>
+      on_sent_packet_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_PACKET_SENDER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/socket/packet_sender_test.cc b/third_party/libwebrtc/net/dcsctp/socket/packet_sender_test.cc
new file mode 100644
index 0000000000..079dc36a41
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/packet_sender_test.cc
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/packet_sender.h"
+
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
+#include "net/dcsctp/socket/mock_dcsctp_socket_callbacks.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::_;
+
+constexpr VerificationTag kVerificationTag(123);
+
+class PacketSenderTest : public testing::Test {
+ protected:
+  PacketSenderTest() : sender_(callbacks_, on_send_fn_.AsStdFunction()) {}
+
+  SctpPacket::Builder PacketBuilder() const {
+    return SctpPacket::Builder(kVerificationTag, options_);
+  }
+
+  DcSctpOptions options_;
+  testing::NiceMock<MockDcSctpSocketCallbacks> callbacks_;
+  testing::MockFunction<void(rtc::ArrayView<const uint8_t>, SendPacketStatus)>
+      on_send_fn_;
+  PacketSender sender_;
+};
+
+TEST_F(PacketSenderTest, SendPacketCallsCallback) {
+  EXPECT_CALL(on_send_fn_, Call(_, SendPacketStatus::kSuccess));
+  EXPECT_TRUE(sender_.Send(PacketBuilder().Add(CookieAckChunk())));
+
+  EXPECT_CALL(callbacks_, SendPacketWithStatus)
+      .WillOnce(testing::Return(SendPacketStatus::kError));
+  EXPECT_CALL(on_send_fn_, Call(_, SendPacketStatus::kError));
+  EXPECT_FALSE(sender_.Send(PacketBuilder().Add(CookieAckChunk())));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/state_cookie.cc b/third_party/libwebrtc/net/dcsctp/socket/state_cookie.cc
new file mode 100644
index 0000000000..86be77aa34
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/state_cookie.cc
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/state_cookie.h"
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/socket/capabilities.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+
+// Magic values, which the state cookie is prefixed with.
+constexpr uint32_t kMagic1 = 1684230979;
+constexpr uint32_t kMagic2 = 1414541360;
+constexpr size_t StateCookie::kCookieSize;
+
+std::vector<uint8_t> StateCookie::Serialize() {
+  std::vector<uint8_t> cookie;
+  cookie.resize(kCookieSize);
+  BoundedByteWriter<kCookieSize> buffer(cookie);
+  buffer.Store32<0>(kMagic1);
+  buffer.Store32<4>(kMagic2);
+  buffer.Store32<8>(*initiate_tag_);
+  buffer.Store32<12>(*initial_tsn_);
+  buffer.Store32<16>(a_rwnd_);
+  buffer.Store32<20>(static_cast<uint32_t>(*tie_tag_ >> 32));
+  buffer.Store32<24>(static_cast<uint32_t>(*tie_tag_));
+  buffer.Store8<28>(capabilities_.partial_reliability);
+  buffer.Store8<29>(capabilities_.message_interleaving);
+  buffer.Store8<30>(capabilities_.reconfig);
+  buffer.Store16<32>(capabilities_.negotiated_maximum_incoming_streams);
+  buffer.Store16<34>(capabilities_.negotiated_maximum_outgoing_streams);
+  return cookie;
+}
+
+absl::optional<StateCookie> StateCookie::Deserialize(
+    rtc::ArrayView<const uint8_t> cookie) {
+  if (cookie.size() != kCookieSize) {
+    RTC_DLOG(LS_WARNING) << "Invalid state cookie: " << cookie.size()
+                         << " bytes";
+    return absl::nullopt;
+  }
+
+  BoundedByteReader<kCookieSize> buffer(cookie);
+  uint32_t magic1 = buffer.Load32<0>();
+  uint32_t magic2 = buffer.Load32<4>();
+  if (magic1 != kMagic1 || magic2 != kMagic2) {
+    RTC_DLOG(LS_WARNING) << "Invalid state cookie; wrong magic";
+    return absl::nullopt;
+  }
+
+  VerificationTag verification_tag(buffer.Load32<8>());
+  TSN initial_tsn(buffer.Load32<12>());
+  uint32_t a_rwnd = buffer.Load32<16>();
+  uint32_t tie_tag_upper = buffer.Load32<20>();
+  uint32_t tie_tag_lower = buffer.Load32<24>();
+  TieTag tie_tag(static_cast<uint64_t>(tie_tag_upper) << 32 |
+                 static_cast<uint64_t>(tie_tag_lower));
+  Capabilities capabilities;
+  capabilities.partial_reliability = buffer.Load8<28>() != 0;
+  capabilities.message_interleaving = buffer.Load8<29>() != 0;
+  capabilities.reconfig = buffer.Load8<30>() != 0;
+  capabilities.negotiated_maximum_incoming_streams = buffer.Load16<32>();
+  capabilities.negotiated_maximum_outgoing_streams = buffer.Load16<34>();
+
+  return StateCookie(verification_tag, initial_tsn, a_rwnd, tie_tag,
+                     capabilities);
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/state_cookie.h b/third_party/libwebrtc/net/dcsctp/socket/state_cookie.h
new file mode 100644
index 0000000000..a26dbf86f7
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/state_cookie.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_STATE_COOKIE_H_
+#define NET_DCSCTP_SOCKET_STATE_COOKIE_H_
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/socket/capabilities.h"
+
+namespace dcsctp {
+
+// This is serialized as a state cookie and put in INIT_ACK. The client then
+// responds with this in COOKIE_ECHO.
+//
+// NOTE: Expect that the client will modify it to try to exploit the library.
+// Do not trust anything in it; no pointers or anything like that.
+class StateCookie {
+ public:
+  static constexpr size_t kCookieSize = 36;
+
+  StateCookie(VerificationTag initiate_tag,
+              TSN initial_tsn,
+              uint32_t a_rwnd,
+              TieTag tie_tag,
+              Capabilities capabilities)
+      : initiate_tag_(initiate_tag),
+        initial_tsn_(initial_tsn),
+        a_rwnd_(a_rwnd),
+        tie_tag_(tie_tag),
+        capabilities_(capabilities) {}
+
+  // Returns a serialized version of this cookie.
+  std::vector<uint8_t> Serialize();
+
+  // Deserializes the cookie, and returns absl::nullopt if that failed.
+  static absl::optional<StateCookie> Deserialize(
+      rtc::ArrayView<const uint8_t> cookie);
+
+  VerificationTag initiate_tag() const { return initiate_tag_; }
+  TSN initial_tsn() const { return initial_tsn_; }
+  uint32_t a_rwnd() const { return a_rwnd_; }
+  TieTag tie_tag() const { return tie_tag_; }
+  const Capabilities& capabilities() const { return capabilities_; }
+
+ private:
+  const VerificationTag initiate_tag_;
+  const TSN initial_tsn_;
+  const uint32_t a_rwnd_;
+  const TieTag tie_tag_;
+  const Capabilities capabilities_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_STATE_COOKIE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/socket/state_cookie_test.cc b/third_party/libwebrtc/net/dcsctp/socket/state_cookie_test.cc
new file mode 100644
index 0000000000..7d8e1339ee
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/state_cookie_test.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/state_cookie.h"
+
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::SizeIs;
+
+TEST(StateCookieTest, SerializeAndDeserialize) {
+  Capabilities capabilities = {.partial_reliability = true,
+                               .message_interleaving = false,
+                               .reconfig = true,
+                               .negotiated_maximum_incoming_streams = 123,
+                               .negotiated_maximum_outgoing_streams = 234};
+  StateCookie cookie(VerificationTag(123), TSN(456),
+                     /*a_rwnd=*/789, TieTag(101112), capabilities);
+  std::vector<uint8_t> serialized = cookie.Serialize();
+  EXPECT_THAT(serialized, SizeIs(StateCookie::kCookieSize));
+  ASSERT_HAS_VALUE_AND_ASSIGN(StateCookie deserialized,
+                              StateCookie::Deserialize(serialized));
+  EXPECT_EQ(deserialized.initiate_tag(), VerificationTag(123));
+  EXPECT_EQ(deserialized.initial_tsn(), TSN(456));
+  EXPECT_EQ(deserialized.a_rwnd(), 789u);
+  EXPECT_EQ(deserialized.tie_tag(), TieTag(101112));
+  EXPECT_TRUE(deserialized.capabilities().partial_reliability);
+  EXPECT_FALSE(deserialized.capabilities().message_interleaving);
+  EXPECT_TRUE(deserialized.capabilities().reconfig);
+  EXPECT_EQ(deserialized.capabilities().negotiated_maximum_incoming_streams,
+            123);
+  EXPECT_EQ(deserialized.capabilities().negotiated_maximum_outgoing_streams,
+            234);
+}
+
+TEST(StateCookieTest, ValidateMagicValue) {
+  Capabilities capabilities = {.partial_reliability = true,
+                               .message_interleaving = false,
+                               .reconfig = true};
+  StateCookie cookie(VerificationTag(123), TSN(456),
+                     /*a_rwnd=*/789, TieTag(101112), capabilities);
+  std::vector<uint8_t> serialized = cookie.Serialize();
+  ASSERT_THAT(serialized, SizeIs(StateCookie::kCookieSize));
+
+  absl::string_view magic(reinterpret_cast<const char*>(serialized.data()), 8);
+  EXPECT_EQ(magic, "dcSCTP00");
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/stream_reset_handler.cc b/third_party/libwebrtc/net/dcsctp/socket/stream_reset_handler.cc
new file mode 100644
index 0000000000..c81b34b626
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/stream_reset_handler.cc
@@ -0,0 +1,352 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/stream_reset_handler.h"
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/common/str_join.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h"
+#include "net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h"
+#include "net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/parameter/reconfiguration_response_parameter.h"
+#include "net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+namespace {
+using ResponseResult = ReconfigurationResponseParameter::Result;
+
+bool DescriptorsAre(const std::vector<ParameterDescriptor>& c,
+                    uint16_t e1,
+                    uint16_t e2) {
+  return (c[0].type == e1 && c[1].type == e2) ||
+         (c[0].type == e2 && c[1].type == e1);
+}
+
+}  // namespace
+
+bool StreamResetHandler::Validate(const ReConfigChunk& chunk) {
+  const Parameters& parameters = chunk.parameters();
+
+  // https://tools.ietf.org/html/rfc6525#section-3.1
+  // "Note that each RE-CONFIG chunk holds at least one parameter
+  // and at most two parameters. Only the following combinations are allowed:"
+  std::vector<ParameterDescriptor> descriptors = parameters.descriptors();
+  if (descriptors.size() == 1) {
+    if ((descriptors[0].type == OutgoingSSNResetRequestParameter::kType) ||
+        (descriptors[0].type == IncomingSSNResetRequestParameter::kType) ||
+        (descriptors[0].type == SSNTSNResetRequestParameter::kType) ||
+        (descriptors[0].type == AddOutgoingStreamsRequestParameter::kType) ||
+        (descriptors[0].type == AddIncomingStreamsRequestParameter::kType) ||
+        (descriptors[0].type == ReconfigurationResponseParameter::kType)) {
+      return true;
+    }
+  } else if (descriptors.size() == 2) {
+    if (DescriptorsAre(descriptors, OutgoingSSNResetRequestParameter::kType,
+                       IncomingSSNResetRequestParameter::kType) ||
+        DescriptorsAre(descriptors, AddOutgoingStreamsRequestParameter::kType,
+                       AddIncomingStreamsRequestParameter::kType) ||
+        DescriptorsAre(descriptors, ReconfigurationResponseParameter::kType,
+                       OutgoingSSNResetRequestParameter::kType) ||
+        DescriptorsAre(descriptors, ReconfigurationResponseParameter::kType,
+                       ReconfigurationResponseParameter::kType)) {
+      return true;
+    }
+  }
+
+  RTC_LOG(LS_WARNING) << "Invalid set of RE-CONFIG parameters";
+  return false;
+}
+
+absl::optional<std::vector<ReconfigurationResponseParameter>>
+StreamResetHandler::Process(const ReConfigChunk& chunk) {
+  if (!Validate(chunk)) {
+    return absl::nullopt;
+  }
+
+  std::vector<ReconfigurationResponseParameter> responses;
+
+  for (const ParameterDescriptor& desc : chunk.parameters().descriptors()) {
+    switch (desc.type) {
+      case OutgoingSSNResetRequestParameter::kType:
+        HandleResetOutgoing(desc, responses);
+        break;
+
+      case IncomingSSNResetRequestParameter::kType:
+        HandleResetIncoming(desc, responses);
+        break;
+
+      case ReconfigurationResponseParameter::kType:
+        HandleResponse(desc);
+        break;
+    }
+  }
+
+  return responses;
+}
+
+void StreamResetHandler::HandleReConfig(ReConfigChunk chunk) {
+  absl::optional<std::vector<ReconfigurationResponseParameter>> responses =
+      Process(chunk);
+
+  if (!responses.has_value()) {
+    ctx_->callbacks().OnError(ErrorKind::kParseFailed,
+                              "Failed to parse RE-CONFIG command");
+    return;
+  }
+
+  if (!responses->empty()) {
+    SctpPacket::Builder b = ctx_->PacketBuilder();
+    Parameters::Builder params_builder;
+    for (const auto& response : *responses) {
+      params_builder.Add(response);
+    }
+    b.Add(ReConfigChunk(params_builder.Build()));
+    ctx_->Send(b);
+  }
+}
+
+bool StreamResetHandler::ValidateReqSeqNbr(
+    ReconfigRequestSN req_seq_nbr,
+    std::vector<ReconfigurationResponseParameter>& responses) {
+  if (req_seq_nbr == last_processed_req_seq_nbr_) {
+    // https://www.rfc-editor.org/rfc/rfc6525.html#section-5.2.1 "If the
+    // received RE-CONFIG chunk contains at least one request and based on the
+    // analysis of the Re-configuration Request Sequence Numbers this is the
+    // last received RE-CONFIG chunk (i.e., a retransmission), the same
+    // RE-CONFIG chunk MUST to be sent back in response, as it was earlier."
+    RTC_DLOG(LS_VERBOSE) << log_prefix_ << "req=" << *req_seq_nbr
+                         << " already processed, returning result="
+                         << ToString(last_processed_req_result_);
+    responses.push_back(ReconfigurationResponseParameter(
+        req_seq_nbr, last_processed_req_result_));
+    return false;
+  }
+
+  if (req_seq_nbr != ReconfigRequestSN(*last_processed_req_seq_nbr_ + 1)) {
+    // Too old, too new, from wrong association etc.
+    // This is expected to happen when handing over a RTCPeerConnection from one
+    // server to another. The client will notice this and may decide to close
+    // old data channels, which may be sent to the wrong (or both) servers
+    // during a handover.
+    RTC_DLOG(LS_VERBOSE) << log_prefix_ << "req=" << *req_seq_nbr
+                         << " bad seq_nbr";
+    responses.push_back(ReconfigurationResponseParameter(
+        req_seq_nbr, ResponseResult::kErrorBadSequenceNumber));
+    return false;
+  }
+
+  return true;
+}
+
+void StreamResetHandler::HandleResetOutgoing(
+    const ParameterDescriptor& descriptor,
+    std::vector<ReconfigurationResponseParameter>& responses) {
+  absl::optional<OutgoingSSNResetRequestParameter> req =
+      OutgoingSSNResetRequestParameter::Parse(descriptor.data);
+  if (!req.has_value()) {
+    ctx_->callbacks().OnError(ErrorKind::kParseFailed,
+                              "Failed to parse Outgoing Reset command");
+    return;
+  }
+
+  if (ValidateReqSeqNbr(req->request_sequence_number(), responses)) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix_
+                         << "Reset outgoing streams with req_seq_nbr="
+                         << *req->request_sequence_number();
+
+    last_processed_req_seq_nbr_ = req->request_sequence_number();
+    last_processed_req_result_ = reassembly_queue_->ResetStreams(
+        *req, data_tracker_->last_cumulative_acked_tsn());
+    if (last_processed_req_result_ == ResponseResult::kSuccessPerformed) {
+      ctx_->callbacks().OnIncomingStreamsReset(req->stream_ids());
+    }
+    responses.push_back(ReconfigurationResponseParameter(
+        req->request_sequence_number(), last_processed_req_result_));
+  }
+}
+
+void StreamResetHandler::HandleResetIncoming(
+    const ParameterDescriptor& descriptor,
+    std::vector<ReconfigurationResponseParameter>& responses) {
+  absl::optional<IncomingSSNResetRequestParameter> req =
+      IncomingSSNResetRequestParameter::Parse(descriptor.data);
+  if (!req.has_value()) {
+    ctx_->callbacks().OnError(ErrorKind::kParseFailed,
+                              "Failed to parse Incoming Reset command");
+    return;
+  }
+  if (ValidateReqSeqNbr(req->request_sequence_number(), responses)) {
+    responses.push_back(ReconfigurationResponseParameter(
+        req->request_sequence_number(), ResponseResult::kSuccessNothingToDo));
+    last_processed_req_seq_nbr_ = req->request_sequence_number();
+  }
+}
+
+void StreamResetHandler::HandleResponse(const ParameterDescriptor& descriptor) {
+  absl::optional<ReconfigurationResponseParameter> resp =
+      ReconfigurationResponseParameter::Parse(descriptor.data);
+  if (!resp.has_value()) {
+    ctx_->callbacks().OnError(
+        ErrorKind::kParseFailed,
+        "Failed to parse Reconfiguration Response command");
+    return;
+  }
+
+  if (current_request_.has_value() && current_request_->has_been_sent() &&
+      resp->response_sequence_number() == current_request_->req_seq_nbr()) {
+    reconfig_timer_->Stop();
+
+    switch (resp->result()) {
+      case ResponseResult::kSuccessNothingToDo:
+      case ResponseResult::kSuccessPerformed:
+        RTC_DLOG(LS_VERBOSE)
+            << log_prefix_ << "Reset stream success, req_seq_nbr="
+            << *current_request_->req_seq_nbr() << ", streams="
+            << StrJoin(current_request_->streams(), ",",
+                       [](rtc::StringBuilder& sb, StreamID stream_id) {
+                         sb << *stream_id;
+                       });
+        ctx_->callbacks().OnStreamsResetPerformed(current_request_->streams());
+        current_request_ = absl::nullopt;
+        retransmission_queue_->CommitResetStreams();
+        break;
+      case ResponseResult::kInProgress:
+        RTC_DLOG(LS_VERBOSE)
+            << log_prefix_ << "Reset stream still pending, req_seq_nbr="
+            << *current_request_->req_seq_nbr() << ", streams="
+            << StrJoin(current_request_->streams(), ",",
+                       [](rtc::StringBuilder& sb, StreamID stream_id) {
+                         sb << *stream_id;
+                       });
+        // Force this request to be sent again, but with new req_seq_nbr.
+        current_request_->PrepareRetransmission();
+        reconfig_timer_->set_duration(ctx_->current_rto());
+        reconfig_timer_->Start();
+        break;
+      case ResponseResult::kErrorRequestAlreadyInProgress:
+      case ResponseResult::kDenied:
+      case ResponseResult::kErrorWrongSSN:
+      case ResponseResult::kErrorBadSequenceNumber:
+        RTC_DLOG(LS_WARNING)
+            << log_prefix_ << "Reset stream error=" << ToString(resp->result())
+            << ", req_seq_nbr=" << *current_request_->req_seq_nbr()
+            << ", streams="
+            << StrJoin(current_request_->streams(), ",",
+                       [](rtc::StringBuilder& sb, StreamID stream_id) {
+                         sb << *stream_id;
+                       });
+        ctx_->callbacks().OnStreamsResetFailed(current_request_->streams(),
+                                               ToString(resp->result()));
+        current_request_ = absl::nullopt;
+        retransmission_queue_->RollbackResetStreams();
+        break;
+    }
+  }
+}
+
+absl::optional<ReConfigChunk> StreamResetHandler::MakeStreamResetRequest() {
+  // Only send stream resets if there are streams to reset, and no current
+  // ongoing request (there can only be one at a time), and if the stream
+  // can be reset.
+  if (current_request_.has_value() ||
+      !retransmission_queue_->HasStreamsReadyToBeReset()) {
+    return absl::nullopt;
+  }
+
+  current_request_.emplace(TSN(*retransmission_queue_->next_tsn() - 1),
+                           retransmission_queue_->GetStreamsReadyToBeReset());
+  reconfig_timer_->set_duration(ctx_->current_rto());
+  reconfig_timer_->Start();
+  return MakeReconfigChunk();
+}
+
+ReConfigChunk StreamResetHandler::MakeReconfigChunk() {
+  // The req_seq_nbr will be empty if the request has never been sent before,
+  // or if it was sent, but the sender responded "in progress", and then the
+  // req_seq_nbr will be cleared to re-send with a new number. But if the
+  // request is re-sent due to timeout (reconfig-timer expiring), the same
+  // req_seq_nbr will be used.
+  RTC_DCHECK(current_request_.has_value());
+
+  if (!current_request_->has_been_sent()) {
+    current_request_->PrepareToSend(next_outgoing_req_seq_nbr_);
+    next_outgoing_req_seq_nbr_ =
+        ReconfigRequestSN(*next_outgoing_req_seq_nbr_ + 1);
+  }
+
+  Parameters::Builder params_builder =
+      Parameters::Builder().Add(OutgoingSSNResetRequestParameter(
+          current_request_->req_seq_nbr(), current_request_->req_seq_nbr(),
+          current_request_->sender_last_assigned_tsn(),
+          current_request_->streams()));
+
+  return ReConfigChunk(params_builder.Build());
+}
+
+void StreamResetHandler::ResetStreams(
+    rtc::ArrayView<const StreamID> outgoing_streams) {
+  for (StreamID stream_id : outgoing_streams) {
+    retransmission_queue_->PrepareResetStream(stream_id);
+  }
+}
+
+absl::optional<DurationMs> StreamResetHandler::OnReconfigTimerExpiry() {
+  if (current_request_->has_been_sent()) {
+    // There is an outstanding request, which timed out while waiting for a
+    // response.
+    if (!ctx_->IncrementTxErrorCounter("RECONFIG timeout")) {
+      // Timed out. The connection will close after processing the timers.
+      return absl::nullopt;
+    }
+  } else {
+    // There is no outstanding request, but there is a prepared one. This means
+    // that the receiver has previously responded "in progress", which resulted
+    // in retrying the request (but with a new req_seq_nbr) after a while.
+  }
+
+  ctx_->Send(ctx_->PacketBuilder().Add(MakeReconfigChunk()));
+  return ctx_->current_rto();
+}
+
+HandoverReadinessStatus StreamResetHandler::GetHandoverReadiness() const {
+  HandoverReadinessStatus status;
+  if (retransmission_queue_->HasStreamsReadyToBeReset()) {
+    status.Add(HandoverUnreadinessReason::kPendingStreamReset);
+  }
+  if (current_request_.has_value()) {
+    status.Add(HandoverUnreadinessReason::kPendingStreamResetRequest);
+  }
+  return status;
+}
+
+void StreamResetHandler::AddHandoverState(DcSctpSocketHandoverState& state) {
+  state.rx.last_completed_reset_req_sn = last_processed_req_seq_nbr_.value();
+  state.tx.next_reset_req_sn = next_outgoing_req_seq_nbr_.value();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/stream_reset_handler.h b/third_party/libwebrtc/net/dcsctp/socket/stream_reset_handler.h
new file mode 100644
index 0000000000..fa32e5fcc9
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/stream_reset_handler.h
@@ -0,0 +1,233 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_STREAM_RESET_HANDLER_H_
+#define NET_DCSCTP_SOCKET_STREAM_RESET_HANDLER_H_
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/bind_front.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/reconfiguration_response_parameter.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "rtc_base/containers/flat_set.h"
+
+namespace dcsctp {
+
+// StreamResetHandler handles sending outgoing stream reset requests (to close
+// an SCTP stream, which translates to closing a data channel).
+//
+// It also handles incoming "outgoing stream reset requests", when the peer
+// wants to close its data channel.
+//
+// Resetting streams is an asynchronous operation where the client will request
+// a request a stream to be reset, but then it might not be performed exactly at
+// this point. First, the sender might need to discard all messages that have
+// been enqueued for this stream, or it may select to wait until all have been
+// sent. At least, it must wait for the currently sending fragmented message to
+// be fully sent, because a stream can't be reset while having received half a
+// message. In the stream reset request, the "sender's last assigned TSN" is
+// provided, which is simply the TSN for which the receiver should've received
+// all messages before this value, before the stream can be reset. Since
+// fragments can get lost or sent out-of-order, the receiver of a request may
+// not have received all the data just yet, and then it will respond to the
+// sender: "In progress". In other words, try again. The sender will then need
+// to start a timer and try the very same request again (but with a new sequence
+// number) until the receiver successfully performs the operation.
+//
+// All this can take some time, and may be driven by timers, so the client will
+// ultimately be notified using callbacks.
+//
+// In this implementation, when a stream is reset, the queued but not-yet-sent
+// messages will be discarded, but that may change in the future. RFC8831 allows
+// both behaviors.
+class StreamResetHandler {
+ public:
+  StreamResetHandler(absl::string_view log_prefix,
+                     Context* context,
+                     TimerManager* timer_manager,
+                     DataTracker* data_tracker,
+                     ReassemblyQueue* reassembly_queue,
+                     RetransmissionQueue* retransmission_queue,
+                     const DcSctpSocketHandoverState* handover_state = nullptr)
+      : log_prefix_(std::string(log_prefix) + "reset: "),
+        ctx_(context),
+        data_tracker_(data_tracker),
+        reassembly_queue_(reassembly_queue),
+        retransmission_queue_(retransmission_queue),
+        reconfig_timer_(timer_manager->CreateTimer(
+            "re-config",
+            absl::bind_front(&StreamResetHandler::OnReconfigTimerExpiry, this),
+            TimerOptions(DurationMs(0)))),
+        next_outgoing_req_seq_nbr_(
+            handover_state
+                ? ReconfigRequestSN(handover_state->tx.next_reset_req_sn)
+                : ReconfigRequestSN(*ctx_->my_initial_tsn())),
+        last_processed_req_seq_nbr_(
+            handover_state ? ReconfigRequestSN(
+                                 handover_state->rx.last_completed_reset_req_sn)
+                           : ReconfigRequestSN(*ctx_->peer_initial_tsn() - 1)),
+        last_processed_req_result_(
+            ReconfigurationResponseParameter::Result::kSuccessNothingToDo) {}
+
+  // Initiates reset of the provided streams. While there can only be one
+  // ongoing stream reset request at any time, this method can be called at any
+  // time and also multiple times. It will enqueue requests that can't be
+  // directly fulfilled, and will asynchronously process them when any ongoing
+  // request has completed.
+  void ResetStreams(rtc::ArrayView<const StreamID> outgoing_streams);
+
+  // Creates a Reset Streams request that must be sent if returned. Will start
+  // the reconfig timer. Will return absl::nullopt if there is no need to
+  // create a request (no streams to reset) or if there already is an ongoing
+  // stream reset request that hasn't completed yet.
+  absl::optional<ReConfigChunk> MakeStreamResetRequest();
+
+  // Called when handling and incoming RE-CONFIG chunk.
+  void HandleReConfig(ReConfigChunk chunk);
+
+  HandoverReadinessStatus GetHandoverReadiness() const;
+
+  void AddHandoverState(DcSctpSocketHandoverState& state);
+
+ private:
+  // Represents a stream request operation. There can only be one ongoing at
+  // any time, and a sent request may either succeed, fail or result in the
+  // receiver signaling that it can't process it right now, and then it will be
+  // retried.
+  class CurrentRequest {
+   public:
+    CurrentRequest(TSN sender_last_assigned_tsn, std::vector<StreamID> streams)
+        : req_seq_nbr_(absl::nullopt),
+          sender_last_assigned_tsn_(sender_last_assigned_tsn),
+          streams_(std::move(streams)) {}
+
+    // Returns the current request sequence number, if this request has been
+    // sent (check `has_been_sent` first). Will return 0 if the request is just
+    // prepared (or scheduled for retransmission) but not yet sent.
+    ReconfigRequestSN req_seq_nbr() const {
+      return req_seq_nbr_.value_or(ReconfigRequestSN(0));
+    }
+
+    // The sender's last assigned TSN, from the retransmission queue. The
+    // receiver uses this to know when all data up to this TSN has been
+    // received, to know when to safely reset the stream.
+    TSN sender_last_assigned_tsn() const { return sender_last_assigned_tsn_; }
+
+    // The streams that are to be reset.
+    const std::vector<StreamID>& streams() const { return streams_; }
+
+    // If this request has been sent yet. If not, then it's either because it
+    // has only been prepared and not yet sent, or because the received couldn't
+    // apply the request, and then the exact same request will be retried, but
+    // with a new sequence number.
+    bool has_been_sent() const { return req_seq_nbr_.has_value(); }
+
+    // If the receiver can't apply the request yet (and answered "In Progress"),
+    // this will be called to prepare the request to be retransmitted at a later
+    // time.
+    void PrepareRetransmission() { req_seq_nbr_ = absl::nullopt; }
+
+    // If the request hasn't been sent yet, this assigns it a request number.
+    void PrepareToSend(ReconfigRequestSN new_req_seq_nbr) {
+      req_seq_nbr_ = new_req_seq_nbr;
+    }
+
+   private:
+    // If this is set, this request has been sent. If it's not set, the request
+    // has been prepared, but has not yet been sent. This is typically used when
+    // the peer responded "in progress" and the same request (but a different
+    // request number) must be sent again.
+    absl::optional<ReconfigRequestSN> req_seq_nbr_;
+    // The sender's (that's us) last assigned TSN, from the retransmission
+    // queue.
+    TSN sender_last_assigned_tsn_;
+    // The streams that are to be reset in this request.
+    const std::vector<StreamID> streams_;
+  };
+
+  // Called to validate an incoming RE-CONFIG chunk.
+  bool Validate(const ReConfigChunk& chunk);
+
+  // Processes a stream stream reconfiguration chunk and may either return
+  // absl::nullopt (on protocol errors), or a list of responses - either 0, 1
+  // or 2.
+  absl::optional<std::vector<ReconfigurationResponseParameter>> Process(
+      const ReConfigChunk& chunk);
+
+  // Creates the actual RE-CONFIG chunk. A request (which set `current_request`)
+  // must have been created prior.
+  ReConfigChunk MakeReconfigChunk();
+
+  // Called to validate the `req_seq_nbr`, that it's the next in sequence. If it
+  // fails to validate, and returns false, it will also add a response to
+  // `responses`.
+  bool ValidateReqSeqNbr(
+      ReconfigRequestSN req_seq_nbr,
+      std::vector<ReconfigurationResponseParameter>& responses);
+
+  // Called when this socket receives an outgoing stream reset request. It might
+  // either be performed straight away, or have to be deferred, and the result
+  // of that will be put in `responses`.
+  void HandleResetOutgoing(
+      const ParameterDescriptor& descriptor,
+      std::vector<ReconfigurationResponseParameter>& responses);
+
+  // Called when this socket receives an incoming stream reset request. This
+  // isn't really supported, but a successful response is put in `responses`.
+  void HandleResetIncoming(
+      const ParameterDescriptor& descriptor,
+      std::vector<ReconfigurationResponseParameter>& responses);
+
+  // Called when receiving a response to an outgoing stream reset request. It
+  // will either commit the stream resetting, if the operation was successful,
+  // or will schedule a retry if it was deferred. And if it failed, the
+  // operation will be rolled back.
+  void HandleResponse(const ParameterDescriptor& descriptor);
+
+  // Expiration handler for the Reconfig timer.
+  absl::optional<DurationMs> OnReconfigTimerExpiry();
+
+  const std::string log_prefix_;
+  Context* ctx_;
+  DataTracker* data_tracker_;
+  ReassemblyQueue* reassembly_queue_;
+  RetransmissionQueue* retransmission_queue_;
+  const std::unique_ptr<Timer> reconfig_timer_;
+
+  // The next sequence number for outgoing stream requests.
+  ReconfigRequestSN next_outgoing_req_seq_nbr_;
+
+  // The current stream request operation.
+  absl::optional<CurrentRequest> current_request_;
+
+  // For incoming requests - last processed request sequence number.
+  ReconfigRequestSN last_processed_req_seq_nbr_;
+  // The result from last processed incoming request
+  ReconfigurationResponseParameter::Result last_processed_req_result_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_STREAM_RESET_HANDLER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/socket/stream_reset_handler_test.cc b/third_party/libwebrtc/net/dcsctp/socket/stream_reset_handler_test.cc
new file mode 100644
index 0000000000..d3fa98752c
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/stream_reset_handler_test.cc
@@ -0,0 +1,786 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/stream_reset_handler.h"
+
+#include <array>
+#include <cstdint>
+#include <memory>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/common/handover_testing.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/parameter/reconfiguration_response_parameter.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/mock_context.h"
+#include "net/dcsctp/socket/mock_dcsctp_socket_callbacks.h"
+#include "net/dcsctp/testing/data_generator.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/mock_send_queue.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::IsEmpty;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::SizeIs;
+using ::testing::UnorderedElementsAre;
+using ResponseResult = ReconfigurationResponseParameter::Result;
+
+constexpr TSN kMyInitialTsn = MockContext::MyInitialTsn();
+constexpr ReconfigRequestSN kMyInitialReqSn = ReconfigRequestSN(*kMyInitialTsn);
+constexpr TSN kPeerInitialTsn = MockContext::PeerInitialTsn();
+constexpr ReconfigRequestSN kPeerInitialReqSn =
+    ReconfigRequestSN(*kPeerInitialTsn);
+constexpr uint32_t kArwnd = 131072;
+constexpr DurationMs kRto = DurationMs(250);
+
+constexpr std::array<uint8_t, 4> kShortPayload = {1, 2, 3, 4};
+
+MATCHER_P3(SctpMessageIs, stream_id, ppid, expected_payload, "") {
+  if (arg.stream_id() != stream_id) {
+    *result_listener << "the stream_id is " << *arg.stream_id();
+    return false;
+  }
+
+  if (arg.ppid() != ppid) {
+    *result_listener << "the ppid is " << *arg.ppid();
+    return false;
+  }
+
+  if (std::vector<uint8_t>(arg.payload().begin(), arg.payload().end()) !=
+      std::vector<uint8_t>(expected_payload.begin(), expected_payload.end())) {
+    *result_listener << "the payload is wrong";
+    return false;
+  }
+  return true;
+}
+
+TSN AddTo(TSN tsn, int delta) {
+  return TSN(*tsn + delta);
+}
+
+ReconfigRequestSN AddTo(ReconfigRequestSN req_sn, int delta) {
+  return ReconfigRequestSN(*req_sn + delta);
+}
+
+class StreamResetHandlerTest : public testing::Test {
+ protected:
+  StreamResetHandlerTest()
+      : ctx_(&callbacks_),
+        timer_manager_([this](webrtc::TaskQueueBase::DelayPrecision precision) {
+          return callbacks_.CreateTimeout(precision);
+        }),
+        delayed_ack_timer_(timer_manager_.CreateTimer(
+            "test/delayed_ack",
+            []() { return absl::nullopt; },
+            TimerOptions(DurationMs(0)))),
+        t3_rtx_timer_(timer_manager_.CreateTimer(
+            "test/t3_rtx",
+            []() { return absl::nullopt; },
+            TimerOptions(DurationMs(0)))),
+        data_tracker_(std::make_unique<DataTracker>("log: ",
+                                                    delayed_ack_timer_.get(),
+                                                    kPeerInitialTsn)),
+        reasm_(std::make_unique<ReassemblyQueue>("log: ",
+                                                 kPeerInitialTsn,
+                                                 kArwnd)),
+        retransmission_queue_(std::make_unique<RetransmissionQueue>(
+            "",
+            &callbacks_,
+            kMyInitialTsn,
+            kArwnd,
+            producer_,
+            [](DurationMs rtt_ms) {},
+            []() {},
+            *t3_rtx_timer_,
+            DcSctpOptions())),
+        handler_(
+            std::make_unique<StreamResetHandler>("log: ",
+                                                 &ctx_,
+                                                 &timer_manager_,
+                                                 data_tracker_.get(),
+                                                 reasm_.get(),
+                                                 retransmission_queue_.get())) {
+    EXPECT_CALL(ctx_, current_rto).WillRepeatedly(Return(kRto));
+  }
+
+  void AdvanceTime(DurationMs duration) {
+    callbacks_.AdvanceTime(kRto);
+    for (;;) {
+      absl::optional<TimeoutID> timeout_id = callbacks_.GetNextExpiredTimeout();
+      if (!timeout_id.has_value()) {
+        break;
+      }
+      timer_manager_.HandleTimeout(*timeout_id);
+    }
+  }
+
+  // Handles the passed in RE-CONFIG `chunk` and returns the responses
+  // that are sent in the response RE-CONFIG.
+  std::vector<ReconfigurationResponseParameter> HandleAndCatchResponse(
+      ReConfigChunk chunk) {
+    handler_->HandleReConfig(std::move(chunk));
+
+    std::vector<uint8_t> payload = callbacks_.ConsumeSentPacket();
+    if (payload.empty()) {
+      EXPECT_TRUE(false);
+      return {};
+    }
+
+    std::vector<ReconfigurationResponseParameter> responses;
+    absl::optional<SctpPacket> p = SctpPacket::Parse(payload);
+    if (!p.has_value()) {
+      EXPECT_TRUE(false);
+      return {};
+    }
+    if (p->descriptors().size() != 1) {
+      EXPECT_TRUE(false);
+      return {};
+    }
+    absl::optional<ReConfigChunk> response_chunk =
+        ReConfigChunk::Parse(p->descriptors()[0].data);
+    if (!response_chunk.has_value()) {
+      EXPECT_TRUE(false);
+      return {};
+    }
+    for (const auto& desc : response_chunk->parameters().descriptors()) {
+      if (desc.type == ReconfigurationResponseParameter::kType) {
+        absl::optional<ReconfigurationResponseParameter> response =
+            ReconfigurationResponseParameter::Parse(desc.data);
+        if (!response.has_value()) {
+          EXPECT_TRUE(false);
+          return {};
+        }
+        responses.emplace_back(*std::move(response));
+      }
+    }
+    return responses;
+  }
+
+  void PerformHandover() {
+    EXPECT_TRUE(handler_->GetHandoverReadiness().IsReady());
+    EXPECT_TRUE(data_tracker_->GetHandoverReadiness().IsReady());
+    EXPECT_TRUE(reasm_->GetHandoverReadiness().IsReady());
+    EXPECT_TRUE(retransmission_queue_->GetHandoverReadiness().IsReady());
+
+    DcSctpSocketHandoverState state;
+    handler_->AddHandoverState(state);
+    data_tracker_->AddHandoverState(state);
+    reasm_->AddHandoverState(state);
+
+    retransmission_queue_->AddHandoverState(state);
+
+    g_handover_state_transformer_for_test(&state);
+
+    data_tracker_ = std::make_unique<DataTracker>(
+        "log: ", delayed_ack_timer_.get(), kPeerInitialTsn);
+    data_tracker_->RestoreFromState(state);
+    reasm_ =
+        std::make_unique<ReassemblyQueue>("log: ", kPeerInitialTsn, kArwnd);
+    reasm_->RestoreFromState(state);
+    retransmission_queue_ = std::make_unique<RetransmissionQueue>(
+        "", &callbacks_, kMyInitialTsn, kArwnd, producer_,
+        [](DurationMs rtt_ms) {}, []() {}, *t3_rtx_timer_, DcSctpOptions(),
+        /*supports_partial_reliability=*/true,
+        /*use_message_interleaving=*/false);
+    retransmission_queue_->RestoreFromState(state);
+    handler_ = std::make_unique<StreamResetHandler>(
+        "log: ", &ctx_, &timer_manager_, data_tracker_.get(), reasm_.get(),
+        retransmission_queue_.get(), &state);
+  }
+
+  DataGenerator gen_;
+  NiceMock<MockDcSctpSocketCallbacks> callbacks_;
+  NiceMock<MockContext> ctx_;
+  NiceMock<MockSendQueue> producer_;
+  TimerManager timer_manager_;
+  std::unique_ptr<Timer> delayed_ack_timer_;
+  std::unique_ptr<Timer> t3_rtx_timer_;
+  std::unique_ptr<DataTracker> data_tracker_;
+  std::unique_ptr<ReassemblyQueue> reasm_;
+  std::unique_ptr<RetransmissionQueue> retransmission_queue_;
+  std::unique_ptr<StreamResetHandler> handler_;
+};
+
+TEST_F(StreamResetHandlerTest, ChunkWithNoParametersReturnsError) {
+  EXPECT_CALL(callbacks_, SendPacketWithStatus).Times(0);
+  EXPECT_CALL(callbacks_, OnError).Times(1);
+  handler_->HandleReConfig(ReConfigChunk(Parameters()));
+}
+
+TEST_F(StreamResetHandlerTest, ChunkWithInvalidParametersReturnsError) {
+  Parameters::Builder builder;
+  // Two OutgoingSSNResetRequestParameter in a RE-CONFIG is not valid.
+  builder.Add(OutgoingSSNResetRequestParameter(ReconfigRequestSN(1),
+                                               ReconfigRequestSN(10),
+                                               kPeerInitialTsn, {StreamID(1)}));
+  builder.Add(OutgoingSSNResetRequestParameter(ReconfigRequestSN(2),
+                                               ReconfigRequestSN(10),
+                                               kPeerInitialTsn, {StreamID(2)}));
+
+  EXPECT_CALL(callbacks_, SendPacketWithStatus).Times(0);
+  EXPECT_CALL(callbacks_, OnError).Times(1);
+  handler_->HandleReConfig(ReConfigChunk(builder.Build()));
+}
+
+TEST_F(StreamResetHandlerTest, FailToDeliverWithoutResettingStream) {
+  reasm_->Add(kPeerInitialTsn, gen_.Ordered({1, 2, 3, 4}, "BE"));
+  reasm_->Add(AddTo(kPeerInitialTsn, 1), gen_.Ordered({1, 2, 3, 4}, "BE"));
+
+  data_tracker_->Observe(kPeerInitialTsn);
+  data_tracker_->Observe(AddTo(kPeerInitialTsn, 1));
+  EXPECT_THAT(reasm_->FlushMessages(),
+              UnorderedElementsAre(
+                  SctpMessageIs(StreamID(1), PPID(53), kShortPayload),
+                  SctpMessageIs(StreamID(1), PPID(53), kShortPayload)));
+
+  gen_.ResetStream();
+  reasm_->Add(AddTo(kPeerInitialTsn, 2), gen_.Ordered({1, 2, 3, 4}, "BE"));
+  EXPECT_THAT(reasm_->FlushMessages(), IsEmpty());
+}
+
+TEST_F(StreamResetHandlerTest, ResetStreamsNotDeferred) {
+  reasm_->Add(kPeerInitialTsn, gen_.Ordered({1, 2, 3, 4}, "BE"));
+  reasm_->Add(AddTo(kPeerInitialTsn, 1), gen_.Ordered({1, 2, 3, 4}, "BE"));
+
+  data_tracker_->Observe(kPeerInitialTsn);
+  data_tracker_->Observe(AddTo(kPeerInitialTsn, 1));
+  EXPECT_THAT(reasm_->FlushMessages(),
+              UnorderedElementsAre(
+                  SctpMessageIs(StreamID(1), PPID(53), kShortPayload),
+                  SctpMessageIs(StreamID(1), PPID(53), kShortPayload)));
+
+  Parameters::Builder builder;
+  builder.Add(OutgoingSSNResetRequestParameter(
+      kPeerInitialReqSn, ReconfigRequestSN(3), AddTo(kPeerInitialTsn, 1),
+      {StreamID(1)}));
+
+  std::vector<ReconfigurationResponseParameter> responses =
+      HandleAndCatchResponse(ReConfigChunk(builder.Build()));
+  EXPECT_THAT(responses, SizeIs(1));
+  EXPECT_EQ(responses[0].result(), ResponseResult::kSuccessPerformed);
+
+  gen_.ResetStream();
+  reasm_->Add(AddTo(kPeerInitialTsn, 2), gen_.Ordered({1, 2, 3, 4}, "BE"));
+  EXPECT_THAT(reasm_->FlushMessages(),
+              UnorderedElementsAre(
+                  SctpMessageIs(StreamID(1), PPID(53), kShortPayload)));
+}
+
+TEST_F(StreamResetHandlerTest, ResetStreamsDeferred) {
+  DataGeneratorOptions opts;
+  opts.message_id = MID(0);
+  reasm_->Add(kPeerInitialTsn, gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+
+  opts.message_id = MID(1);
+  reasm_->Add(AddTo(kPeerInitialTsn, 1),
+              gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+
+  data_tracker_->Observe(kPeerInitialTsn);
+  data_tracker_->Observe(AddTo(kPeerInitialTsn, 1));
+  EXPECT_THAT(reasm_->FlushMessages(),
+              UnorderedElementsAre(
+                  SctpMessageIs(StreamID(1), PPID(53), kShortPayload),
+                  SctpMessageIs(StreamID(1), PPID(53), kShortPayload)));
+
+  Parameters::Builder builder;
+  builder.Add(OutgoingSSNResetRequestParameter(
+      kPeerInitialReqSn, ReconfigRequestSN(3), AddTo(kPeerInitialTsn, 3),
+      {StreamID(1)}));
+
+  std::vector<ReconfigurationResponseParameter> responses =
+      HandleAndCatchResponse(ReConfigChunk(builder.Build()));
+  EXPECT_THAT(responses, SizeIs(1));
+  EXPECT_EQ(responses[0].result(), ResponseResult::kInProgress);
+
+  opts.message_id = MID(1);
+  opts.ppid = PPID(5);
+  reasm_->Add(AddTo(kPeerInitialTsn, 5),
+              gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+  reasm_->MaybeResetStreamsDeferred(AddTo(kPeerInitialTsn, 1));
+
+  opts.message_id = MID(0);
+  opts.ppid = PPID(4);
+  reasm_->Add(AddTo(kPeerInitialTsn, 4),
+              gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+  reasm_->MaybeResetStreamsDeferred(AddTo(kPeerInitialTsn, 1));
+
+  opts.message_id = MID(3);
+  opts.ppid = PPID(3);
+  reasm_->Add(AddTo(kPeerInitialTsn, 3),
+              gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+  reasm_->MaybeResetStreamsDeferred(AddTo(kPeerInitialTsn, 1));
+
+  opts.message_id = MID(2);
+  opts.ppid = PPID(2);
+  reasm_->Add(AddTo(kPeerInitialTsn, 2),
+              gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+  reasm_->MaybeResetStreamsDeferred(AddTo(kPeerInitialTsn, 5));
+
+  EXPECT_THAT(
+      reasm_->FlushMessages(),
+      UnorderedElementsAre(SctpMessageIs(StreamID(1), PPID(2), kShortPayload),
+                           SctpMessageIs(StreamID(1), PPID(3), kShortPayload),
+                           SctpMessageIs(StreamID(1), PPID(4), kShortPayload),
+                           SctpMessageIs(StreamID(1), PPID(5), kShortPayload)));
+}
+
+TEST_F(StreamResetHandlerTest, SendOutgoingRequestDirectly) {
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(42)));
+  handler_->ResetStreams(std::vector<StreamID>({StreamID(42)}));
+
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+      .WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
+
+  absl::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
+  ASSERT_TRUE(reconfig.has_value());
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req,
+      reconfig->parameters().get<OutgoingSSNResetRequestParameter>());
+
+  EXPECT_EQ(req.request_sequence_number(), kMyInitialReqSn);
+  EXPECT_EQ(req.sender_last_assigned_tsn(),
+            TSN(*retransmission_queue_->next_tsn() - 1));
+  EXPECT_THAT(req.stream_ids(), UnorderedElementsAre(StreamID(42)));
+}
+
+TEST_F(StreamResetHandlerTest, ResetMultipleStreamsInOneRequest) {
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(40)));
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(41)));
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(42))).Times(2);
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(43)));
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(44)));
+  handler_->ResetStreams(std::vector<StreamID>({StreamID(42)}));
+  handler_->ResetStreams(
+      std::vector<StreamID>({StreamID(43), StreamID(44), StreamID(41)}));
+  handler_->ResetStreams(std::vector<StreamID>({StreamID(42), StreamID(40)}));
+
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+      .WillOnce(Return(
+          std::vector<StreamID>({StreamID(40), StreamID(41), StreamID(42),
+                                 StreamID(43), StreamID(44)})));
+  absl::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
+  ASSERT_TRUE(reconfig.has_value());
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req,
+      reconfig->parameters().get<OutgoingSSNResetRequestParameter>());
+
+  EXPECT_EQ(req.request_sequence_number(), kMyInitialReqSn);
+  EXPECT_EQ(req.sender_last_assigned_tsn(),
+            TSN(*retransmission_queue_->next_tsn() - 1));
+  EXPECT_THAT(req.stream_ids(),
+              UnorderedElementsAre(StreamID(40), StreamID(41), StreamID(42),
+                                   StreamID(43), StreamID(44)));
+}
+
+TEST_F(StreamResetHandlerTest, SendOutgoingRequestDeferred) {
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(42)));
+  handler_->ResetStreams(std::vector<StreamID>({StreamID(42)}));
+
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset())
+      .WillOnce(Return(false))
+      .WillOnce(Return(false))
+      .WillOnce(Return(true));
+
+  EXPECT_FALSE(handler_->MakeStreamResetRequest().has_value());
+  EXPECT_FALSE(handler_->MakeStreamResetRequest().has_value());
+  EXPECT_TRUE(handler_->MakeStreamResetRequest().has_value());
+}
+
+TEST_F(StreamResetHandlerTest, SendOutgoingResettingOnPositiveResponse) {
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(42)));
+  handler_->ResetStreams(std::vector<StreamID>({StreamID(42)}));
+
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+      .WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
+
+  absl::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
+  ASSERT_TRUE(reconfig.has_value());
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req,
+      reconfig->parameters().get<OutgoingSSNResetRequestParameter>());
+
+  Parameters::Builder builder;
+  builder.Add(ReconfigurationResponseParameter(
+      req.request_sequence_number(), ResponseResult::kSuccessPerformed));
+  ReConfigChunk response_reconfig(builder.Build());
+
+  EXPECT_CALL(producer_, CommitResetStreams);
+  EXPECT_CALL(producer_, RollbackResetStreams).Times(0);
+
+  // Processing a response shouldn't result in sending anything.
+  EXPECT_CALL(callbacks_, OnError).Times(0);
+  EXPECT_CALL(callbacks_, SendPacketWithStatus).Times(0);
+  handler_->HandleReConfig(std::move(response_reconfig));
+}
+
+TEST_F(StreamResetHandlerTest, SendOutgoingResetRollbackOnError) {
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(42)));
+  handler_->ResetStreams(std::vector<StreamID>({StreamID(42)}));
+
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+      .WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
+
+  absl::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
+  ASSERT_TRUE(reconfig.has_value());
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req,
+      reconfig->parameters().get<OutgoingSSNResetRequestParameter>());
+
+  Parameters::Builder builder;
+  builder.Add(ReconfigurationResponseParameter(
+      req.request_sequence_number(), ResponseResult::kErrorBadSequenceNumber));
+  ReConfigChunk response_reconfig(builder.Build());
+
+  EXPECT_CALL(producer_, CommitResetStreams).Times(0);
+  EXPECT_CALL(producer_, RollbackResetStreams);
+
+  // Only requests should result in sending responses.
+  EXPECT_CALL(callbacks_, OnError).Times(0);
+  EXPECT_CALL(callbacks_, SendPacketWithStatus).Times(0);
+  handler_->HandleReConfig(std::move(response_reconfig));
+}
+
+TEST_F(StreamResetHandlerTest, SendOutgoingResetRetransmitOnInProgress) {
+  static constexpr StreamID kStreamToReset = StreamID(42);
+
+  EXPECT_CALL(producer_, PrepareResetStream(kStreamToReset));
+  handler_->ResetStreams(std::vector<StreamID>({kStreamToReset}));
+
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+      .WillOnce(Return(std::vector<StreamID>({kStreamToReset})));
+
+  absl::optional<ReConfigChunk> reconfig1 = handler_->MakeStreamResetRequest();
+  ASSERT_TRUE(reconfig1.has_value());
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req1,
+      reconfig1->parameters().get<OutgoingSSNResetRequestParameter>());
+
+  // Simulate that the peer responded "In Progress".
+  Parameters::Builder builder;
+  builder.Add(ReconfigurationResponseParameter(req1.request_sequence_number(),
+                                               ResponseResult::kInProgress));
+  ReConfigChunk response_reconfig(builder.Build());
+
+  EXPECT_CALL(producer_, CommitResetStreams()).Times(0);
+  EXPECT_CALL(producer_, RollbackResetStreams()).Times(0);
+
+  // Processing a response shouldn't result in sending anything.
+  EXPECT_CALL(callbacks_, OnError).Times(0);
+  EXPECT_CALL(callbacks_, SendPacketWithStatus).Times(0);
+  handler_->HandleReConfig(std::move(response_reconfig));
+
+  // Let some time pass, so that the reconfig timer expires, and retries the
+  // same request.
+  EXPECT_CALL(callbacks_, SendPacketWithStatus).Times(1);
+  AdvanceTime(kRto);
+
+  std::vector<uint8_t> payload = callbacks_.ConsumeSentPacket();
+  ASSERT_FALSE(payload.empty());
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SctpPacket packet, SctpPacket::Parse(payload));
+  ASSERT_THAT(packet.descriptors(), SizeIs(1));
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      ReConfigChunk reconfig2,
+      ReConfigChunk::Parse(packet.descriptors()[0].data));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req2,
+      reconfig2.parameters().get<OutgoingSSNResetRequestParameter>());
+
+  EXPECT_EQ(req2.request_sequence_number(),
+            AddTo(req1.request_sequence_number(), 1));
+  EXPECT_THAT(req2.stream_ids(), UnorderedElementsAre(kStreamToReset));
+}
+
+TEST_F(StreamResetHandlerTest, ResetWhileRequestIsSentWillQueue) {
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(42)));
+  handler_->ResetStreams(std::vector<StreamID>({StreamID(42)}));
+
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+      .WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
+
+  absl::optional<ReConfigChunk> reconfig1 = handler_->MakeStreamResetRequest();
+  ASSERT_TRUE(reconfig1.has_value());
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req1,
+      reconfig1->parameters().get<OutgoingSSNResetRequestParameter>());
+  EXPECT_EQ(req1.request_sequence_number(), kMyInitialReqSn);
+  EXPECT_EQ(req1.sender_last_assigned_tsn(),
+            AddTo(retransmission_queue_->next_tsn(), -1));
+  EXPECT_THAT(req1.stream_ids(), UnorderedElementsAre(StreamID(42)));
+
+  // Streams reset while the request is in-flight will be queued.
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(41)));
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(43)));
+  StreamID stream_ids[] = {StreamID(41), StreamID(43)};
+  handler_->ResetStreams(stream_ids);
+  EXPECT_EQ(handler_->MakeStreamResetRequest(), absl::nullopt);
+
+  Parameters::Builder builder;
+  builder.Add(ReconfigurationResponseParameter(
+      req1.request_sequence_number(), ResponseResult::kSuccessPerformed));
+  ReConfigChunk response_reconfig(builder.Build());
+
+  EXPECT_CALL(producer_, CommitResetStreams()).Times(1);
+  EXPECT_CALL(producer_, RollbackResetStreams()).Times(0);
+
+  // Processing a response shouldn't result in sending anything.
+  EXPECT_CALL(callbacks_, OnError).Times(0);
+  EXPECT_CALL(callbacks_, SendPacketWithStatus).Times(0);
+  handler_->HandleReConfig(std::move(response_reconfig));
+
+  // Response has been processed. A new request can be sent.
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+      .WillOnce(Return(std::vector<StreamID>({StreamID(41), StreamID(43)})));
+
+  absl::optional<ReConfigChunk> reconfig2 = handler_->MakeStreamResetRequest();
+  ASSERT_TRUE(reconfig2.has_value());
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req2,
+      reconfig2->parameters().get<OutgoingSSNResetRequestParameter>());
+  EXPECT_EQ(req2.request_sequence_number(), AddTo(kMyInitialReqSn, 1));
+  EXPECT_EQ(req2.sender_last_assigned_tsn(),
+            TSN(*retransmission_queue_->next_tsn() - 1));
+  EXPECT_THAT(req2.stream_ids(),
+              UnorderedElementsAre(StreamID(41), StreamID(43)));
+}
+
+TEST_F(StreamResetHandlerTest, SendIncomingResetJustReturnsNothingPerformed) {
+  Parameters::Builder builder;
+  builder.Add(
+      IncomingSSNResetRequestParameter(kPeerInitialReqSn, {StreamID(1)}));
+
+  std::vector<ReconfigurationResponseParameter> responses =
+      HandleAndCatchResponse(ReConfigChunk(builder.Build()));
+  ASSERT_THAT(responses, SizeIs(1));
+  EXPECT_THAT(responses[0].response_sequence_number(), kPeerInitialReqSn);
+  EXPECT_THAT(responses[0].result(), ResponseResult::kSuccessNothingToDo);
+}
+
+TEST_F(StreamResetHandlerTest, SendSameRequestTwiceIsIdempotent) {
+  // Simulate that receiving the same chunk twice (due to network issues,
+  // or retransmissions, causing a RECONFIG to be re-received) is idempotent.
+  for (int i = 0; i < 2; ++i) {
+    Parameters::Builder builder;
+    builder.Add(OutgoingSSNResetRequestParameter(
+        kPeerInitialReqSn, ReconfigRequestSN(3), AddTo(kPeerInitialTsn, 1),
+        {StreamID(1)}));
+
+    std::vector<ReconfigurationResponseParameter> responses1 =
+        HandleAndCatchResponse(ReConfigChunk(builder.Build()));
+    EXPECT_THAT(responses1, SizeIs(1));
+    EXPECT_EQ(responses1[0].result(), ResponseResult::kInProgress);
+  }
+}
+
+TEST_F(StreamResetHandlerTest,
+       HandoverIsAllowedOnlyWhenNoStreamIsBeingOrWillBeReset) {
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(42)));
+  handler_->ResetStreams(std::vector<StreamID>({StreamID(42)}));
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_EQ(
+      handler_->GetHandoverReadiness(),
+      HandoverReadinessStatus(HandoverUnreadinessReason::kPendingStreamReset));
+
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset())
+      .WillOnce(Return(true))
+      .WillOnce(Return(false));
+  EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+      .WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
+
+  ASSERT_TRUE(handler_->MakeStreamResetRequest().has_value());
+  EXPECT_EQ(handler_->GetHandoverReadiness(),
+            HandoverReadinessStatus(
+                HandoverUnreadinessReason::kPendingStreamResetRequest));
+
+  // Reset more streams while the request is in-flight.
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(41)));
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(43)));
+  StreamID stream_ids[] = {StreamID(41), StreamID(43)};
+  handler_->ResetStreams(stream_ids);
+
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_EQ(handler_->GetHandoverReadiness(),
+            HandoverReadinessStatus()
+                .Add(HandoverUnreadinessReason::kPendingStreamResetRequest)
+                .Add(HandoverUnreadinessReason::kPendingStreamReset));
+
+  // Processing a response to first request.
+  EXPECT_CALL(producer_, CommitResetStreams()).Times(1);
+  handler_->HandleReConfig(
+      ReConfigChunk(Parameters::Builder()
+                        .Add(ReconfigurationResponseParameter(
+                            kMyInitialReqSn, ResponseResult::kSuccessPerformed))
+                        .Build()));
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_EQ(
+      handler_->GetHandoverReadiness(),
+      HandoverReadinessStatus(HandoverUnreadinessReason::kPendingStreamReset));
+
+  // Second request can be sent.
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset())
+      .WillOnce(Return(true))
+      .WillOnce(Return(false));
+  EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+      .WillOnce(Return(std::vector<StreamID>({StreamID(41), StreamID(43)})));
+
+  ASSERT_TRUE(handler_->MakeStreamResetRequest().has_value());
+  EXPECT_EQ(handler_->GetHandoverReadiness(),
+            HandoverReadinessStatus(
+                HandoverUnreadinessReason::kPendingStreamResetRequest));
+
+  // Processing a response to second request.
+  EXPECT_CALL(producer_, CommitResetStreams()).Times(1);
+  handler_->HandleReConfig(ReConfigChunk(
+      Parameters::Builder()
+          .Add(ReconfigurationResponseParameter(
+              AddTo(kMyInitialReqSn, 1), ResponseResult::kSuccessPerformed))
+          .Build()));
+
+  // Seconds response has been processed. No pending resets.
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(false));
+
+  EXPECT_TRUE(handler_->GetHandoverReadiness().IsReady());
+}
+
+TEST_F(StreamResetHandlerTest, HandoverInInitialState) {
+  PerformHandover();
+
+  EXPECT_CALL(producer_, PrepareResetStream(StreamID(42)));
+  handler_->ResetStreams(std::vector<StreamID>({StreamID(42)}));
+
+  EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+  EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+      .WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
+
+  absl::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
+  ASSERT_TRUE(reconfig.has_value());
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      OutgoingSSNResetRequestParameter req,
+      reconfig->parameters().get<OutgoingSSNResetRequestParameter>());
+
+  EXPECT_EQ(req.request_sequence_number(), kMyInitialReqSn);
+  EXPECT_EQ(req.sender_last_assigned_tsn(),
+            TSN(*retransmission_queue_->next_tsn() - 1));
+  EXPECT_THAT(req.stream_ids(), UnorderedElementsAre(StreamID(42)));
+}
+
+TEST_F(StreamResetHandlerTest, HandoverAfterHavingResetOneStream) {
+  // Reset one stream
+  {
+    EXPECT_CALL(producer_, PrepareResetStream(StreamID(42)));
+    handler_->ResetStreams(std::vector<StreamID>({StreamID(42)}));
+
+    EXPECT_CALL(producer_, HasStreamsReadyToBeReset())
+        .WillOnce(Return(true))
+        .WillOnce(Return(false));
+    EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+        .WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
+
+    ASSERT_HAS_VALUE_AND_ASSIGN(ReConfigChunk reconfig,
+                                handler_->MakeStreamResetRequest());
+    ASSERT_HAS_VALUE_AND_ASSIGN(
+        OutgoingSSNResetRequestParameter req,
+        reconfig.parameters().get<OutgoingSSNResetRequestParameter>());
+    EXPECT_EQ(req.request_sequence_number(), kMyInitialReqSn);
+    EXPECT_EQ(req.sender_last_assigned_tsn(),
+              TSN(*retransmission_queue_->next_tsn() - 1));
+    EXPECT_THAT(req.stream_ids(), UnorderedElementsAre(StreamID(42)));
+
+    EXPECT_CALL(producer_, CommitResetStreams()).Times(1);
+    handler_->HandleReConfig(
+        ReConfigChunk(Parameters::Builder()
+                          .Add(ReconfigurationResponseParameter(
+                              req.request_sequence_number(),
+                              ResponseResult::kSuccessPerformed))
+                          .Build()));
+  }
+
+  PerformHandover();
+
+  // Reset another stream after handover
+  {
+    EXPECT_CALL(producer_, PrepareResetStream(StreamID(43)));
+    handler_->ResetStreams(std::vector<StreamID>({StreamID(43)}));
+
+    EXPECT_CALL(producer_, HasStreamsReadyToBeReset()).WillOnce(Return(true));
+    EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
+        .WillOnce(Return(std::vector<StreamID>({StreamID(43)})));
+
+    ASSERT_HAS_VALUE_AND_ASSIGN(ReConfigChunk reconfig,
+                                handler_->MakeStreamResetRequest());
+    ASSERT_HAS_VALUE_AND_ASSIGN(
+        OutgoingSSNResetRequestParameter req,
+        reconfig.parameters().get<OutgoingSSNResetRequestParameter>());
+
+    EXPECT_EQ(req.request_sequence_number(),
+              ReconfigRequestSN(kMyInitialReqSn.value() + 1));
+    EXPECT_EQ(req.sender_last_assigned_tsn(),
+              TSN(*retransmission_queue_->next_tsn() - 1));
+    EXPECT_THAT(req.stream_ids(), UnorderedElementsAre(StreamID(43)));
+  }
+}
+
+TEST_F(StreamResetHandlerTest, PerformCloseAfterOneFirstFailing) {
+  // Inject a stream reset on the first expected TSN (which hasn't been seen).
+  Parameters::Builder builder;
+  builder.Add(OutgoingSSNResetRequestParameter(
+      kPeerInitialReqSn, ReconfigRequestSN(3), kPeerInitialTsn, {StreamID(1)}));
+
+  // The socket is expected to say "in progress" as that TSN hasn't been seen.
+  std::vector<ReconfigurationResponseParameter> responses =
+      HandleAndCatchResponse(ReConfigChunk(builder.Build()));
+  EXPECT_THAT(responses, SizeIs(1));
+  EXPECT_EQ(responses[0].result(), ResponseResult::kInProgress);
+
+  // Let the socket receive the TSN.
+  DataGeneratorOptions opts;
+  opts.message_id = MID(0);
+  reasm_->Add(kPeerInitialTsn, gen_.Ordered({1, 2, 3, 4}, "BE", opts));
+  reasm_->MaybeResetStreamsDeferred(kPeerInitialTsn);
+  data_tracker_->Observe(kPeerInitialTsn);
+
+  // And emulate that time has passed, and the peer retries the stream reset,
+  // but now with an incremented request sequence number.
+  Parameters::Builder builder2;
+  builder2.Add(OutgoingSSNResetRequestParameter(
+      ReconfigRequestSN(*kPeerInitialReqSn + 1), ReconfigRequestSN(3),
+      kPeerInitialTsn, {StreamID(1)}));
+
+  // This is supposed to be handled well.
+  std::vector<ReconfigurationResponseParameter> responses2 =
+      HandleAndCatchResponse(ReConfigChunk(builder2.Build()));
+  EXPECT_THAT(responses2, SizeIs(1));
+  EXPECT_EQ(responses2[0].result(), ResponseResult::kSuccessPerformed);
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/transmission_control_block.cc b/third_party/libwebrtc/net/dcsctp/socket/transmission_control_block.cc
new file mode 100644
index 0000000000..1dcf394813
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/transmission_control_block.cc
@@ -0,0 +1,320 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/socket/transmission_control_block.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/capabilities.h"
+#include "net/dcsctp/socket/stream_reset_handler.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "net/dcsctp/tx/retransmission_timeout.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+
+TransmissionControlBlock::TransmissionControlBlock(
+    TimerManager& timer_manager,
+    absl::string_view log_prefix,
+    const DcSctpOptions& options,
+    const Capabilities& capabilities,
+    DcSctpSocketCallbacks& callbacks,
+    SendQueue& send_queue,
+    VerificationTag my_verification_tag,
+    TSN my_initial_tsn,
+    VerificationTag peer_verification_tag,
+    TSN peer_initial_tsn,
+    size_t a_rwnd,
+    TieTag tie_tag,
+    PacketSender& packet_sender,
+    std::function<bool()> is_connection_established)
+    : log_prefix_(log_prefix),
+      options_(options),
+      timer_manager_(timer_manager),
+      capabilities_(capabilities),
+      callbacks_(callbacks),
+      t3_rtx_(timer_manager_.CreateTimer(
+          "t3-rtx",
+          absl::bind_front(&TransmissionControlBlock::OnRtxTimerExpiry, this),
+          TimerOptions(options.rto_initial,
+                       TimerBackoffAlgorithm::kExponential,
+                       /*max_restarts=*/absl::nullopt,
+                       options.max_timer_backoff_duration))),
+      delayed_ack_timer_(timer_manager_.CreateTimer(
+          "delayed-ack",
+          absl::bind_front(&TransmissionControlBlock::OnDelayedAckTimerExpiry,
+                           this),
+          TimerOptions(options.delayed_ack_max_timeout,
+                       TimerBackoffAlgorithm::kExponential,
+                       /*max_restarts=*/0,
+                       /*max_backoff_duration=*/absl::nullopt,
+                       webrtc::TaskQueueBase::DelayPrecision::kHigh))),
+      my_verification_tag_(my_verification_tag),
+      my_initial_tsn_(my_initial_tsn),
+      peer_verification_tag_(peer_verification_tag),
+      peer_initial_tsn_(peer_initial_tsn),
+      tie_tag_(tie_tag),
+      is_connection_established_(std::move(is_connection_established)),
+      packet_sender_(packet_sender),
+      rto_(options),
+      tx_error_counter_(log_prefix, options),
+      data_tracker_(log_prefix, delayed_ack_timer_.get(), peer_initial_tsn),
+      reassembly_queue_(log_prefix,
+                        peer_initial_tsn,
+                        options.max_receiver_window_buffer_size,
+                        capabilities.message_interleaving),
+      retransmission_queue_(
+          log_prefix,
+          &callbacks_,
+          my_initial_tsn,
+          a_rwnd,
+          send_queue,
+          absl::bind_front(&TransmissionControlBlock::ObserveRTT, this),
+          [this]() { tx_error_counter_.Clear(); },
+          *t3_rtx_,
+          options,
+          capabilities.partial_reliability,
+          capabilities.message_interleaving),
+      stream_reset_handler_(log_prefix,
+                            this,
+                            &timer_manager,
+                            &data_tracker_,
+                            &reassembly_queue_,
+                            &retransmission_queue_),
+      heartbeat_handler_(log_prefix, options, this, &timer_manager_) {
+  send_queue.EnableMessageInterleaving(capabilities.message_interleaving);
+}
+
+void TransmissionControlBlock::ObserveRTT(DurationMs rtt) {
+  DurationMs prev_rto = rto_.rto();
+  rto_.ObserveRTT(rtt);
+  RTC_DLOG(LS_VERBOSE) << log_prefix_ << "new rtt=" << *rtt
+                       << ", srtt=" << *rto_.srtt() << ", rto=" << *rto_.rto()
+                       << " (" << *prev_rto << ")";
+  t3_rtx_->set_duration(rto_.rto());
+
+  DurationMs delayed_ack_tmo =
+      std::min(rto_.rto() * 0.5, options_.delayed_ack_max_timeout);
+  delayed_ack_timer_->set_duration(delayed_ack_tmo);
+}
+
+absl::optional<DurationMs> TransmissionControlBlock::OnRtxTimerExpiry() {
+  TimeMs now = callbacks_.TimeMillis();
+  RTC_DLOG(LS_INFO) << log_prefix_ << "Timer " << t3_rtx_->name()
+                    << " has expired";
+  if (cookie_echo_chunk_.has_value()) {
+    // In the COOKIE_ECHO state, let the T1-COOKIE timer trigger
+    // retransmissions, to avoid having two timers doing that.
+    RTC_DLOG(LS_VERBOSE) << "Not retransmitting as T1-cookie is active.";
+  } else {
+    if (IncrementTxErrorCounter("t3-rtx expired")) {
+      retransmission_queue_.HandleT3RtxTimerExpiry();
+      SendBufferedPackets(now);
+    }
+  }
+  return absl::nullopt;
+}
+
+absl::optional<DurationMs> TransmissionControlBlock::OnDelayedAckTimerExpiry() {
+  data_tracker_.HandleDelayedAckTimerExpiry();
+  MaybeSendSack();
+  return absl::nullopt;
+}
+
+void TransmissionControlBlock::MaybeSendSack() {
+  if (data_tracker_.ShouldSendAck(/*also_if_delayed=*/false)) {
+    SctpPacket::Builder builder = PacketBuilder();
+    builder.Add(
+        data_tracker_.CreateSelectiveAck(reassembly_queue_.remaining_bytes()));
+    Send(builder);
+  }
+}
+
+void TransmissionControlBlock::MaybeSendForwardTsn(SctpPacket::Builder& builder,
+                                                   TimeMs now) {
+  if (now >= limit_forward_tsn_until_ &&
+      retransmission_queue_.ShouldSendForwardTsn(now)) {
+    if (capabilities_.message_interleaving) {
+      builder.Add(retransmission_queue_.CreateIForwardTsn());
+    } else {
+      builder.Add(retransmission_queue_.CreateForwardTsn());
+    }
+    packet_sender_.Send(builder);
+    // https://datatracker.ietf.org/doc/html/rfc3758
+    // "IMPLEMENTATION NOTE: An implementation may wish to limit the number of
+    // duplicate FORWARD TSN chunks it sends by ... waiting a full RTT before
+    // sending a duplicate FORWARD TSN."
+    // "Any delay applied to the sending of FORWARD TSN chunk SHOULD NOT exceed
+    // 200ms and MUST NOT exceed 500ms".
+    limit_forward_tsn_until_ = now + std::min(DurationMs(200), rto_.srtt());
+  }
+}
+
+void TransmissionControlBlock::MaybeSendFastRetransmit() {
+  if (!retransmission_queue_.has_data_to_be_fast_retransmitted()) {
+    return;
+  }
+
+  // https://datatracker.ietf.org/doc/html/rfc4960#section-7.2.4
+  // "Determine how many of the earliest (i.e., lowest TSN) DATA chunks marked
+  // for retransmission will fit into a single packet, subject to constraint of
+  // the path MTU of the destination transport address to which the packet is
+  // being sent.  Call this value K. Retransmit those K DATA chunks in a single
+  // packet.  When a Fast Retransmit is being performed, the sender SHOULD
+  // ignore the value of cwnd and SHOULD NOT delay retransmission for this
+  // single packet."
+
+  SctpPacket::Builder builder(peer_verification_tag_, options_);
+  auto chunks = retransmission_queue_.GetChunksForFastRetransmit(
+      builder.bytes_remaining());
+  for (auto& [tsn, data] : chunks) {
+    if (capabilities_.message_interleaving) {
+      builder.Add(IDataChunk(tsn, std::move(data), false));
+    } else {
+      builder.Add(DataChunk(tsn, std::move(data), false));
+    }
+  }
+  packet_sender_.Send(builder);
+}
+
+void TransmissionControlBlock::SendBufferedPackets(SctpPacket::Builder& builder,
+                                                   TimeMs now) {
+  for (int packet_idx = 0;
+       packet_idx < options_.max_burst && retransmission_queue_.can_send_data();
+       ++packet_idx) {
+    // Only add control chunks to the first packet that is sent, if sending
+    // multiple packets in one go (as allowed by the congestion window).
+    if (packet_idx == 0) {
+      if (cookie_echo_chunk_.has_value()) {
+        // https://tools.ietf.org/html/rfc4960#section-5.1
+        // "The COOKIE ECHO chunk can be bundled with any pending outbound DATA
+        // chunks, but it MUST be the first chunk in the packet..."
+        RTC_DCHECK(builder.empty());
+        builder.Add(*cookie_echo_chunk_);
+      }
+
+      // https://tools.ietf.org/html/rfc4960#section-6
+      // "Before an endpoint transmits a DATA chunk, if any received DATA
+      // chunks have not been acknowledged (e.g., due to delayed ack), the
+      // sender should create a SACK and bundle it with the outbound DATA chunk,
+      // as long as the size of the final SCTP packet does not exceed the
+      // current MTU."
+      if (data_tracker_.ShouldSendAck(/*also_if_delayed=*/true)) {
+        builder.Add(data_tracker_.CreateSelectiveAck(
+            reassembly_queue_.remaining_bytes()));
+      }
+      MaybeSendForwardTsn(builder, now);
+      absl::optional<ReConfigChunk> reconfig =
+          stream_reset_handler_.MakeStreamResetRequest();
+      if (reconfig.has_value()) {
+        builder.Add(*reconfig);
+      }
+    }
+
+    auto chunks =
+        retransmission_queue_.GetChunksToSend(now, builder.bytes_remaining());
+    for (auto& [tsn, data] : chunks) {
+      if (capabilities_.message_interleaving) {
+        builder.Add(IDataChunk(tsn, std::move(data), false));
+      } else {
+        builder.Add(DataChunk(tsn, std::move(data), false));
+      }
+    }
+
+    if (!packet_sender_.Send(builder)) {
+      break;
+    }
+
+    if (cookie_echo_chunk_.has_value()) {
+      // https://tools.ietf.org/html/rfc4960#section-5.1
+      // "...  until the COOKIE ACK is returned the sender MUST NOT send any
+      // other packets to the peer."
+      break;
+    }
+  }
+}
+
+std::string TransmissionControlBlock::ToString() const {
+  rtc::StringBuilder sb;
+
+  sb.AppendFormat(
+      "verification_tag=%08x, last_cumulative_ack=%u, capabilities=",
+      *peer_verification_tag_, *data_tracker_.last_cumulative_acked_tsn());
+
+  if (capabilities_.partial_reliability) {
+    sb << "PR,";
+  }
+  if (capabilities_.message_interleaving) {
+    sb << "IL,";
+  }
+  if (capabilities_.reconfig) {
+    sb << "Reconfig,";
+  }
+  sb << " max_in=" << capabilities_.negotiated_maximum_incoming_streams;
+  sb << " max_out=" << capabilities_.negotiated_maximum_outgoing_streams;
+
+  return sb.Release();
+}
+
+HandoverReadinessStatus TransmissionControlBlock::GetHandoverReadiness() const {
+  HandoverReadinessStatus status;
+  status.Add(data_tracker_.GetHandoverReadiness());
+  status.Add(stream_reset_handler_.GetHandoverReadiness());
+  status.Add(reassembly_queue_.GetHandoverReadiness());
+  status.Add(retransmission_queue_.GetHandoverReadiness());
+  return status;
+}
+
+void TransmissionControlBlock::AddHandoverState(
+    DcSctpSocketHandoverState& state) {
+  state.capabilities.partial_reliability = capabilities_.partial_reliability;
+  state.capabilities.message_interleaving = capabilities_.message_interleaving;
+  state.capabilities.reconfig = capabilities_.reconfig;
+  state.capabilities.negotiated_maximum_incoming_streams =
+      capabilities_.negotiated_maximum_incoming_streams;
+  state.capabilities.negotiated_maximum_outgoing_streams =
+      capabilities_.negotiated_maximum_outgoing_streams;
+
+  state.my_verification_tag = my_verification_tag().value();
+  state.peer_verification_tag = peer_verification_tag().value();
+  state.my_initial_tsn = my_initial_tsn().value();
+  state.peer_initial_tsn = peer_initial_tsn().value();
+  state.tie_tag = tie_tag().value();
+
+  data_tracker_.AddHandoverState(state);
+  stream_reset_handler_.AddHandoverState(state);
+  reassembly_queue_.AddHandoverState(state);
+  retransmission_queue_.AddHandoverState(state);
+}
+
+void TransmissionControlBlock::RestoreFromState(
+    const DcSctpSocketHandoverState& state) {
+  data_tracker_.RestoreFromState(state);
+  retransmission_queue_.RestoreFromState(state);
+  reassembly_queue_.RestoreFromState(state);
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/socket/transmission_control_block.h b/third_party/libwebrtc/net/dcsctp/socket/transmission_control_block.h
new file mode 100644
index 0000000000..8e0e9a3ec5
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/socket/transmission_control_block.h
@@ -0,0 +1,193 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_TRANSMISSION_CONTROL_BLOCK_H_
+#define NET_DCSCTP_SOCKET_TRANSMISSION_CONTROL_BLOCK_H_
+
+#include <cstdint>
+#include <functional>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/bind_front.h"
+#include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/capabilities.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/socket/heartbeat_handler.h"
+#include "net/dcsctp/socket/packet_sender.h"
+#include "net/dcsctp/socket/stream_reset_handler.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_error_counter.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "net/dcsctp/tx/retransmission_timeout.h"
+#include "net/dcsctp/tx/send_queue.h"
+
+namespace dcsctp {
+
+// The TransmissionControlBlock (TCB) represents an open connection to a peer,
+// and holds all the resources for that. If the connection is e.g. shutdown,
+// closed or restarted, this object will be deleted and/or replaced.
+class TransmissionControlBlock : public Context {
+ public:
+  TransmissionControlBlock(TimerManager& timer_manager,
+                           absl::string_view log_prefix,
+                           const DcSctpOptions& options,
+                           const Capabilities& capabilities,
+                           DcSctpSocketCallbacks& callbacks,
+                           SendQueue& send_queue,
+                           VerificationTag my_verification_tag,
+                           TSN my_initial_tsn,
+                           VerificationTag peer_verification_tag,
+                           TSN peer_initial_tsn,
+                           size_t a_rwnd,
+                           TieTag tie_tag,
+                           PacketSender& packet_sender,
+                           std::function<bool()> is_connection_established);
+
+  // Implementation of `Context`.
+  bool is_connection_established() const override {
+    return is_connection_established_();
+  }
+  TSN my_initial_tsn() const override { return my_initial_tsn_; }
+  TSN peer_initial_tsn() const override { return peer_initial_tsn_; }
+  DcSctpSocketCallbacks& callbacks() const override { return callbacks_; }
+  void ObserveRTT(DurationMs rtt) override;
+  DurationMs current_rto() const override { return rto_.rto(); }
+  bool IncrementTxErrorCounter(absl::string_view reason) override {
+    return tx_error_counter_.Increment(reason);
+  }
+  void ClearTxErrorCounter() override { tx_error_counter_.Clear(); }
+  SctpPacket::Builder PacketBuilder() const override {
+    return SctpPacket::Builder(peer_verification_tag_, options_);
+  }
+  bool HasTooManyTxErrors() const override {
+    return tx_error_counter_.IsExhausted();
+  }
+  void Send(SctpPacket::Builder& builder) override {
+    packet_sender_.Send(builder);
+  }
+
+  // Other accessors
+  DataTracker& data_tracker() { return data_tracker_; }
+  ReassemblyQueue& reassembly_queue() { return reassembly_queue_; }
+  RetransmissionQueue& retransmission_queue() { return retransmission_queue_; }
+  StreamResetHandler& stream_reset_handler() { return stream_reset_handler_; }
+  HeartbeatHandler& heartbeat_handler() { return heartbeat_handler_; }
+  size_t cwnd() const { return retransmission_queue_.cwnd(); }
+  DurationMs current_srtt() const { return rto_.srtt(); }
+
+  // Returns this socket's verification tag, set in all packet headers.
+  VerificationTag my_verification_tag() const { return my_verification_tag_; }
+  // Returns the peer's verification tag, which should be in received packets.
+  VerificationTag peer_verification_tag() const {
+    return peer_verification_tag_;
+  }
+  // All negotiated supported capabilities.
+  const Capabilities& capabilities() const { return capabilities_; }
+  // A 64-bit tie-tag, used to e.g. detect reconnections.
+  TieTag tie_tag() const { return tie_tag_; }
+
+  // Sends a SACK, if there is a need to.
+  void MaybeSendSack();
+
+  // Sends a FORWARD-TSN, if it is needed and allowed (rate-limited).
+  void MaybeSendForwardTsn(SctpPacket::Builder& builder, TimeMs now);
+
+  // Will be set while the socket is in kCookieEcho state. In this state, there
+  // can only be a single packet outstanding, and it must contain the COOKIE
+  // ECHO chunk as the first chunk in that packet, until the COOKIE ACK has been
+  // received, which will make the socket call `ClearCookieEchoChunk`.
+  void SetCookieEchoChunk(CookieEchoChunk chunk) {
+    cookie_echo_chunk_ = std::move(chunk);
+  }
+
+  // Called when the COOKIE ACK chunk has been received, to allow further
+  // packets to be sent.
+  void ClearCookieEchoChunk() { cookie_echo_chunk_ = absl::nullopt; }
+
+  bool has_cookie_echo_chunk() const { return cookie_echo_chunk_.has_value(); }
+
+  void MaybeSendFastRetransmit();
+
+  // Fills `builder` (which may already be filled with control chunks) with
+  // other control and data chunks, and sends packets as much as can be
+  // allowed by the congestion control algorithm.
+  void SendBufferedPackets(SctpPacket::Builder& builder, TimeMs now);
+
+  // As above, but without passing in a builder. If `cookie_echo_chunk_` is
+  // present, then only one packet will be sent, with this chunk as the first
+  // chunk.
+  void SendBufferedPackets(TimeMs now) {
+    SctpPacket::Builder builder(peer_verification_tag_, options_);
+    SendBufferedPackets(builder, now);
+  }
+
+  // Returns a textual representation of this object, for logging.
+  std::string ToString() const;
+
+  HandoverReadinessStatus GetHandoverReadiness() const;
+
+  void AddHandoverState(DcSctpSocketHandoverState& state);
+  void RestoreFromState(const DcSctpSocketHandoverState& handover_state);
+
+ private:
+  // Will be called when the retransmission timer (t3-rtx) expires.
+  absl::optional<DurationMs> OnRtxTimerExpiry();
+  // Will be called when the delayed ack timer expires.
+  absl::optional<DurationMs> OnDelayedAckTimerExpiry();
+
+  const std::string log_prefix_;
+  const DcSctpOptions options_;
+  TimerManager& timer_manager_;
+  // Negotiated capabilities that both peers support.
+  const Capabilities capabilities_;
+  DcSctpSocketCallbacks& callbacks_;
+  // The data retransmission timer, called t3-rtx in SCTP.
+  const std::unique_ptr<Timer> t3_rtx_;
+  // Delayed ack timer, which triggers when acks should be sent (when delayed).
+  const std::unique_ptr<Timer> delayed_ack_timer_;
+  const VerificationTag my_verification_tag_;
+  const TSN my_initial_tsn_;
+  const VerificationTag peer_verification_tag_;
+  const TSN peer_initial_tsn_;
+  // Nonce, used to detect reconnections.
+  const TieTag tie_tag_;
+  const std::function<bool()> is_connection_established_;
+  PacketSender& packet_sender_;
+  // Rate limiting of FORWARD-TSN. Next can be sent at or after this timestamp.
+  TimeMs limit_forward_tsn_until_ = TimeMs(0);
+
+  RetransmissionTimeout rto_;
+  RetransmissionErrorCounter tx_error_counter_;
+  DataTracker data_tracker_;
+  ReassemblyQueue reassembly_queue_;
+  RetransmissionQueue retransmission_queue_;
+  StreamResetHandler stream_reset_handler_;
+  HeartbeatHandler heartbeat_handler_;
+
+  // Only valid when the socket state == State::kCookieEchoed. In this state,
+  // the socket must wait for COOKIE ACK to continue sending any packets (not
+  // including a COOKIE ECHO). So if `cookie_echo_chunk_` is present, the
+  // SendBufferedChunks will always only just send one packet, with this chunk
+  // as the first chunk in the packet.
+  absl::optional<CookieEchoChunk> cookie_echo_chunk_ = absl::nullopt;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_TRANSMISSION_CONTROL_BLOCK_H_
diff --git a/third_party/libwebrtc/net/dcsctp/testing/BUILD.gn b/third_party/libwebrtc/net/dcsctp/testing/BUILD.gn
new file mode 100644
index 0000000000..7e005a1f0c
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/testing/BUILD.gn
@@ -0,0 +1,33 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("testing_macros") {
+  testonly = true
+  sources = [ "testing_macros.h" ]
+}
+
+rtc_library("data_generator") {
+  testonly = true
+  deps = [
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../common:internal_types",
+    "../packet:data",
+    "../public:types",
+  ]
+  sources = [
+    "data_generator.cc",
+    "data_generator.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
diff --git a/third_party/libwebrtc/net/dcsctp/testing/data_generator.cc b/third_party/libwebrtc/net/dcsctp/testing/data_generator.cc
new file mode 100644
index 0000000000..e4f9f91384
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/testing/data_generator.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/testing/data_generator.h"
+
+#include <cstdint>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+constexpr PPID kPpid = PPID(53);
+
+Data DataGenerator::Ordered(std::vector<uint8_t> payload,
+                            absl::string_view flags,
+                            const DataGeneratorOptions opts) {
+  Data::IsBeginning is_beginning(flags.find('B') != std::string::npos);
+  Data::IsEnd is_end(flags.find('E') != std::string::npos);
+
+  if (is_beginning) {
+    fsn_ = FSN(0);
+  } else {
+    fsn_ = FSN(*fsn_ + 1);
+  }
+  MID message_id = opts.message_id.value_or(message_id_);
+  Data ret = Data(opts.stream_id, SSN(static_cast<uint16_t>(*message_id)),
+                  message_id, fsn_, opts.ppid, std::move(payload), is_beginning,
+                  is_end, IsUnordered(false));
+
+  if (is_end) {
+    message_id_ = MID(*message_id + 1);
+  }
+  return ret;
+}
+
+Data DataGenerator::Unordered(std::vector<uint8_t> payload,
+                              absl::string_view flags,
+                              const DataGeneratorOptions opts) {
+  Data::IsBeginning is_beginning(flags.find('B') != std::string::npos);
+  Data::IsEnd is_end(flags.find('E') != std::string::npos);
+
+  if (is_beginning) {
+    fsn_ = FSN(0);
+  } else {
+    fsn_ = FSN(*fsn_ + 1);
+  }
+  MID message_id = opts.message_id.value_or(message_id_);
+  Data ret = Data(opts.stream_id, SSN(0), message_id, fsn_, kPpid,
+                  std::move(payload), is_beginning, is_end, IsUnordered(true));
+  if (is_end) {
+    message_id_ = MID(*message_id + 1);
+  }
+  return ret;
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/testing/data_generator.h b/third_party/libwebrtc/net/dcsctp/testing/data_generator.h
new file mode 100644
index 0000000000..f917c740a7
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/testing/data_generator.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TESTING_DATA_GENERATOR_H_
+#define NET_DCSCTP_TESTING_DATA_GENERATOR_H_
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/data.h"
+
+namespace dcsctp {
+
+struct DataGeneratorOptions {
+  StreamID stream_id = StreamID(1);
+  absl::optional<MID> message_id = absl::nullopt;
+  PPID ppid = PPID(53);
+};
+
+// Generates Data with correct sequence numbers, and used only in unit tests.
+class DataGenerator {
+ public:
+  explicit DataGenerator(MID start_message_id = MID(0))
+      : message_id_(start_message_id) {}
+
+  // Generates ordered "data" with the provided `payload` and flags, which can
+  // contain "B" for setting the "is_beginning" flag, and/or "E" for setting the
+  // "is_end" flag.
+  Data Ordered(std::vector<uint8_t> payload,
+               absl::string_view flags = "",
+               DataGeneratorOptions opts = {});
+
+  // Generates unordered "data" with the provided `payload` and flags, which can
+  // contain "B" for setting the "is_beginning" flag, and/or "E" for setting the
+  // "is_end" flag.
+  Data Unordered(std::vector<uint8_t> payload,
+                 absl::string_view flags = "",
+                 DataGeneratorOptions opts = {});
+
+  // Resets the Message ID identifier - simulating a "stream reset".
+  void ResetStream() { message_id_ = MID(0); }
+
+ private:
+  MID message_id_;
+  FSN fsn_ = FSN(0);
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TESTING_DATA_GENERATOR_H_
diff --git a/third_party/libwebrtc/net/dcsctp/testing/testing_macros.h b/third_party/libwebrtc/net/dcsctp/testing/testing_macros.h
new file mode 100644
index 0000000000..5cbdfffdce
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/testing/testing_macros.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TESTING_TESTING_MACROS_H_
+#define NET_DCSCTP_TESTING_TESTING_MACROS_H_
+
+#include <utility>
+
+namespace dcsctp {
+
+#define DCSCTP_CONCAT_INNER_(x, y) x##y
+#define DCSCTP_CONCAT_(x, y) DCSCTP_CONCAT_INNER_(x, y)
+
+// Similar to ASSERT_OK_AND_ASSIGN, this works with an absl::optional<> instead
+// of an absl::StatusOr<>.
+#define ASSERT_HAS_VALUE_AND_ASSIGN(lhs, rexpr)                     \
+  auto DCSCTP_CONCAT_(tmp_opt_val__, __LINE__) = rexpr;             \
+  ASSERT_TRUE(DCSCTP_CONCAT_(tmp_opt_val__, __LINE__).has_value()); \
+  lhs = *std::move(DCSCTP_CONCAT_(tmp_opt_val__, __LINE__));
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TESTING_TESTING_MACROS_H_
diff --git a/third_party/libwebrtc/net/dcsctp/timer/BUILD.gn b/third_party/libwebrtc/net/dcsctp/timer/BUILD.gn
new file mode 100644
index 0000000000..d3be1ec872
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/timer/BUILD.gn
@@ -0,0 +1,74 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("timer") {
+  deps = [
+    "../../../api:array_view",
+    "../../../api/task_queue:task_queue",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:strong_alias",
+    "../../../rtc_base/containers:flat_map",
+    "../../../rtc_base/containers:flat_set",
+    "../public:socket",
+    "../public:types",
+  ]
+  sources = [
+    "fake_timeout.h",
+    "timer.cc",
+    "timer.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("task_queue_timeout") {
+  deps = [
+    "../../../api:array_view",
+    "../../../api/task_queue:pending_task_safety_flag",
+    "../../../api/task_queue:task_queue",
+    "../../../api/units:time_delta",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../public:socket",
+    "../public:types",
+  ]
+  sources = [
+    "task_queue_timeout.cc",
+    "task_queue_timeout.h",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("dcsctp_timer_unittests") {
+    testonly = true
+
+    defines = []
+    deps = [
+      ":task_queue_timeout",
+      ":timer",
+      "../../../api:array_view",
+      "../../../api/task_queue:task_queue",
+      "../../../api/task_queue/test:mock_task_queue_base",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:gunit_helpers",
+      "../../../test:test_support",
+      "../../../test/time_controller:time_controller",
+      "../public:socket",
+    ]
+    sources = [
+      "task_queue_timeout_test.cc",
+      "timer_test.cc",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+}
diff --git a/third_party/libwebrtc/net/dcsctp/timer/fake_timeout.h b/third_party/libwebrtc/net/dcsctp/timer/fake_timeout.h
new file mode 100644
index 0000000000..74ffe5af29
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/timer/fake_timeout.h
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TIMER_FAKE_TIMEOUT_H_
+#define NET_DCSCTP_TIMER_FAKE_TIMEOUT_H_
+
+#include <cstdint>
+#include <functional>
+#include <limits>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/public/timeout.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/containers/flat_set.h"
+
+namespace dcsctp {
+
+// A timeout used in tests.
+class FakeTimeout : public Timeout {
+ public:
+  FakeTimeout(std::function<TimeMs()> get_time,
+              std::function<void(FakeTimeout*)> on_delete)
+      : get_time_(std::move(get_time)), on_delete_(std::move(on_delete)) {}
+
+  ~FakeTimeout() override { on_delete_(this); }
+
+  void Start(DurationMs duration_ms, TimeoutID timeout_id) override {
+    RTC_DCHECK(expiry_ == TimeMs::InfiniteFuture());
+    timeout_id_ = timeout_id;
+    expiry_ = get_time_() + duration_ms;
+  }
+  void Stop() override {
+    RTC_DCHECK(expiry_ != TimeMs::InfiniteFuture());
+    expiry_ = TimeMs::InfiniteFuture();
+  }
+
+  bool EvaluateHasExpired(TimeMs now) {
+    if (now >= expiry_) {
+      expiry_ = TimeMs::InfiniteFuture();
+      return true;
+    }
+    return false;
+  }
+
+  TimeoutID timeout_id() const { return timeout_id_; }
+
+ private:
+  const std::function<TimeMs()> get_time_;
+  const std::function<void(FakeTimeout*)> on_delete_;
+
+  TimeoutID timeout_id_ = TimeoutID(0);
+  TimeMs expiry_ = TimeMs::InfiniteFuture();
+};
+
+class FakeTimeoutManager {
+ public:
+  // The `get_time` function must return the current time, relative to any
+  // epoch.
+  explicit FakeTimeoutManager(std::function<TimeMs()> get_time)
+      : get_time_(std::move(get_time)) {}
+
+  std::unique_ptr<FakeTimeout> CreateTimeout() {
+    auto timer = std::make_unique<FakeTimeout>(
+        get_time_, [this](FakeTimeout* timer) { timers_.erase(timer); });
+    timers_.insert(timer.get());
+    return timer;
+  }
+  std::unique_ptr<FakeTimeout> CreateTimeout(
+      webrtc::TaskQueueBase::DelayPrecision precision) {
+    // FakeTimeout does not support implement |precision|.
+    return CreateTimeout();
+  }
+
+  // NOTE: This can't return a vector, as calling EvaluateHasExpired requires
+  // calling socket->HandleTimeout directly afterwards, as the owning Timer
+  // still believes it's running, and it needs to be updated to set
+  // Timer::is_running_ to false before you operate on the Timer or Timeout
+  // again.
+  absl::optional<TimeoutID> GetNextExpiredTimeout() {
+    TimeMs now = get_time_();
+    std::vector<TimeoutID> expired_timers;
+    for (auto& timer : timers_) {
+      if (timer->EvaluateHasExpired(now)) {
+        return timer->timeout_id();
+      }
+    }
+    return absl::nullopt;
+  }
+
+ private:
+  const std::function<TimeMs()> get_time_;
+  webrtc::flat_set<FakeTimeout*> timers_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TIMER_FAKE_TIMEOUT_H_
diff --git a/third_party/libwebrtc/net/dcsctp/timer/task_queue_timeout.cc b/third_party/libwebrtc/net/dcsctp/timer/task_queue_timeout.cc
new file mode 100644
index 0000000000..6c43640d39
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/timer/task_queue_timeout.cc
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/timer/task_queue_timeout.h"
+
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+
+TaskQueueTimeoutFactory::TaskQueueTimeout::TaskQueueTimeout(
+    TaskQueueTimeoutFactory& parent,
+    webrtc::TaskQueueBase::DelayPrecision precision)
+    : parent_(parent),
+      precision_(precision),
+      pending_task_safety_flag_(webrtc::PendingTaskSafetyFlag::Create()) {}
+
+TaskQueueTimeoutFactory::TaskQueueTimeout::~TaskQueueTimeout() {
+  RTC_DCHECK_RUN_ON(&parent_.thread_checker_);
+  pending_task_safety_flag_->SetNotAlive();
+}
+
+void TaskQueueTimeoutFactory::TaskQueueTimeout::Start(DurationMs duration_ms,
+                                                      TimeoutID timeout_id) {
+  RTC_DCHECK_RUN_ON(&parent_.thread_checker_);
+  RTC_DCHECK(timeout_expiration_ == TimeMs::InfiniteFuture());
+  timeout_expiration_ = parent_.get_time_() + duration_ms;
+  timeout_id_ = timeout_id;
+
+  if (timeout_expiration_ >= posted_task_expiration_) {
+    // There is already a running task, and it's scheduled to expire sooner than
+    // the new expiration time. Don't do anything; The `timeout_expiration_` has
+    // already been updated and if the delayed task _does_ expire and the timer
+    // hasn't been stopped, that will be noticed in the timeout handler, and the
+    // task will be re-scheduled. Most timers are stopped before they expire.
+    return;
+  }
+
+  if (posted_task_expiration_ != TimeMs::InfiniteFuture()) {
+    RTC_DLOG(LS_VERBOSE) << "New timeout duration is less than scheduled - "
+                            "ghosting old delayed task.";
+    // There is already a scheduled delayed task, but its expiration time is
+    // further away than the new expiration, so it can't be used. It will be
+    // "killed" by replacing the safety flag. This is not expected to happen
+    // especially often; Mainly when a timer did exponential backoff and
+    // later recovered.
+    pending_task_safety_flag_->SetNotAlive();
+    pending_task_safety_flag_ = webrtc::PendingTaskSafetyFlag::Create();
+  }
+
+  posted_task_expiration_ = timeout_expiration_;
+  parent_.task_queue_.PostDelayedTaskWithPrecision(
+      precision_,
+      webrtc::SafeTask(
+          pending_task_safety_flag_,
+          [timeout_id, this]() {
+            RTC_DLOG(LS_VERBOSE) << "Timout expired: " << timeout_id.value();
+            RTC_DCHECK_RUN_ON(&parent_.thread_checker_);
+            RTC_DCHECK(posted_task_expiration_ != TimeMs::InfiniteFuture());
+            posted_task_expiration_ = TimeMs::InfiniteFuture();
+
+            if (timeout_expiration_ == TimeMs::InfiniteFuture()) {
+              // The timeout was stopped before it expired. Very common.
+            } else {
+              // Note that the timeout might have been restarted, which updated
+              // `timeout_expiration_` but left the scheduled task running. So
+              // if it's not quite time to trigger the timeout yet, schedule a
+              // new delayed task with what's remaining and retry at that point
+              // in time.
+              DurationMs remaining = timeout_expiration_ - parent_.get_time_();
+              timeout_expiration_ = TimeMs::InfiniteFuture();
+              if (*remaining > 0) {
+                Start(remaining, timeout_id_);
+              } else {
+                // It has actually triggered.
+                RTC_DLOG(LS_VERBOSE)
+                    << "Timout triggered: " << timeout_id.value();
+                parent_.on_expired_(timeout_id_);
+              }
+            }
+          }),
+      webrtc::TimeDelta::Millis(duration_ms.value()));
+}
+
+void TaskQueueTimeoutFactory::TaskQueueTimeout::Stop() {
+  // As the TaskQueue doesn't support deleting a posted task, just mark the
+  // timeout as not running.
+  RTC_DCHECK_RUN_ON(&parent_.thread_checker_);
+  timeout_expiration_ = TimeMs::InfiniteFuture();
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/timer/task_queue_timeout.h b/third_party/libwebrtc/net/dcsctp/timer/task_queue_timeout.h
new file mode 100644
index 0000000000..faae14464f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/timer/task_queue_timeout.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TIMER_TASK_QUEUE_TIMEOUT_H_
+#define NET_DCSCTP_TIMER_TASK_QUEUE_TIMEOUT_H_
+
+#include <memory>
+#include <utility>
+
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/public/timeout.h"
+
+namespace dcsctp {
+
+// The TaskQueueTimeoutFactory creates `Timeout` instances, which schedules
+// itself to be triggered on the provided `task_queue`, which may be a thread,
+// an actual TaskQueue or something else which supports posting a delayed task.
+//
+// Note that each `DcSctpSocket` must have its own `TaskQueueTimeoutFactory`,
+// as the `TimeoutID` are not unique among sockets.
+//
+// This class must outlive any created Timeout that it has created. Note that
+// the `DcSctpSocket` will ensure that all Timeouts are deleted when the socket
+// is destructed, so this means that this class must outlive the `DcSctpSocket`.
+//
+// This class, and the timeouts created it, are not thread safe.
+class TaskQueueTimeoutFactory {
+ public:
+  // The `get_time` function must return the current time, relative to any
+  // epoch. Whenever a timeout expires, the `on_expired` callback will be
+  // triggered, and then the client should provided `timeout_id` to
+  // `DcSctpSocketInterface::HandleTimeout`.
+  TaskQueueTimeoutFactory(webrtc::TaskQueueBase& task_queue,
+                          std::function<TimeMs()> get_time,
+                          std::function<void(TimeoutID timeout_id)> on_expired)
+      : task_queue_(task_queue),
+        get_time_(std::move(get_time)),
+        on_expired_(std::move(on_expired)) {}
+
+  // Creates an implementation of `Timeout`.
+  std::unique_ptr<Timeout> CreateTimeout(
+      webrtc::TaskQueueBase::DelayPrecision precision =
+          webrtc::TaskQueueBase::DelayPrecision::kLow) {
+    return std::make_unique<TaskQueueTimeout>(*this, precision);
+  }
+
+ private:
+  class TaskQueueTimeout : public Timeout {
+   public:
+    TaskQueueTimeout(TaskQueueTimeoutFactory& parent,
+                     webrtc::TaskQueueBase::DelayPrecision precision);
+    ~TaskQueueTimeout();
+
+    void Start(DurationMs duration_ms, TimeoutID timeout_id) override;
+    void Stop() override;
+
+   private:
+    TaskQueueTimeoutFactory& parent_;
+    const webrtc::TaskQueueBase::DelayPrecision precision_;
+    // A safety flag to ensure that posted tasks to the task queue don't
+    // reference these object when they go out of scope. Note that this safety
+    // flag will be re-created if the scheduled-but-not-yet-expired task is not
+    // to be run. This happens when there is a posted delayed task with an
+    // expiration time _further away_ than what is now the expected expiration
+    // time. In this scenario, a new delayed task has to be posted with a
+    // shorter duration and the old task has to be forgotten.
+    rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> pending_task_safety_flag_;
+    // The time when the posted delayed task is set to expire. Will be set to
+    // the infinite future if there is no such task running.
+    TimeMs posted_task_expiration_ = TimeMs::InfiniteFuture();
+    // The time when the timeout expires. It will be set to the infinite future
+    // if the timeout is not running/not started.
+    TimeMs timeout_expiration_ = TimeMs::InfiniteFuture();
+    // The current timeout ID that will be reported when expired.
+    TimeoutID timeout_id_ = TimeoutID(0);
+  };
+
+  RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
+  webrtc::TaskQueueBase& task_queue_;
+  const std::function<TimeMs()> get_time_;
+  const std::function<void(TimeoutID)> on_expired_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TIMER_TASK_QUEUE_TIMEOUT_H_
diff --git a/third_party/libwebrtc/net/dcsctp/timer/task_queue_timeout_test.cc b/third_party/libwebrtc/net/dcsctp/timer/task_queue_timeout_test.cc
new file mode 100644
index 0000000000..f360ba7a58
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/timer/task_queue_timeout_test.cc
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/timer/task_queue_timeout.h"
+
+#include <memory>
+
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/test/mock_task_queue_base.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::_;
+using ::testing::MockFunction;
+using ::testing::NiceMock;
+
+class TaskQueueTimeoutTest : public testing::Test {
+ protected:
+  TaskQueueTimeoutTest()
+      : time_controller_(webrtc::Timestamp::Millis(1234)),
+        task_queue_(time_controller_.GetMainThread()),
+        factory_(
+            *task_queue_,
+            [this]() {
+              return TimeMs(time_controller_.GetClock()->CurrentTime().ms());
+            },
+            on_expired_.AsStdFunction()) {}
+
+  void AdvanceTime(DurationMs duration) {
+    time_controller_.AdvanceTime(webrtc::TimeDelta::Millis(*duration));
+  }
+
+  MockFunction<void(TimeoutID)> on_expired_;
+  webrtc::GlobalSimulatedTimeController time_controller_;
+
+  rtc::Thread* task_queue_;
+  TaskQueueTimeoutFactory factory_;
+};
+
+TEST_F(TaskQueueTimeoutTest, StartPostsDelayedTask) {
+  std::unique_ptr<Timeout> timeout = factory_.CreateTimeout();
+  timeout->Start(DurationMs(1000), TimeoutID(1));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTime(DurationMs(999));
+
+  EXPECT_CALL(on_expired_, Call(TimeoutID(1)));
+  AdvanceTime(DurationMs(1));
+}
+
+TEST_F(TaskQueueTimeoutTest, StopBeforeExpiringDoesntTrigger) {
+  std::unique_ptr<Timeout> timeout = factory_.CreateTimeout();
+  timeout->Start(DurationMs(1000), TimeoutID(1));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTime(DurationMs(999));
+
+  timeout->Stop();
+
+  AdvanceTime(DurationMs(1));
+  AdvanceTime(DurationMs(1000));
+}
+
+TEST_F(TaskQueueTimeoutTest, RestartPrologingTimeoutDuration) {
+  std::unique_ptr<Timeout> timeout = factory_.CreateTimeout();
+  timeout->Start(DurationMs(1000), TimeoutID(1));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTime(DurationMs(500));
+
+  timeout->Restart(DurationMs(1000), TimeoutID(2));
+
+  AdvanceTime(DurationMs(999));
+
+  EXPECT_CALL(on_expired_, Call(TimeoutID(2)));
+  AdvanceTime(DurationMs(1));
+}
+
+TEST_F(TaskQueueTimeoutTest, RestartWithShorterDurationExpiresWhenExpected) {
+  std::unique_ptr<Timeout> timeout = factory_.CreateTimeout();
+  timeout->Start(DurationMs(1000), TimeoutID(1));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTime(DurationMs(500));
+
+  timeout->Restart(DurationMs(200), TimeoutID(2));
+
+  AdvanceTime(DurationMs(199));
+
+  EXPECT_CALL(on_expired_, Call(TimeoutID(2)));
+  AdvanceTime(DurationMs(1));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTime(DurationMs(1000));
+}
+
+TEST_F(TaskQueueTimeoutTest, KilledBeforeExpired) {
+  std::unique_ptr<Timeout> timeout = factory_.CreateTimeout();
+  timeout->Start(DurationMs(1000), TimeoutID(1));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTime(DurationMs(500));
+
+  timeout = nullptr;
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTime(DurationMs(1000));
+}
+
+TEST(TaskQueueTimeoutWithMockTaskQueueTest, CanSetTimeoutPrecisionToLow) {
+  NiceMock<webrtc::MockTaskQueueBase> mock_task_queue;
+  EXPECT_CALL(mock_task_queue, PostDelayedTask(_, _));
+  TaskQueueTimeoutFactory factory(
+      mock_task_queue, []() { return TimeMs(1337); },
+      [](TimeoutID timeout_id) {});
+  std::unique_ptr<Timeout> timeout =
+      factory.CreateTimeout(webrtc::TaskQueueBase::DelayPrecision::kLow);
+  timeout->Start(DurationMs(1), TimeoutID(1));
+}
+
+TEST(TaskQueueTimeoutWithMockTaskQueueTest, CanSetTimeoutPrecisionToHigh) {
+  NiceMock<webrtc::MockTaskQueueBase> mock_task_queue;
+  EXPECT_CALL(mock_task_queue, PostDelayedHighPrecisionTask(_, _));
+  TaskQueueTimeoutFactory factory(
+      mock_task_queue, []() { return TimeMs(1337); },
+      [](TimeoutID timeout_id) {});
+  std::unique_ptr<Timeout> timeout =
+      factory.CreateTimeout(webrtc::TaskQueueBase::DelayPrecision::kHigh);
+  timeout->Start(DurationMs(1), TimeoutID(1));
+}
+
+TEST(TaskQueueTimeoutWithMockTaskQueueTest, TimeoutPrecisionIsLowByDefault) {
+  NiceMock<webrtc::MockTaskQueueBase> mock_task_queue;
+  EXPECT_CALL(mock_task_queue, PostDelayedTask(_, _));
+  TaskQueueTimeoutFactory factory(
+      mock_task_queue, []() { return TimeMs(1337); },
+      [](TimeoutID timeout_id) {});
+  std::unique_ptr<Timeout> timeout = factory.CreateTimeout();
+  timeout->Start(DurationMs(1), TimeoutID(1));
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/timer/timer.cc b/third_party/libwebrtc/net/dcsctp/timer/timer.cc
new file mode 100644
index 0000000000..bde07638a5
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/timer/timer.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/timer/timer.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "net/dcsctp/public/timeout.h"
+#include "rtc_base/checks.h"
+
+namespace dcsctp {
+namespace {
+TimeoutID MakeTimeoutId(TimerID timer_id, TimerGeneration generation) {
+  return TimeoutID(static_cast<uint64_t>(*timer_id) << 32 | *generation);
+}
+
+DurationMs GetBackoffDuration(const TimerOptions& options,
+                              DurationMs base_duration,
+                              int expiration_count) {
+  switch (options.backoff_algorithm) {
+    case TimerBackoffAlgorithm::kFixed:
+      return base_duration;
+    case TimerBackoffAlgorithm::kExponential: {
+      int32_t duration_ms = *base_duration;
+
+      while (expiration_count > 0 && duration_ms < *Timer::kMaxTimerDuration) {
+        duration_ms *= 2;
+        --expiration_count;
+
+        if (options.max_backoff_duration.has_value() &&
+            duration_ms > **options.max_backoff_duration) {
+          return *options.max_backoff_duration;
+        }
+      }
+
+      return DurationMs(std::min(duration_ms, *Timer::kMaxTimerDuration));
+    }
+  }
+}
+}  // namespace
+
+constexpr DurationMs Timer::kMaxTimerDuration;
+
+Timer::Timer(TimerID id,
+             absl::string_view name,
+             OnExpired on_expired,
+             UnregisterHandler unregister_handler,
+             std::unique_ptr<Timeout> timeout,
+             const TimerOptions& options)
+    : id_(id),
+      name_(name),
+      options_(options),
+      on_expired_(std::move(on_expired)),
+      unregister_handler_(std::move(unregister_handler)),
+      timeout_(std::move(timeout)),
+      duration_(options.duration) {}
+
+Timer::~Timer() {
+  Stop();
+  unregister_handler_();
+}
+
+void Timer::Start() {
+  expiration_count_ = 0;
+  if (!is_running()) {
+    is_running_ = true;
+    generation_ = TimerGeneration(*generation_ + 1);
+    timeout_->Start(duration_, MakeTimeoutId(id_, generation_));
+  } else {
+    // Timer was running - stop and restart it, to make it expire in `duration_`
+    // from now.
+    generation_ = TimerGeneration(*generation_ + 1);
+    timeout_->Restart(duration_, MakeTimeoutId(id_, generation_));
+  }
+}
+
+void Timer::Stop() {
+  if (is_running()) {
+    timeout_->Stop();
+    expiration_count_ = 0;
+    is_running_ = false;
+  }
+}
+
+void Timer::Trigger(TimerGeneration generation) {
+  if (is_running_ && generation == generation_) {
+    ++expiration_count_;
+    is_running_ = false;
+    if (!options_.max_restarts.has_value() ||
+        expiration_count_ <= *options_.max_restarts) {
+      // The timer should still be running after this triggers. Start a new
+      // timer. Note that it might be very quickly restarted again, if the
+      // `on_expired_` callback returns a new duration.
+      is_running_ = true;
+      DurationMs duration =
+          GetBackoffDuration(options_, duration_, expiration_count_);
+      generation_ = TimerGeneration(*generation_ + 1);
+      timeout_->Start(duration, MakeTimeoutId(id_, generation_));
+    }
+
+    absl::optional<DurationMs> new_duration = on_expired_();
+    if (new_duration.has_value() && new_duration != duration_) {
+      duration_ = new_duration.value();
+      if (is_running_) {
+        // Restart it with new duration.
+        timeout_->Stop();
+
+        DurationMs duration =
+            GetBackoffDuration(options_, duration_, expiration_count_);
+        generation_ = TimerGeneration(*generation_ + 1);
+        timeout_->Start(duration, MakeTimeoutId(id_, generation_));
+      }
+    }
+  }
+}
+
+void TimerManager::HandleTimeout(TimeoutID timeout_id) {
+  TimerID timer_id(*timeout_id >> 32);
+  TimerGeneration generation(*timeout_id);
+  auto it = timers_.find(timer_id);
+  if (it != timers_.end()) {
+    it->second->Trigger(generation);
+  }
+}
+
+std::unique_ptr<Timer> TimerManager::CreateTimer(absl::string_view name,
+                                                 Timer::OnExpired on_expired,
+                                                 const TimerOptions& options) {
+  next_id_ = TimerID(*next_id_ + 1);
+  TimerID id = next_id_;
+  // This would overflow after 4 billion timers created, which in SCTP would be
+  // after 800 million reconnections on a single socket. Ensure this will never
+  // happen.
+  RTC_CHECK_NE(*id, std::numeric_limits<uint32_t>::max());
+  std::unique_ptr<Timeout> timeout = create_timeout_(options.precision);
+  RTC_CHECK(timeout != nullptr);
+  auto timer = absl::WrapUnique(new Timer(
+      id, name, std::move(on_expired), [this, id]() { timers_.erase(id); },
+      std::move(timeout), options));
+  timers_[id] = timer.get();
+  return timer;
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/timer/timer.h b/third_party/libwebrtc/net/dcsctp/timer/timer.h
new file mode 100644
index 0000000000..31b496dc81
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/timer/timer.h
@@ -0,0 +1,212 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TIMER_TIMER_H_
+#define NET_DCSCTP_TIMER_TIMER_H_
+
+#include <stdint.h>
+
+#include <algorithm>
+#include <functional>
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/public/timeout.h"
+#include "rtc_base/strong_alias.h"
+
+namespace dcsctp {
+
+using TimerID = webrtc::StrongAlias<class TimerIDTag, uint32_t>;
+using TimerGeneration = webrtc::StrongAlias<class TimerGenerationTag, uint32_t>;
+
+enum class TimerBackoffAlgorithm {
+  // The base duration will be used for any restart.
+  kFixed,
+  // An exponential backoff is used for restarts, with a 2x multiplier, meaning
+  // that every restart will use a duration that is twice as long as the
+  // previous.
+  kExponential,
+};
+
+struct TimerOptions {
+  explicit TimerOptions(DurationMs duration)
+      : TimerOptions(duration, TimerBackoffAlgorithm::kExponential) {}
+  TimerOptions(DurationMs duration, TimerBackoffAlgorithm backoff_algorithm)
+      : TimerOptions(duration, backoff_algorithm, absl::nullopt) {}
+  TimerOptions(DurationMs duration,
+               TimerBackoffAlgorithm backoff_algorithm,
+               absl::optional<int> max_restarts)
+      : TimerOptions(duration, backoff_algorithm, max_restarts, absl::nullopt) {
+  }
+  TimerOptions(DurationMs duration,
+               TimerBackoffAlgorithm backoff_algorithm,
+               absl::optional<int> max_restarts,
+               absl::optional<DurationMs> max_backoff_duration)
+      : TimerOptions(duration,
+                     backoff_algorithm,
+                     max_restarts,
+                     max_backoff_duration,
+                     webrtc::TaskQueueBase::DelayPrecision::kLow) {}
+  TimerOptions(DurationMs duration,
+               TimerBackoffAlgorithm backoff_algorithm,
+               absl::optional<int> max_restarts,
+               absl::optional<DurationMs> max_backoff_duration,
+               webrtc::TaskQueueBase::DelayPrecision precision)
+      : duration(duration),
+        backoff_algorithm(backoff_algorithm),
+        max_restarts(max_restarts),
+        max_backoff_duration(max_backoff_duration),
+        precision(precision) {}
+
+  // The initial timer duration. Can be overridden with `set_duration`.
+  const DurationMs duration;
+  // If the duration should be increased (using exponential backoff) when it is
+  // restarted. If not set, the same duration will be used.
+  const TimerBackoffAlgorithm backoff_algorithm;
+  // The maximum number of times that the timer will be automatically restarted,
+  // or absl::nullopt if there is no limit.
+  const absl::optional<int> max_restarts;
+  // The maximum timeout value for exponential backoff.
+  const absl::optional<DurationMs> max_backoff_duration;
+  // The precision of the webrtc::TaskQueueBase used for scheduling.
+  const webrtc::TaskQueueBase::DelayPrecision precision;
+};
+
+// A high-level timer (in contrast to the low-level `Timeout` class).
+//
+// Timers are started and can be stopped or restarted. When a timer expires,
+// the provided `on_expired` callback will be triggered. A timer is
+// automatically restarted, as long as the number of restarts is below the
+// configurable `max_restarts` parameter. The `is_running` property can be
+// queried to know if it's still running after having expired.
+//
+// When a timer is restarted, it will use a configurable `backoff_algorithm` to
+// possibly adjust the duration of the next expiry. It is also possible to
+// return a new base duration (which is the duration before it's adjusted by the
+// backoff algorithm).
+class Timer {
+ public:
+  // The maximum timer duration - one day.
+  static constexpr DurationMs kMaxTimerDuration = DurationMs(24 * 3600 * 1000);
+
+  // When expired, the timer handler can optionally return a new duration which
+  // will be set as `duration` and used as base duration when the timer is
+  // restarted and as input to the backoff algorithm.
+  using OnExpired = std::function<absl::optional<DurationMs>()>;
+
+  // TimerManager will have pointers to these instances, so they must not move.
+  Timer(const Timer&) = delete;
+  Timer& operator=(const Timer&) = delete;
+
+  ~Timer();
+
+  // Starts the timer if it's stopped or restarts the timer if it's already
+  // running. The `expiration_count` will be reset.
+  void Start();
+
+  // Stops the timer. This can also be called when the timer is already stopped.
+  // The `expiration_count` will be reset.
+  void Stop();
+
+  // Sets the base duration. The actual timer duration may be larger depending
+  // on the backoff algorithm.
+  void set_duration(DurationMs duration) {
+    duration_ = std::min(duration, kMaxTimerDuration);
+  }
+
+  // Retrieves the base duration. The actual timer duration may be larger
+  // depending on the backoff algorithm.
+  DurationMs duration() const { return duration_; }
+
+  // Returns the number of times the timer has expired.
+  int expiration_count() const { return expiration_count_; }
+
+  // Returns the timer's options.
+  const TimerOptions& options() const { return options_; }
+
+  // Returns the name of the timer.
+  absl::string_view name() const { return name_; }
+
+  // Indicates if this timer is currently running.
+  bool is_running() const { return is_running_; }
+
+ private:
+  friend class TimerManager;
+  using UnregisterHandler = std::function<void()>;
+  Timer(TimerID id,
+        absl::string_view name,
+        OnExpired on_expired,
+        UnregisterHandler unregister,
+        std::unique_ptr<Timeout> timeout,
+        const TimerOptions& options);
+
+  // Called by TimerManager. Will trigger the callback and increment
+  // `expiration_count`. The timer will automatically be restarted at the
+  // duration as decided by the backoff algorithm, unless the
+  // `TimerOptions::max_restarts` has been reached and then it will be stopped
+  // and `is_running()` will return false.
+  void Trigger(TimerGeneration generation);
+
+  const TimerID id_;
+  const std::string name_;
+  const TimerOptions options_;
+  const OnExpired on_expired_;
+  const UnregisterHandler unregister_handler_;
+  const std::unique_ptr<Timeout> timeout_;
+
+  DurationMs duration_;
+
+  // Increased on each start, and is matched on Trigger, to avoid races. And by
+  // race, meaning that a timeout - which may be evaluated/expired on a
+  // different thread while this thread has stopped that timer already. Note
+  // that the entire socket is not thread-safe, so `TimerManager::HandleTimeout`
+  // is never executed concurrently with any timer starting/stopping.
+  //
+  // This will wrap around after 4 billion timer restarts, and if it wraps
+  // around, it would just trigger _this_ timer in advance (but it's hard to
+  // restart it 4 billion times within its duration).
+  TimerGeneration generation_ = TimerGeneration(0);
+  bool is_running_ = false;
+  // Incremented each time time has expired and reset when stopped or restarted.
+  int expiration_count_ = 0;
+};
+
+// Creates and manages timers.
+class TimerManager {
+ public:
+  explicit TimerManager(
+      std::function<std::unique_ptr<Timeout>(
+          webrtc::TaskQueueBase::DelayPrecision)> create_timeout)
+      : create_timeout_(std::move(create_timeout)) {}
+
+  // Creates a timer with name `name` that will expire (when started) after
+  // `options.duration` and call `on_expired`. There are more `options` that
+  // affects the behavior. Note that timers are created initially stopped.
+  std::unique_ptr<Timer> CreateTimer(absl::string_view name,
+                                     Timer::OnExpired on_expired,
+                                     const TimerOptions& options);
+
+  void HandleTimeout(TimeoutID timeout_id);
+
+ private:
+  const std::function<std::unique_ptr<Timeout>(
+      webrtc::TaskQueueBase::DelayPrecision)>
+      create_timeout_;
+  std::map<TimerID, Timer*> timers_;
+  TimerID next_id_ = TimerID(0);
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TIMER_TIMER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/timer/timer_test.cc b/third_party/libwebrtc/net/dcsctp/timer/timer_test.cc
new file mode 100644
index 0000000000..4aebe65b48
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/timer/timer_test.cc
@@ -0,0 +1,459 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/timer/timer.h"
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/public/timeout.h"
+#include "net/dcsctp/timer/fake_timeout.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::Return;
+
+class TimerTest : public testing::Test {
+ protected:
+  TimerTest()
+      : timeout_manager_([this]() { return now_; }),
+        manager_([this](webrtc::TaskQueueBase::DelayPrecision precision) {
+          return timeout_manager_.CreateTimeout(precision);
+        }) {
+    ON_CALL(on_expired_, Call).WillByDefault(Return(absl::nullopt));
+  }
+
+  void AdvanceTimeAndRunTimers(DurationMs duration) {
+    now_ = now_ + duration;
+
+    for (;;) {
+      absl::optional<TimeoutID> timeout_id =
+          timeout_manager_.GetNextExpiredTimeout();
+      if (!timeout_id.has_value()) {
+        break;
+      }
+      manager_.HandleTimeout(*timeout_id);
+    }
+  }
+
+  TimeMs now_ = TimeMs(0);
+  FakeTimeoutManager timeout_manager_;
+  TimerManager manager_;
+  testing::MockFunction<absl::optional<DurationMs>()> on_expired_;
+};
+
+TEST_F(TimerTest, TimerIsInitiallyStopped) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kFixed));
+
+  EXPECT_FALSE(t1->is_running());
+}
+
+TEST_F(TimerTest, TimerExpiresAtGivenTime) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kFixed));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  t1->Start();
+  EXPECT_TRUE(t1->is_running());
+
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+}
+
+TEST_F(TimerTest, TimerReschedulesAfterExpiredWithFixedBackoff) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kFixed));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  t1->Start();
+  EXPECT_EQ(t1->expiration_count(), 0);
+
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  // Fire first time
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_TRUE(t1->is_running());
+  EXPECT_EQ(t1->expiration_count(), 1);
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  // Second time
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_TRUE(t1->is_running());
+  EXPECT_EQ(t1->expiration_count(), 2);
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  // Third time
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_TRUE(t1->is_running());
+  EXPECT_EQ(t1->expiration_count(), 3);
+}
+
+TEST_F(TimerTest, TimerWithNoRestarts) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kFixed,
+                   /*max_restart=*/0));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  t1->Start();
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  // Fire first time
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+
+  EXPECT_FALSE(t1->is_running());
+
+  // Second time - shouldn't fire
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(5000));
+  EXPECT_FALSE(t1->is_running());
+}
+
+TEST_F(TimerTest, TimerWithOneRestart) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kFixed,
+                   /*max_restart=*/1));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  t1->Start();
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  // Fire first time
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_TRUE(t1->is_running());
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  // Second time - max restart limit reached.
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_FALSE(t1->is_running());
+
+  // Third time - should not fire.
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(5000));
+  EXPECT_FALSE(t1->is_running());
+}
+
+TEST_F(TimerTest, TimerWithTwoRestart) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kFixed,
+                   /*max_restart=*/2));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  t1->Start();
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  // Fire first time
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_TRUE(t1->is_running());
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  // Second time
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_TRUE(t1->is_running());
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  // Third time
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_FALSE(t1->is_running());
+}
+
+TEST_F(TimerTest, TimerWithExponentialBackoff) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kExponential));
+
+  t1->Start();
+
+  // Fire first time at 5 seconds
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(5000));
+
+  // Second time at 5*2^1 = 10 seconds later.
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(9000));
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+
+  // Third time at 5*2^2 = 20 seconds later.
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(19000));
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+
+  // Fourth time at 5*2^3 = 40 seconds later.
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(39000));
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+}
+
+TEST_F(TimerTest, StartTimerWillStopAndStart) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kExponential));
+
+  t1->Start();
+
+  AdvanceTimeAndRunTimers(DurationMs(3000));
+
+  t1->Start();
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(2000));
+
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(3000));
+}
+
+TEST_F(TimerTest, ExpirationCounterWillResetIfStopped) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kExponential));
+
+  t1->Start();
+
+  // Fire first time at 5 seconds
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(5000));
+  EXPECT_EQ(t1->expiration_count(), 1);
+
+  // Second time at 5*2^1 = 10 seconds later.
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(9000));
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_EQ(t1->expiration_count(), 2);
+
+  t1->Start();
+  EXPECT_EQ(t1->expiration_count(), 0);
+
+  // Third time at 5*2^0 = 5 seconds later.
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_EQ(t1->expiration_count(), 1);
+}
+
+TEST_F(TimerTest, StopTimerWillMakeItNotExpire) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kExponential));
+
+  t1->Start();
+  EXPECT_TRUE(t1->is_running());
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+  t1->Stop();
+  EXPECT_FALSE(t1->is_running());
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+}
+
+TEST_F(TimerTest, ReturningNewDurationWhenExpired) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(5000), TimerBackoffAlgorithm::kFixed));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  t1->Start();
+  EXPECT_EQ(t1->duration(), DurationMs(5000));
+
+  AdvanceTimeAndRunTimers(DurationMs(4000));
+
+  // Fire first time
+  EXPECT_CALL(on_expired_, Call).WillOnce(Return(DurationMs(2000)));
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_EQ(t1->duration(), DurationMs(2000));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+
+  // Second time
+  EXPECT_CALL(on_expired_, Call).WillOnce(Return(DurationMs(10000)));
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+  EXPECT_EQ(t1->duration(), DurationMs(10000));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(9000));
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+}
+
+TEST_F(TimerTest, TimersHaveMaximumDuration) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(1000), TimerBackoffAlgorithm::kExponential));
+
+  t1->set_duration(DurationMs(2 * *Timer::kMaxTimerDuration));
+  EXPECT_EQ(t1->duration(), Timer::kMaxTimerDuration);
+}
+
+TEST_F(TimerTest, TimersHaveMaximumBackoffDuration) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(1000), TimerBackoffAlgorithm::kExponential));
+
+  t1->Start();
+
+  int max_exponent = static_cast<int>(log2(*Timer::kMaxTimerDuration / 1000));
+  for (int i = 0; i < max_exponent; ++i) {
+    EXPECT_CALL(on_expired_, Call).Times(1);
+    AdvanceTimeAndRunTimers(DurationMs(1000 * (1 << i)));
+  }
+
+  // Reached the maximum duration.
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(Timer::kMaxTimerDuration);
+
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(Timer::kMaxTimerDuration);
+
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(Timer::kMaxTimerDuration);
+
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(Timer::kMaxTimerDuration);
+}
+
+TEST_F(TimerTest, TimerCanBeStartedFromWithinExpirationHandler) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(1000), TimerBackoffAlgorithm::kFixed));
+
+  t1->Start();
+
+  // Start a timer, but don't return any new duration in callback.
+  EXPECT_CALL(on_expired_, Call).WillOnce([&]() {
+    EXPECT_TRUE(t1->is_running());
+    t1->set_duration(DurationMs(5000));
+    t1->Start();
+    return absl::nullopt;
+  });
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(4999));
+
+  // Start a timer, and return any new duration in callback.
+  EXPECT_CALL(on_expired_, Call).WillOnce([&]() {
+    EXPECT_TRUE(t1->is_running());
+    t1->set_duration(DurationMs(5000));
+    t1->Start();
+    return absl::make_optional(DurationMs(8000));
+  });
+  AdvanceTimeAndRunTimers(DurationMs(1));
+
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(7999));
+
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1));
+}
+
+TEST_F(TimerTest, DurationStaysWithinMaxTimerBackOffDuration) {
+  std::unique_ptr<Timer> t1 = manager_.CreateTimer(
+      "t1", on_expired_.AsStdFunction(),
+      TimerOptions(DurationMs(1000), TimerBackoffAlgorithm::kExponential,
+                   /*max_restarts=*/absl::nullopt, DurationMs(5000)));
+
+  t1->Start();
+
+  // Initial timeout, 1000 ms
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1000));
+
+  // Exponential backoff -> 2000 ms
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(1999));
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1));
+
+  // Exponential backoff -> 4000 ms
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(3999));
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1));
+
+  // Limited backoff -> 5000ms
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(4999));
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1));
+
+  // ... where it plateaus
+  EXPECT_CALL(on_expired_, Call).Times(0);
+  AdvanceTimeAndRunTimers(DurationMs(4999));
+  EXPECT_CALL(on_expired_, Call).Times(1);
+  AdvanceTimeAndRunTimers(DurationMs(1));
+}
+
+TEST(TimerManagerTest, TimerManagerPassesPrecisionToCreateTimeoutMethod) {
+  FakeTimeoutManager timeout_manager([&]() { return TimeMs(0); });
+  absl::optional<webrtc::TaskQueueBase::DelayPrecision> create_timer_precison;
+  TimerManager manager([&](webrtc::TaskQueueBase::DelayPrecision precision) {
+    create_timer_precison = precision;
+    return timeout_manager.CreateTimeout(precision);
+  });
+  // Default TimerOptions.
+  manager.CreateTimer(
+      "test_timer", []() { return absl::optional<DurationMs>(); },
+      TimerOptions(DurationMs(123)));
+  EXPECT_EQ(create_timer_precison, webrtc::TaskQueueBase::DelayPrecision::kLow);
+  // High precision TimerOptions.
+  manager.CreateTimer(
+      "test_timer", []() { return absl::optional<DurationMs>(); },
+      TimerOptions(DurationMs(123), TimerBackoffAlgorithm::kExponential,
+                   absl::nullopt, absl::nullopt,
+                   webrtc::TaskQueueBase::DelayPrecision::kHigh));
+  EXPECT_EQ(create_timer_precison,
+            webrtc::TaskQueueBase::DelayPrecision::kHigh);
+  // Low precision TimerOptions.
+  manager.CreateTimer(
+      "test_timer", []() { return absl::optional<DurationMs>(); },
+      TimerOptions(DurationMs(123), TimerBackoffAlgorithm::kExponential,
+                   absl::nullopt, absl::nullopt,
+                   webrtc::TaskQueueBase::DelayPrecision::kLow));
+  EXPECT_EQ(create_timer_precison, webrtc::TaskQueueBase::DelayPrecision::kLow);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/BUILD.gn b/third_party/libwebrtc/net/dcsctp/tx/BUILD.gn
new file mode 100644
index 0000000000..43fd41639e
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/BUILD.gn
@@ -0,0 +1,209 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("send_queue") {
+  deps = [
+    "../../../api:array_view",
+    "../common:internal_types",
+    "../packet:chunk",
+    "../packet:data",
+    "../public:socket",
+    "../public:types",
+  ]
+  sources = [ "send_queue.h" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("rr_send_queue") {
+  deps = [
+    ":send_queue",
+    ":stream_scheduler",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base/containers:flat_map",
+    "../common:str_join",
+    "../packet:data",
+    "../public:socket",
+    "../public:types",
+  ]
+  sources = [
+    "rr_send_queue.cc",
+    "rr_send_queue.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("stream_scheduler") {
+  deps = [
+    ":send_queue",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:strong_alias",
+    "../../../rtc_base/containers:flat_set",
+    "../common:str_join",
+    "../packet:chunk",
+    "../packet:data",
+    "../packet:sctp_packet",
+    "../public:socket",
+    "../public:types",
+  ]
+  sources = [
+    "stream_scheduler.cc",
+    "stream_scheduler.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("retransmission_error_counter") {
+  deps = [
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../public:types",
+  ]
+  sources = [
+    "retransmission_error_counter.cc",
+    "retransmission_error_counter.h",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_library("retransmission_timeout") {
+  deps = [
+    "../../../rtc_base:checks",
+    "../public:types",
+  ]
+  sources = [
+    "retransmission_timeout.cc",
+    "retransmission_timeout.h",
+  ]
+}
+
+rtc_library("outstanding_data") {
+  deps = [
+    ":retransmission_timeout",
+    ":send_queue",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../common:math",
+    "../common:sequence_numbers",
+    "../common:str_join",
+    "../packet:chunk",
+    "../packet:data",
+    "../public:socket",
+    "../public:types",
+    "../timer",
+  ]
+  sources = [
+    "outstanding_data.cc",
+    "outstanding_data.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("retransmission_queue") {
+  deps = [
+    ":outstanding_data",
+    ":retransmission_timeout",
+    ":send_queue",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:stringutils",
+    "../common:math",
+    "../common:sequence_numbers",
+    "../common:str_join",
+    "../packet:chunk",
+    "../packet:data",
+    "../public:socket",
+    "../public:types",
+    "../timer",
+  ]
+  sources = [
+    "retransmission_queue.cc",
+    "retransmission_queue.h",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_source_set("mock_send_queue") {
+    testonly = true
+    deps = [
+      ":send_queue",
+      "../../../api:array_view",
+      "../../../test:test_support",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+    sources = [ "mock_send_queue.h" ]
+  }
+
+  rtc_library("dcsctp_tx_unittests") {
+    testonly = true
+
+    deps = [
+      ":mock_send_queue",
+      ":outstanding_data",
+      ":retransmission_error_counter",
+      ":retransmission_queue",
+      ":retransmission_timeout",
+      ":rr_send_queue",
+      ":send_queue",
+      ":stream_scheduler",
+      "../../../api:array_view",
+      "../../../api/task_queue:task_queue",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:gunit_helpers",
+      "../../../test:test_support",
+      "../common:handover_testing",
+      "../common:math",
+      "../common:sequence_numbers",
+      "../packet:chunk",
+      "../packet:data",
+      "../packet:sctp_packet",
+      "../public:socket",
+      "../public:types",
+      "../socket:mock_callbacks",
+      "../socket:mock_callbacks",
+      "../testing:data_generator",
+      "../testing:testing_macros",
+      "../timer",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+    sources = [
+      "outstanding_data_test.cc",
+      "retransmission_error_counter_test.cc",
+      "retransmission_queue_test.cc",
+      "retransmission_timeout_test.cc",
+      "rr_send_queue_test.cc",
+      "stream_scheduler_test.cc",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/net/dcsctp/tx/mock_send_queue.h b/third_party/libwebrtc/net/dcsctp/tx/mock_send_queue.h
new file mode 100644
index 0000000000..0c8f5d141d
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/mock_send_queue.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TX_MOCK_SEND_QUEUE_H_
+#define NET_DCSCTP_TX_MOCK_SEND_QUEUE_H_
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/tx/send_queue.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+
+class MockSendQueue : public SendQueue {
+ public:
+  MockSendQueue() {
+    ON_CALL(*this, Produce).WillByDefault([](TimeMs now, size_t max_size) {
+      return absl::nullopt;
+    });
+  }
+
+  MOCK_METHOD(absl::optional<SendQueue::DataToSend>,
+              Produce,
+              (TimeMs now, size_t max_size),
+              (override));
+  MOCK_METHOD(bool,
+              Discard,
+              (IsUnordered unordered, StreamID stream_id, MID message_id),
+              (override));
+  MOCK_METHOD(void, PrepareResetStream, (StreamID stream_id), (override));
+  MOCK_METHOD(bool, HasStreamsReadyToBeReset, (), (const, override));
+  MOCK_METHOD(std::vector<StreamID>, GetStreamsReadyToBeReset, (), (override));
+  MOCK_METHOD(void, CommitResetStreams, (), (override));
+  MOCK_METHOD(void, RollbackResetStreams, (), (override));
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(size_t, buffered_amount, (StreamID stream_id), (const, override));
+  MOCK_METHOD(size_t, total_buffered_amount, (), (const, override));
+  MOCK_METHOD(size_t,
+              buffered_amount_low_threshold,
+              (StreamID stream_id),
+              (const, override));
+  MOCK_METHOD(void,
+              SetBufferedAmountLowThreshold,
+              (StreamID stream_id, size_t bytes),
+              (override));
+  MOCK_METHOD(void, EnableMessageInterleaving, (bool enabled), (override));
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TX_MOCK_SEND_QUEUE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/tx/outstanding_data.cc b/third_party/libwebrtc/net/dcsctp/tx/outstanding_data.cc
new file mode 100644
index 0000000000..4f1e863056
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/outstanding_data.cc
@@ -0,0 +1,543 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/outstanding_data.h"
+
+#include <algorithm>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "net/dcsctp/common/math.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/public/types.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+
+// The number of times a packet must be NACKed before it's retransmitted.
+// See https://tools.ietf.org/html/rfc4960#section-7.2.4
+constexpr uint8_t kNumberOfNacksForRetransmission = 3;
+
+// Returns how large a chunk will be, serialized, carrying the data
+size_t OutstandingData::GetSerializedChunkSize(const Data& data) const {
+  return RoundUpTo4(data_chunk_header_size_ + data.size());
+}
+
+void OutstandingData::Item::Ack() {
+  if (lifecycle_ != Lifecycle::kAbandoned) {
+    lifecycle_ = Lifecycle::kActive;
+  }
+  ack_state_ = AckState::kAcked;
+}
+
+OutstandingData::Item::NackAction OutstandingData::Item::Nack(
+    bool retransmit_now) {
+  ack_state_ = AckState::kNacked;
+  ++nack_count_;
+  if (!should_be_retransmitted() && !is_abandoned() &&
+      (retransmit_now || nack_count_ >= kNumberOfNacksForRetransmission)) {
+    // Nacked enough times - it's considered lost.
+    if (num_retransmissions_ < *max_retransmissions_) {
+      lifecycle_ = Lifecycle::kToBeRetransmitted;
+      return NackAction::kRetransmit;
+    }
+    Abandon();
+    return NackAction::kAbandon;
+  }
+  return NackAction::kNothing;
+}
+
+void OutstandingData::Item::MarkAsRetransmitted() {
+  lifecycle_ = Lifecycle::kActive;
+  ack_state_ = AckState::kUnacked;
+
+  nack_count_ = 0;
+  ++num_retransmissions_;
+}
+
+void OutstandingData::Item::Abandon() {
+  lifecycle_ = Lifecycle::kAbandoned;
+}
+
+bool OutstandingData::Item::has_expired(TimeMs now) const {
+  return expires_at_ <= now;
+}
+
+bool OutstandingData::IsConsistent() const {
+  size_t actual_outstanding_bytes = 0;
+  size_t actual_outstanding_items = 0;
+
+  std::set<UnwrappedTSN> combined_to_be_retransmitted;
+  combined_to_be_retransmitted.insert(to_be_retransmitted_.begin(),
+                                      to_be_retransmitted_.end());
+  combined_to_be_retransmitted.insert(to_be_fast_retransmitted_.begin(),
+                                      to_be_fast_retransmitted_.end());
+
+  std::set<UnwrappedTSN> actual_combined_to_be_retransmitted;
+  for (const auto& [tsn, item] : outstanding_data_) {
+    if (item.is_outstanding()) {
+      actual_outstanding_bytes += GetSerializedChunkSize(item.data());
+      ++actual_outstanding_items;
+    }
+
+    if (item.should_be_retransmitted()) {
+      actual_combined_to_be_retransmitted.insert(tsn);
+    }
+  }
+
+  if (outstanding_data_.empty() &&
+      next_tsn_ != last_cumulative_tsn_ack_.next_value()) {
+    return false;
+  }
+
+  return actual_outstanding_bytes == outstanding_bytes_ &&
+         actual_outstanding_items == outstanding_items_ &&
+         actual_combined_to_be_retransmitted == combined_to_be_retransmitted;
+}
+
+void OutstandingData::AckChunk(AckInfo& ack_info,
+                               std::map<UnwrappedTSN, Item>::iterator iter) {
+  if (!iter->second.is_acked()) {
+    size_t serialized_size = GetSerializedChunkSize(iter->second.data());
+    ack_info.bytes_acked += serialized_size;
+    if (iter->second.is_outstanding()) {
+      outstanding_bytes_ -= serialized_size;
+      --outstanding_items_;
+    }
+    if (iter->second.should_be_retransmitted()) {
+      RTC_DCHECK(to_be_fast_retransmitted_.find(iter->first) ==
+                 to_be_fast_retransmitted_.end());
+      to_be_retransmitted_.erase(iter->first);
+    }
+    iter->second.Ack();
+    ack_info.highest_tsn_acked =
+        std::max(ack_info.highest_tsn_acked, iter->first);
+  }
+}
+
+OutstandingData::AckInfo OutstandingData::HandleSack(
+    UnwrappedTSN cumulative_tsn_ack,
+    rtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
+    bool is_in_fast_recovery) {
+  OutstandingData::AckInfo ack_info(cumulative_tsn_ack);
+  // Erase all items up to cumulative_tsn_ack.
+  RemoveAcked(cumulative_tsn_ack, ack_info);
+
+  // ACK packets reported in the gap ack blocks
+  AckGapBlocks(cumulative_tsn_ack, gap_ack_blocks, ack_info);
+
+  // NACK and possibly mark for retransmit chunks that weren't acked.
+  NackBetweenAckBlocks(cumulative_tsn_ack, gap_ack_blocks, is_in_fast_recovery,
+                       ack_info);
+
+  RTC_DCHECK(IsConsistent());
+  return ack_info;
+}
+
+void OutstandingData::RemoveAcked(UnwrappedTSN cumulative_tsn_ack,
+                                  AckInfo& ack_info) {
+  auto first_unacked = outstanding_data_.upper_bound(cumulative_tsn_ack);
+
+  for (auto iter = outstanding_data_.begin(); iter != first_unacked; ++iter) {
+    AckChunk(ack_info, iter);
+    if (iter->second.lifecycle_id().IsSet()) {
+      RTC_DCHECK(iter->second.data().is_end);
+      if (iter->second.is_abandoned()) {
+        ack_info.abandoned_lifecycle_ids.push_back(iter->second.lifecycle_id());
+      } else {
+        ack_info.acked_lifecycle_ids.push_back(iter->second.lifecycle_id());
+      }
+    }
+  }
+
+  outstanding_data_.erase(outstanding_data_.begin(), first_unacked);
+  last_cumulative_tsn_ack_ = cumulative_tsn_ack;
+}
+
+void OutstandingData::AckGapBlocks(
+    UnwrappedTSN cumulative_tsn_ack,
+    rtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
+    AckInfo& ack_info) {
+  // Mark all non-gaps as ACKED (but they can't be removed) as (from RFC)
+  // "SCTP considers the information carried in the Gap Ack Blocks in the
+  // SACK chunk as advisory.". Note that when NR-SACK is supported, this can be
+  // handled differently.
+
+  for (auto& block : gap_ack_blocks) {
+    auto start = outstanding_data_.lower_bound(
+        UnwrappedTSN::AddTo(cumulative_tsn_ack, block.start));
+    auto end = outstanding_data_.upper_bound(
+        UnwrappedTSN::AddTo(cumulative_tsn_ack, block.end));
+    for (auto iter = start; iter != end; ++iter) {
+      AckChunk(ack_info, iter);
+    }
+  }
+}
+
+void OutstandingData::NackBetweenAckBlocks(
+    UnwrappedTSN cumulative_tsn_ack,
+    rtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
+    bool is_in_fast_recovery,
+    OutstandingData::AckInfo& ack_info) {
+  // Mark everything between the blocks as NACKED/TO_BE_RETRANSMITTED.
+  // https://tools.ietf.org/html/rfc4960#section-7.2.4
+  // "Mark the DATA chunk(s) with three miss indications for retransmission."
+  // "For each incoming SACK, miss indications are incremented only for
+  // missing TSNs prior to the highest TSN newly acknowledged in the SACK."
+  //
+  // What this means is that only when there is a increasing stream of data
+  // received and there are new packets seen (since last time), packets that are
+  // in-flight and between gaps should be nacked. This means that SCTP relies on
+  // the T3-RTX-timer to re-send packets otherwise.
+  UnwrappedTSN max_tsn_to_nack = ack_info.highest_tsn_acked;
+  if (is_in_fast_recovery && cumulative_tsn_ack > last_cumulative_tsn_ack_) {
+    // https://tools.ietf.org/html/rfc4960#section-7.2.4
+    // "If an endpoint is in Fast Recovery and a SACK arrives that advances
+    // the Cumulative TSN Ack Point, the miss indications are incremented for
+    // all TSNs reported missing in the SACK."
+    max_tsn_to_nack = UnwrappedTSN::AddTo(
+        cumulative_tsn_ack,
+        gap_ack_blocks.empty() ? 0 : gap_ack_blocks.rbegin()->end);
+  }
+
+  UnwrappedTSN prev_block_last_acked = cumulative_tsn_ack;
+  for (auto& block : gap_ack_blocks) {
+    UnwrappedTSN cur_block_first_acked =
+        UnwrappedTSN::AddTo(cumulative_tsn_ack, block.start);
+    for (auto iter = outstanding_data_.upper_bound(prev_block_last_acked);
+         iter != outstanding_data_.lower_bound(cur_block_first_acked); ++iter) {
+      if (iter->first <= max_tsn_to_nack) {
+        ack_info.has_packet_loss |=
+            NackItem(iter->first, iter->second, /*retransmit_now=*/false,
+                     /*do_fast_retransmit=*/!is_in_fast_recovery);
+      }
+    }
+    prev_block_last_acked = UnwrappedTSN::AddTo(cumulative_tsn_ack, block.end);
+  }
+
+  // Note that packets are not NACKED which are above the highest gap-ack-block
+  // (or above the cumulative ack TSN if no gap-ack-blocks) as only packets
+  // up until the highest_tsn_acked (see above) should be considered when
+  // NACKing.
+}
+
+bool OutstandingData::NackItem(UnwrappedTSN tsn,
+                               Item& item,
+                               bool retransmit_now,
+                               bool do_fast_retransmit) {
+  if (item.is_outstanding()) {
+    outstanding_bytes_ -= GetSerializedChunkSize(item.data());
+    --outstanding_items_;
+  }
+
+  switch (item.Nack(retransmit_now)) {
+    case Item::NackAction::kNothing:
+      return false;
+    case Item::NackAction::kRetransmit:
+      if (do_fast_retransmit) {
+        to_be_fast_retransmitted_.insert(tsn);
+      } else {
+        to_be_retransmitted_.insert(tsn);
+      }
+      RTC_DLOG(LS_VERBOSE) << *tsn.Wrap() << " marked for retransmission";
+      break;
+    case Item::NackAction::kAbandon:
+      AbandonAllFor(item);
+      break;
+  }
+  return true;
+}
+
+void OutstandingData::AbandonAllFor(const Item& item) {
+  // Erase all remaining chunks from the producer, if any.
+  if (discard_from_send_queue_(item.data().is_unordered, item.data().stream_id,
+                               item.data().message_id)) {
+    // There were remaining chunks to be produced for this message. Since the
+    // receiver may have already received all chunks (up till now) for this
+    // message, we can't just FORWARD-TSN to the last fragment in this
+    // (abandoned) message and start sending a new message, as the receiver will
+    // then see a new message before the end of the previous one was seen (or
+    // skipped over). So create a new fragment, representing the end, that the
+    // received will never see as it is abandoned immediately and used as cum
+    // TSN in the sent FORWARD-TSN.
+    UnwrappedTSN tsn = next_tsn_;
+    next_tsn_.Increment();
+    Data message_end(item.data().stream_id, item.data().ssn,
+                     item.data().message_id, item.data().fsn, item.data().ppid,
+                     std::vector<uint8_t>(), Data::IsBeginning(false),
+                     Data::IsEnd(true), item.data().is_unordered);
+    Item& added_item =
+        outstanding_data_
+            .emplace(std::piecewise_construct, std::forward_as_tuple(tsn),
+                     std::forward_as_tuple(std::move(message_end), TimeMs(0),
+                                           MaxRetransmits::NoLimit(),
+                                           TimeMs::InfiniteFuture(),
+                                           LifecycleId::NotSet()))
+            .first->second;
+    // The added chunk shouldn't be included in `outstanding_bytes`, so set it
+    // as acked.
+    added_item.Ack();
+    RTC_DLOG(LS_VERBOSE) << "Adding unsent end placeholder for message at tsn="
+                         << *tsn.Wrap();
+  }
+
+  for (auto& [tsn, other] : outstanding_data_) {
+    if (!other.is_abandoned() &&
+        other.data().stream_id == item.data().stream_id &&
+        other.data().is_unordered == item.data().is_unordered &&
+        other.data().message_id == item.data().message_id) {
+      RTC_DLOG(LS_VERBOSE) << "Marking chunk " << *tsn.Wrap()
+                           << " as abandoned";
+      if (other.should_be_retransmitted()) {
+        to_be_fast_retransmitted_.erase(tsn);
+        to_be_retransmitted_.erase(tsn);
+      }
+      other.Abandon();
+    }
+  }
+}
+
+std::vector<std::pair<TSN, Data>> OutstandingData::ExtractChunksThatCanFit(
+    std::set<UnwrappedTSN>& chunks,
+    size_t max_size) {
+  std::vector<std::pair<TSN, Data>> result;
+
+  for (auto it = chunks.begin(); it != chunks.end();) {
+    UnwrappedTSN tsn = *it;
+    auto elem = outstanding_data_.find(tsn);
+    RTC_DCHECK(elem != outstanding_data_.end());
+    Item& item = elem->second;
+    RTC_DCHECK(item.should_be_retransmitted());
+    RTC_DCHECK(!item.is_outstanding());
+    RTC_DCHECK(!item.is_abandoned());
+    RTC_DCHECK(!item.is_acked());
+
+    size_t serialized_size = GetSerializedChunkSize(item.data());
+    if (serialized_size <= max_size) {
+      item.MarkAsRetransmitted();
+      result.emplace_back(tsn.Wrap(), item.data().Clone());
+      max_size -= serialized_size;
+      outstanding_bytes_ += serialized_size;
+      ++outstanding_items_;
+      it = chunks.erase(it);
+    } else {
+      ++it;
+    }
+    // No point in continuing if the packet is full.
+    if (max_size <= data_chunk_header_size_) {
+      break;
+    }
+  }
+  return result;
+}
+
+std::vector<std::pair<TSN, Data>>
+OutstandingData::GetChunksToBeFastRetransmitted(size_t max_size) {
+  std::vector<std::pair<TSN, Data>> result =
+      ExtractChunksThatCanFit(to_be_fast_retransmitted_, max_size);
+
+  // https://datatracker.ietf.org/doc/html/rfc4960#section-7.2.4
+  // "Those TSNs marked for retransmission due to the Fast-Retransmit algorithm
+  // that did not fit in the sent datagram carrying K other TSNs are also marked
+  // as ineligible for a subsequent Fast Retransmit.  However, as they are
+  // marked for retransmission they will be retransmitted later on as soon as
+  // cwnd allows."
+  if (!to_be_fast_retransmitted_.empty()) {
+    to_be_retransmitted_.insert(to_be_fast_retransmitted_.begin(),
+                                to_be_fast_retransmitted_.end());
+    to_be_fast_retransmitted_.clear();
+  }
+
+  RTC_DCHECK(IsConsistent());
+  return result;
+}
+
+std::vector<std::pair<TSN, Data>> OutstandingData::GetChunksToBeRetransmitted(
+    size_t max_size) {
+  // Chunks scheduled for fast retransmission must be sent first.
+  RTC_DCHECK(to_be_fast_retransmitted_.empty());
+  return ExtractChunksThatCanFit(to_be_retransmitted_, max_size);
+}
+
+void OutstandingData::ExpireOutstandingChunks(TimeMs now) {
+  for (const auto& [tsn, item] : outstanding_data_) {
+    // Chunks that are nacked can be expired. Care should be taken not to expire
+    // unacked (in-flight) chunks as they might have been received, but the SACK
+    // is either delayed or in-flight and may be received later.
+    if (item.is_abandoned()) {
+      // Already abandoned.
+    } else if (item.is_nacked() && item.has_expired(now)) {
+      RTC_DLOG(LS_VERBOSE) << "Marking nacked chunk " << *tsn.Wrap()
+                           << " and message " << *item.data().message_id
+                           << " as expired";
+      AbandonAllFor(item);
+    } else {
+      // A non-expired chunk. No need to iterate any further.
+      break;
+    }
+  }
+  RTC_DCHECK(IsConsistent());
+}
+
+UnwrappedTSN OutstandingData::highest_outstanding_tsn() const {
+  return outstanding_data_.empty() ? last_cumulative_tsn_ack_
+                                   : outstanding_data_.rbegin()->first;
+}
+
+absl::optional<UnwrappedTSN> OutstandingData::Insert(
+    const Data& data,
+    TimeMs time_sent,
+    MaxRetransmits max_retransmissions,
+    TimeMs expires_at,
+    LifecycleId lifecycle_id) {
+  UnwrappedTSN tsn = next_tsn_;
+  next_tsn_.Increment();
+
+  // All chunks are always padded to be even divisible by 4.
+  size_t chunk_size = GetSerializedChunkSize(data);
+  outstanding_bytes_ += chunk_size;
+  ++outstanding_items_;
+  auto it = outstanding_data_
+                .emplace(std::piecewise_construct, std::forward_as_tuple(tsn),
+                         std::forward_as_tuple(data.Clone(), time_sent,
+                                               max_retransmissions, expires_at,
+                                               lifecycle_id))
+                .first;
+
+  if (it->second.has_expired(time_sent)) {
+    // No need to send it - it was expired when it was in the send
+    // queue.
+    RTC_DLOG(LS_VERBOSE) << "Marking freshly produced chunk "
+                         << *it->first.Wrap() << " and message "
+                         << *it->second.data().message_id << " as expired";
+    AbandonAllFor(it->second);
+    RTC_DCHECK(IsConsistent());
+    return absl::nullopt;
+  }
+
+  RTC_DCHECK(IsConsistent());
+  return tsn;
+}
+
+void OutstandingData::NackAll() {
+  for (auto& [tsn, item] : outstanding_data_) {
+    if (!item.is_acked()) {
+      NackItem(tsn, item, /*retransmit_now=*/true,
+               /*do_fast_retransmit=*/false);
+    }
+  }
+  RTC_DCHECK(IsConsistent());
+}
+
+absl::optional<DurationMs> OutstandingData::MeasureRTT(TimeMs now,
+                                                       UnwrappedTSN tsn) const {
+  auto it = outstanding_data_.find(tsn);
+  if (it != outstanding_data_.end() && !it->second.has_been_retransmitted()) {
+    // https://tools.ietf.org/html/rfc4960#section-6.3.1
+    // "Karn's algorithm: RTT measurements MUST NOT be made using
+    // packets that were retransmitted (and thus for which it is ambiguous
+    // whether the reply was for the first instance of the chunk or for a
+    // later instance)"
+    return now - it->second.time_sent();
+  }
+  return absl::nullopt;
+}
+
+std::vector<std::pair<TSN, OutstandingData::State>>
+OutstandingData::GetChunkStatesForTesting() const {
+  std::vector<std::pair<TSN, State>> states;
+  states.emplace_back(last_cumulative_tsn_ack_.Wrap(), State::kAcked);
+  for (const auto& [tsn, item] : outstanding_data_) {
+    State state;
+    if (item.is_abandoned()) {
+      state = State::kAbandoned;
+    } else if (item.should_be_retransmitted()) {
+      state = State::kToBeRetransmitted;
+    } else if (item.is_acked()) {
+      state = State::kAcked;
+    } else if (item.is_outstanding()) {
+      state = State::kInFlight;
+    } else {
+      state = State::kNacked;
+    }
+
+    states.emplace_back(tsn.Wrap(), state);
+  }
+  return states;
+}
+
+bool OutstandingData::ShouldSendForwardTsn() const {
+  if (!outstanding_data_.empty()) {
+    auto it = outstanding_data_.begin();
+    return it->first == last_cumulative_tsn_ack_.next_value() &&
+           it->second.is_abandoned();
+  }
+  return false;
+}
+
+ForwardTsnChunk OutstandingData::CreateForwardTsn() const {
+  std::map<StreamID, SSN> skipped_per_ordered_stream;
+  UnwrappedTSN new_cumulative_ack = last_cumulative_tsn_ack_;
+
+  for (const auto& [tsn, item] : outstanding_data_) {
+    if ((tsn != new_cumulative_ack.next_value()) || !item.is_abandoned()) {
+      break;
+    }
+    new_cumulative_ack = tsn;
+    if (!item.data().is_unordered &&
+        item.data().ssn > skipped_per_ordered_stream[item.data().stream_id]) {
+      skipped_per_ordered_stream[item.data().stream_id] = item.data().ssn;
+    }
+  }
+
+  std::vector<ForwardTsnChunk::SkippedStream> skipped_streams;
+  skipped_streams.reserve(skipped_per_ordered_stream.size());
+  for (const auto& [stream_id, ssn] : skipped_per_ordered_stream) {
+    skipped_streams.emplace_back(stream_id, ssn);
+  }
+  return ForwardTsnChunk(new_cumulative_ack.Wrap(), std::move(skipped_streams));
+}
+
+IForwardTsnChunk OutstandingData::CreateIForwardTsn() const {
+  std::map<std::pair<IsUnordered, StreamID>, MID> skipped_per_stream;
+  UnwrappedTSN new_cumulative_ack = last_cumulative_tsn_ack_;
+
+  for (const auto& [tsn, item] : outstanding_data_) {
+    if ((tsn != new_cumulative_ack.next_value()) || !item.is_abandoned()) {
+      break;
+    }
+    new_cumulative_ack = tsn;
+    std::pair<IsUnordered, StreamID> stream_id =
+        std::make_pair(item.data().is_unordered, item.data().stream_id);
+
+    if (item.data().message_id > skipped_per_stream[stream_id]) {
+      skipped_per_stream[stream_id] = item.data().message_id;
+    }
+  }
+
+  std::vector<IForwardTsnChunk::SkippedStream> skipped_streams;
+  skipped_streams.reserve(skipped_per_stream.size());
+  for (const auto& [stream, message_id] : skipped_per_stream) {
+    skipped_streams.emplace_back(stream.first, stream.second, message_id);
+  }
+
+  return IForwardTsnChunk(new_cumulative_ack.Wrap(),
+                          std::move(skipped_streams));
+}
+
+void OutstandingData::ResetSequenceNumbers(UnwrappedTSN next_tsn,
+                                           UnwrappedTSN last_cumulative_tsn) {
+  RTC_DCHECK(outstanding_data_.empty());
+  RTC_DCHECK(next_tsn_ == last_cumulative_tsn_ack_.next_value());
+  RTC_DCHECK(next_tsn == last_cumulative_tsn.next_value());
+  next_tsn_ = next_tsn;
+  last_cumulative_tsn_ack_ = last_cumulative_tsn;
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/outstanding_data.h b/third_party/libwebrtc/net/dcsctp/tx/outstanding_data.h
new file mode 100644
index 0000000000..6b4b7121fb
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/outstanding_data.h
@@ -0,0 +1,350 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TX_OUTSTANDING_DATA_H_
+#define NET_DCSCTP_TX_OUTSTANDING_DATA_H_
+
+#include <map>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// This class keeps track of outstanding data chunks (sent, not yet acked) and
+// handles acking, nacking, rescheduling and abandoning.
+class OutstandingData {
+ public:
+  // State for DATA chunks (message fragments) in the queue - used in tests.
+  enum class State {
+    // The chunk has been sent but not received yet (from the sender's point of
+    // view, as no SACK has been received yet that reference this chunk).
+    kInFlight,
+    // A SACK has been received which explicitly marked this chunk as missing -
+    // it's now NACKED and may be retransmitted if NACKED enough times.
+    kNacked,
+    // A chunk that will be retransmitted when possible.
+    kToBeRetransmitted,
+    // A SACK has been received which explicitly marked this chunk as received.
+    kAcked,
+    // A chunk whose message has expired or has been retransmitted too many
+    // times (RFC3758). It will not be retransmitted anymore.
+    kAbandoned,
+  };
+
+  // Contains variables scoped to a processing of an incoming SACK.
+  struct AckInfo {
+    explicit AckInfo(UnwrappedTSN cumulative_tsn_ack)
+        : highest_tsn_acked(cumulative_tsn_ack) {}
+
+    // Bytes acked by increasing cumulative_tsn_ack and gap_ack_blocks.
+    size_t bytes_acked = 0;
+
+    // Indicates if this SACK indicates that packet loss has occurred. Just
+    // because a packet is missing in the SACK doesn't necessarily mean that
+    // there is packet loss as that packet might be in-flight and received
+    // out-of-order. But when it has been reported missing consecutive times, it
+    // will eventually be considered "lost" and this will be set.
+    bool has_packet_loss = false;
+
+    // Highest TSN Newly Acknowledged, an SCTP variable.
+    UnwrappedTSN highest_tsn_acked;
+
+    // The set of lifecycle IDs that were acked using cumulative_tsn_ack.
+    std::vector<LifecycleId> acked_lifecycle_ids;
+    // The set of lifecycle IDs that were acked, but had been abandoned.
+    std::vector<LifecycleId> abandoned_lifecycle_ids;
+  };
+
+  OutstandingData(
+      size_t data_chunk_header_size,
+      UnwrappedTSN next_tsn,
+      UnwrappedTSN last_cumulative_tsn_ack,
+      std::function<bool(IsUnordered, StreamID, MID)> discard_from_send_queue)
+      : data_chunk_header_size_(data_chunk_header_size),
+        next_tsn_(next_tsn),
+        last_cumulative_tsn_ack_(last_cumulative_tsn_ack),
+        discard_from_send_queue_(std::move(discard_from_send_queue)) {}
+
+  AckInfo HandleSack(
+      UnwrappedTSN cumulative_tsn_ack,
+      rtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
+      bool is_in_fast_recovery);
+
+  // Returns as many of the chunks that are eligible for fast retransmissions
+  // and that would fit in a single packet of `max_size`. The eligible chunks
+  // that didn't fit will be marked for (normal) retransmission and will not be
+  // returned if this method is called again.
+  std::vector<std::pair<TSN, Data>> GetChunksToBeFastRetransmitted(
+      size_t max_size);
+
+  // Given `max_size` of space left in a packet, which chunks can be added to
+  // it?
+  std::vector<std::pair<TSN, Data>> GetChunksToBeRetransmitted(size_t max_size);
+
+  size_t outstanding_bytes() const { return outstanding_bytes_; }
+
+  // Returns the number of DATA chunks that are in-flight.
+  size_t outstanding_items() const { return outstanding_items_; }
+
+  // Given the current time `now_ms`, expire and abandon outstanding (sent at
+  // least once) chunks that have a limited lifetime.
+  void ExpireOutstandingChunks(TimeMs now);
+
+  bool empty() const { return outstanding_data_.empty(); }
+
+  bool has_data_to_be_fast_retransmitted() const {
+    return !to_be_fast_retransmitted_.empty();
+  }
+
+  bool has_data_to_be_retransmitted() const {
+    return !to_be_retransmitted_.empty() || !to_be_fast_retransmitted_.empty();
+  }
+
+  UnwrappedTSN last_cumulative_tsn_ack() const {
+    return last_cumulative_tsn_ack_;
+  }
+
+  UnwrappedTSN next_tsn() const { return next_tsn_; }
+
+  UnwrappedTSN highest_outstanding_tsn() const;
+
+  // Schedules `data` to be sent, with the provided partial reliability
+  // parameters. Returns the TSN if the item was actually added and scheduled to
+  // be sent, and absl::nullopt if it shouldn't be sent.
+  absl::optional<UnwrappedTSN> Insert(
+      const Data& data,
+      TimeMs time_sent,
+      MaxRetransmits max_retransmissions = MaxRetransmits::NoLimit(),
+      TimeMs expires_at = TimeMs::InfiniteFuture(),
+      LifecycleId lifecycle_id = LifecycleId::NotSet());
+
+  // Nacks all outstanding data.
+  void NackAll();
+
+  // Creates a FORWARD-TSN chunk.
+  ForwardTsnChunk CreateForwardTsn() const;
+
+  // Creates an I-FORWARD-TSN chunk.
+  IForwardTsnChunk CreateIForwardTsn() const;
+
+  // Given the current time and a TSN, it returns the measured RTT between when
+  // the chunk was sent and now. It takes into acccount Karn's algorithm, so if
+  // the chunk has ever been retransmitted, it will return absl::nullopt.
+  absl::optional<DurationMs> MeasureRTT(TimeMs now, UnwrappedTSN tsn) const;
+
+  // Returns the internal state of all queued chunks. This is only used in
+  // unit-tests.
+  std::vector<std::pair<TSN, State>> GetChunkStatesForTesting() const;
+
+  // Returns true if the next chunk that is not acked by the peer has been
+  // abandoned, which means that a FORWARD-TSN should be sent.
+  bool ShouldSendForwardTsn() const;
+
+  // Sets the next TSN to be used. This is used in handover.
+  void ResetSequenceNumbers(UnwrappedTSN next_tsn,
+                            UnwrappedTSN last_cumulative_tsn);
+
+ private:
+  // A fragmented message's DATA chunk while in the retransmission queue, and
+  // its associated metadata.
+  class Item {
+   public:
+    enum class NackAction {
+      kNothing,
+      kRetransmit,
+      kAbandon,
+    };
+
+    Item(Data data,
+         TimeMs time_sent,
+         MaxRetransmits max_retransmissions,
+         TimeMs expires_at,
+         LifecycleId lifecycle_id)
+        : time_sent_(time_sent),
+          max_retransmissions_(max_retransmissions),
+          expires_at_(expires_at),
+          lifecycle_id_(lifecycle_id),
+          data_(std::move(data)) {}
+
+    Item(const Item&) = delete;
+    Item& operator=(const Item&) = delete;
+
+    TimeMs time_sent() const { return time_sent_; }
+
+    const Data& data() const { return data_; }
+
+    // Acks an item.
+    void Ack();
+
+    // Nacks an item. If it has been nacked enough times, or if `retransmit_now`
+    // is set, it might be marked for retransmission. If the item has reached
+    // its max retransmission value, it will instead be abandoned. The action
+    // performed is indicated as return value.
+    NackAction Nack(bool retransmit_now);
+
+    // Prepares the item to be retransmitted. Sets it as outstanding and
+    // clears all nack counters.
+    void MarkAsRetransmitted();
+
+    // Marks this item as abandoned.
+    void Abandon();
+
+    bool is_outstanding() const { return ack_state_ == AckState::kUnacked; }
+    bool is_acked() const { return ack_state_ == AckState::kAcked; }
+    bool is_nacked() const { return ack_state_ == AckState::kNacked; }
+    bool is_abandoned() const { return lifecycle_ == Lifecycle::kAbandoned; }
+
+    // Indicates if this chunk should be retransmitted.
+    bool should_be_retransmitted() const {
+      return lifecycle_ == Lifecycle::kToBeRetransmitted;
+    }
+    // Indicates if this chunk has ever been retransmitted.
+    bool has_been_retransmitted() const { return num_retransmissions_ > 0; }
+
+    // Given the current time, and the current state of this DATA chunk, it will
+    // indicate if it has expired (SCTP Partial Reliability Extension).
+    bool has_expired(TimeMs now) const;
+
+    LifecycleId lifecycle_id() const { return lifecycle_id_; }
+
+   private:
+    enum class Lifecycle : uint8_t {
+      // The chunk is alive (sent, received, etc)
+      kActive,
+      // The chunk is scheduled to be retransmitted, and will then transition to
+      // become active.
+      kToBeRetransmitted,
+      // The chunk has been abandoned. This is a terminal state.
+      kAbandoned
+    };
+    enum class AckState : uint8_t {
+      // The chunk is in-flight.
+      kUnacked,
+      // The chunk has been received and acknowledged.
+      kAcked,
+      // The chunk has been nacked and is possibly lost.
+      kNacked
+    };
+
+    // NOTE: This data structure has been optimized for size, by ordering fields
+    // to avoid unnecessary padding.
+
+    // When the packet was sent, and placed in this queue.
+    const TimeMs time_sent_;
+    // If the message was sent with a maximum number of retransmissions, this is
+    // set to that number. The value zero (0) means that it will never be
+    // retransmitted.
+    const MaxRetransmits max_retransmissions_;
+
+    // Indicates the life cycle status of this chunk.
+    Lifecycle lifecycle_ = Lifecycle::kActive;
+    // Indicates the presence of this chunk, if it's in flight (Unacked), has
+    // been received (Acked) or is possibly lost (Nacked).
+    AckState ack_state_ = AckState::kUnacked;
+
+    // The number of times the DATA chunk has been nacked (by having received a
+    // SACK which doesn't include it). Will be cleared on retransmissions.
+    uint8_t nack_count_ = 0;
+    // The number of times the DATA chunk has been retransmitted.
+    uint16_t num_retransmissions_ = 0;
+
+    // At this exact millisecond, the item is considered expired. If the message
+    // is not to be expired, this is set to the infinite future.
+    const TimeMs expires_at_;
+
+    // An optional lifecycle id, which may only be set for the last fragment.
+    const LifecycleId lifecycle_id_;
+
+    // The actual data to send/retransmit.
+    const Data data_;
+  };
+
+  // Returns how large a chunk will be, serialized, carrying the data
+  size_t GetSerializedChunkSize(const Data& data) const;
+
+  // Given a `cumulative_tsn_ack` from an incoming SACK, will remove those items
+  // in the retransmission queue up until this value and will update `ack_info`
+  // by setting `bytes_acked_by_cumulative_tsn_ack`.
+  void RemoveAcked(UnwrappedTSN cumulative_tsn_ack, AckInfo& ack_info);
+
+  // Will mark the chunks covered by the `gap_ack_blocks` from an incoming SACK
+  // as "acked" and update `ack_info` by adding new TSNs to `added_tsns`.
+  void AckGapBlocks(UnwrappedTSN cumulative_tsn_ack,
+                    rtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
+                    AckInfo& ack_info);
+
+  // Mark chunks reported as "missing", as "nacked" or "to be retransmitted"
+  // depending how many times this has happened. Only packets up until
+  // `ack_info.highest_tsn_acked` (highest TSN newly acknowledged) are
+  // nacked/retransmitted. The method will set `ack_info.has_packet_loss`.
+  void NackBetweenAckBlocks(
+      UnwrappedTSN cumulative_tsn_ack,
+      rtc::ArrayView<const SackChunk::GapAckBlock> gap_ack_blocks,
+      bool is_in_fast_recovery,
+      OutstandingData::AckInfo& ack_info);
+
+  // Process the acknowledgement of the chunk referenced by `iter` and updates
+  // state in `ack_info` and the object's state.
+  void AckChunk(AckInfo& ack_info, std::map<UnwrappedTSN, Item>::iterator iter);
+
+  // Helper method to process an incoming nack of an item and perform the
+  // correct operations given the action indicated when nacking an item (e.g.
+  // retransmitting or abandoning). The return value indicate if an action was
+  // performed, meaning that packet loss was detected and acted upon. If
+  // `do_fast_retransmit` is set and if the item has been nacked sufficiently
+  // many times so that it should be retransmitted, this will schedule it to be
+  // "fast retransmitted". This is only done just before going into fast
+  // recovery.
+  bool NackItem(UnwrappedTSN tsn,
+                Item& item,
+                bool retransmit_now,
+                bool do_fast_retransmit);
+
+  // Given that a message fragment, `item` has been abandoned, abandon all other
+  // fragments that share the same message - both never-before-sent fragments
+  // that are still in the SendQueue and outstanding chunks.
+  void AbandonAllFor(const OutstandingData::Item& item);
+
+  std::vector<std::pair<TSN, Data>> ExtractChunksThatCanFit(
+      std::set<UnwrappedTSN>& chunks,
+      size_t max_size);
+
+  bool IsConsistent() const;
+
+  // The size of the data chunk (DATA/I-DATA) header that is used.
+  const size_t data_chunk_header_size_;
+  // Next TSN to used.
+  UnwrappedTSN next_tsn_;
+  // The last cumulative TSN ack number.
+  UnwrappedTSN last_cumulative_tsn_ack_;
+  // Callback when to discard items from the send queue.
+  std::function<bool(IsUnordered, StreamID, MID)> discard_from_send_queue_;
+
+  std::map<UnwrappedTSN, Item> outstanding_data_;
+  // The number of bytes that are in-flight (sent but not yet acked or nacked).
+  size_t outstanding_bytes_ = 0;
+  // The number of DATA chunks that are in-flight (sent but not yet acked or
+  // nacked).
+  size_t outstanding_items_ = 0;
+  // Data chunks that are eligible for fast retransmission.
+  std::set<UnwrappedTSN> to_be_fast_retransmitted_;
+  // Data chunks that are to be retransmitted.
+  std::set<UnwrappedTSN> to_be_retransmitted_;
+};
+}  // namespace dcsctp
+#endif  // NET_DCSCTP_TX_OUTSTANDING_DATA_H_
diff --git a/third_party/libwebrtc/net/dcsctp/tx/outstanding_data_test.cc b/third_party/libwebrtc/net/dcsctp/tx/outstanding_data_test.cc
new file mode 100644
index 0000000000..cdca40cfef
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/outstanding_data_test.cc
@@ -0,0 +1,591 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/outstanding_data.h"
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "net/dcsctp/common/math.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/testing/data_generator.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::MockFunction;
+using State = ::dcsctp::OutstandingData::State;
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+using ::testing::Pair;
+using ::testing::Return;
+using ::testing::StrictMock;
+
+constexpr TimeMs kNow(42);
+
+class OutstandingDataTest : public testing::Test {
+ protected:
+  OutstandingDataTest()
+      : gen_(MID(42)),
+        buf_(DataChunk::kHeaderSize,
+             unwrapper_.Unwrap(TSN(10)),
+             unwrapper_.Unwrap(TSN(9)),
+             on_discard_.AsStdFunction()) {}
+
+  UnwrappedTSN::Unwrapper unwrapper_;
+  DataGenerator gen_;
+  StrictMock<MockFunction<bool(IsUnordered, StreamID, MID)>> on_discard_;
+  OutstandingData buf_;
+};
+
+TEST_F(OutstandingDataTest, HasInitialState) {
+  EXPECT_TRUE(buf_.empty());
+  EXPECT_EQ(buf_.outstanding_bytes(), 0u);
+  EXPECT_EQ(buf_.outstanding_items(), 0u);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+  EXPECT_EQ(buf_.last_cumulative_tsn_ack().Wrap(), TSN(9));
+  EXPECT_EQ(buf_.next_tsn().Wrap(), TSN(10));
+  EXPECT_EQ(buf_.highest_outstanding_tsn().Wrap(), TSN(9));
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked)));
+  EXPECT_FALSE(buf_.ShouldSendForwardTsn());
+}
+
+TEST_F(OutstandingDataTest, InsertChunk) {
+  ASSERT_HAS_VALUE_AND_ASSIGN(UnwrappedTSN tsn,
+                              buf_.Insert(gen_.Ordered({1}, "BE"), kNow));
+
+  EXPECT_EQ(tsn.Wrap(), TSN(10));
+
+  EXPECT_EQ(buf_.outstanding_bytes(), DataChunk::kHeaderSize + RoundUpTo4(1));
+  EXPECT_EQ(buf_.outstanding_items(), 1u);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+  EXPECT_EQ(buf_.last_cumulative_tsn_ack().Wrap(), TSN(9));
+  EXPECT_EQ(buf_.next_tsn().Wrap(), TSN(11));
+  EXPECT_EQ(buf_.highest_outstanding_tsn().Wrap(), TSN(10));
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),
+                          Pair(TSN(10), State::kInFlight)));
+}
+
+TEST_F(OutstandingDataTest, AcksSingleChunk) {
+  buf_.Insert(gen_.Ordered({1}, "BE"), kNow);
+  OutstandingData::AckInfo ack =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(10)), {}, false);
+
+  EXPECT_EQ(ack.bytes_acked, DataChunk::kHeaderSize + RoundUpTo4(1));
+  EXPECT_EQ(ack.highest_tsn_acked.Wrap(), TSN(10));
+  EXPECT_FALSE(ack.has_packet_loss);
+
+  EXPECT_EQ(buf_.outstanding_bytes(), 0u);
+  EXPECT_EQ(buf_.outstanding_items(), 0u);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+  EXPECT_EQ(buf_.last_cumulative_tsn_ack().Wrap(), TSN(10));
+  EXPECT_EQ(buf_.next_tsn().Wrap(), TSN(11));
+  EXPECT_EQ(buf_.highest_outstanding_tsn().Wrap(), TSN(10));
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(10), State::kAcked)));
+}
+
+TEST_F(OutstandingDataTest, AcksPreviousChunkDoesntUpdate) {
+  buf_.Insert(gen_.Ordered({1}, "BE"), kNow);
+  buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), {}, false);
+
+  EXPECT_EQ(buf_.outstanding_bytes(), DataChunk::kHeaderSize + RoundUpTo4(1));
+  EXPECT_EQ(buf_.outstanding_items(), 1u);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+  EXPECT_EQ(buf_.last_cumulative_tsn_ack().Wrap(), TSN(9));
+  EXPECT_EQ(buf_.next_tsn().Wrap(), TSN(11));
+  EXPECT_EQ(buf_.highest_outstanding_tsn().Wrap(), TSN(10));
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),
+                          Pair(TSN(10), State::kInFlight)));
+}
+
+TEST_F(OutstandingDataTest, AcksAndNacksWithGapAckBlocks) {
+  buf_.Insert(gen_.Ordered({1}, "B"), kNow);
+  buf_.Insert(gen_.Ordered({1}, "E"), kNow);
+
+  std::vector<SackChunk::GapAckBlock> gab = {SackChunk::GapAckBlock(2, 2)};
+  OutstandingData::AckInfo ack =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab, false);
+  EXPECT_EQ(ack.bytes_acked, DataChunk::kHeaderSize + RoundUpTo4(1));
+  EXPECT_EQ(ack.highest_tsn_acked.Wrap(), TSN(11));
+  EXPECT_FALSE(ack.has_packet_loss);
+
+  EXPECT_EQ(buf_.outstanding_bytes(), 0u);
+  EXPECT_EQ(buf_.outstanding_items(), 0u);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+  EXPECT_EQ(buf_.last_cumulative_tsn_ack().Wrap(), TSN(9));
+  EXPECT_EQ(buf_.next_tsn().Wrap(), TSN(12));
+  EXPECT_EQ(buf_.highest_outstanding_tsn().Wrap(), TSN(11));
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),    //
+                          Pair(TSN(10), State::kNacked),  //
+                          Pair(TSN(11), State::kAcked)));
+}
+
+TEST_F(OutstandingDataTest, NacksThreeTimesWithSameTsnDoesntRetransmit) {
+  buf_.Insert(gen_.Ordered({1}, "B"), kNow);
+  buf_.Insert(gen_.Ordered({1}, "E"), kNow);
+
+  std::vector<SackChunk::GapAckBlock> gab1 = {SackChunk::GapAckBlock(2, 2)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab1, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab1, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab1, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),    //
+                          Pair(TSN(10), State::kNacked),  //
+                          Pair(TSN(11), State::kAcked)));
+}
+
+TEST_F(OutstandingDataTest, NacksThreeTimesResultsInRetransmission) {
+  buf_.Insert(gen_.Ordered({1}, "B"), kNow);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow);
+  buf_.Insert(gen_.Ordered({1}, "E"), kNow);
+
+  std::vector<SackChunk::GapAckBlock> gab1 = {SackChunk::GapAckBlock(2, 2)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab1, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab2 = {SackChunk::GapAckBlock(2, 3)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab2, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab3 = {SackChunk::GapAckBlock(2, 4)};
+  OutstandingData::AckInfo ack =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab3, false);
+  EXPECT_EQ(ack.bytes_acked, DataChunk::kHeaderSize + RoundUpTo4(1));
+  EXPECT_EQ(ack.highest_tsn_acked.Wrap(), TSN(13));
+  EXPECT_TRUE(ack.has_packet_loss);
+
+  EXPECT_TRUE(buf_.has_data_to_be_retransmitted());
+
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),               //
+                          Pair(TSN(10), State::kToBeRetransmitted),  //
+                          Pair(TSN(11), State::kAcked),              //
+                          Pair(TSN(12), State::kAcked),              //
+                          Pair(TSN(13), State::kAcked)));
+
+  EXPECT_THAT(buf_.GetChunksToBeFastRetransmitted(1000),
+              ElementsAre(Pair(TSN(10), _)));
+  EXPECT_THAT(buf_.GetChunksToBeRetransmitted(1000), IsEmpty());
+}
+
+TEST_F(OutstandingDataTest, NacksThreeTimesResultsInAbandoning) {
+  static constexpr MaxRetransmits kMaxRetransmissions(0);
+  buf_.Insert(gen_.Ordered({1}, "B"), kNow, kMaxRetransmissions);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, kMaxRetransmissions);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, kMaxRetransmissions);
+  buf_.Insert(gen_.Ordered({1}, "E"), kNow, kMaxRetransmissions);
+
+  std::vector<SackChunk::GapAckBlock> gab1 = {SackChunk::GapAckBlock(2, 2)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab1, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab2 = {SackChunk::GapAckBlock(2, 3)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab2, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  EXPECT_CALL(on_discard_, Call(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(false));
+  std::vector<SackChunk::GapAckBlock> gab3 = {SackChunk::GapAckBlock(2, 4)};
+  OutstandingData::AckInfo ack =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab3, false);
+  EXPECT_EQ(ack.bytes_acked, DataChunk::kHeaderSize + RoundUpTo4(1));
+  EXPECT_EQ(ack.highest_tsn_acked.Wrap(), TSN(13));
+  EXPECT_TRUE(ack.has_packet_loss);
+
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+  EXPECT_EQ(buf_.next_tsn().Wrap(), TSN(14));
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),       //
+                          Pair(TSN(10), State::kAbandoned),  //
+                          Pair(TSN(11), State::kAbandoned),  //
+                          Pair(TSN(12), State::kAbandoned),  //
+                          Pair(TSN(13), State::kAbandoned)));
+}
+
+TEST_F(OutstandingDataTest, NacksThreeTimesResultsInAbandoningWithPlaceholder) {
+  static constexpr MaxRetransmits kMaxRetransmissions(0);
+  buf_.Insert(gen_.Ordered({1}, "B"), kNow, kMaxRetransmissions);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, kMaxRetransmissions);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, kMaxRetransmissions);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, kMaxRetransmissions);
+
+  std::vector<SackChunk::GapAckBlock> gab1 = {SackChunk::GapAckBlock(2, 2)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab1, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab2 = {SackChunk::GapAckBlock(2, 3)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab2, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  EXPECT_CALL(on_discard_, Call(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(true));
+  std::vector<SackChunk::GapAckBlock> gab3 = {SackChunk::GapAckBlock(2, 4)};
+  OutstandingData::AckInfo ack =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab3, false);
+  EXPECT_EQ(ack.bytes_acked, DataChunk::kHeaderSize + RoundUpTo4(1));
+  EXPECT_EQ(ack.highest_tsn_acked.Wrap(), TSN(13));
+  EXPECT_TRUE(ack.has_packet_loss);
+
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+  EXPECT_EQ(buf_.next_tsn().Wrap(), TSN(15));
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),       //
+                          Pair(TSN(10), State::kAbandoned),  //
+                          Pair(TSN(11), State::kAbandoned),  //
+                          Pair(TSN(12), State::kAbandoned),  //
+                          Pair(TSN(13), State::kAbandoned),  //
+                          Pair(TSN(14), State::kAbandoned)));
+}
+
+TEST_F(OutstandingDataTest, ExpiresChunkBeforeItIsInserted) {
+  static constexpr TimeMs kExpiresAt = kNow + DurationMs(1);
+  EXPECT_TRUE(buf_.Insert(gen_.Ordered({1}, "B"), kNow,
+                          MaxRetransmits::NoLimit(), kExpiresAt)
+                  .has_value());
+  EXPECT_TRUE(buf_.Insert(gen_.Ordered({1}, ""), kNow + DurationMs(0),
+                          MaxRetransmits::NoLimit(), kExpiresAt)
+                  .has_value());
+
+  EXPECT_CALL(on_discard_, Call(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(false));
+  EXPECT_FALSE(buf_.Insert(gen_.Ordered({1}, "E"), kNow + DurationMs(1),
+                           MaxRetransmits::NoLimit(), kExpiresAt)
+                   .has_value());
+
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+  EXPECT_EQ(buf_.last_cumulative_tsn_ack().Wrap(), TSN(9));
+  EXPECT_EQ(buf_.next_tsn().Wrap(), TSN(13));
+  EXPECT_EQ(buf_.highest_outstanding_tsn().Wrap(), TSN(12));
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),       //
+                          Pair(TSN(10), State::kAbandoned),  //
+                          Pair(TSN(11), State::kAbandoned),
+                          Pair(TSN(12), State::kAbandoned)));
+}
+
+TEST_F(OutstandingDataTest, CanGenerateForwardTsn) {
+  static constexpr MaxRetransmits kMaxRetransmissions(0);
+  buf_.Insert(gen_.Ordered({1}, "B"), kNow, kMaxRetransmissions);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, kMaxRetransmissions);
+  buf_.Insert(gen_.Ordered({1}, "E"), kNow, kMaxRetransmissions);
+
+  EXPECT_CALL(on_discard_, Call(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(false));
+  buf_.NackAll();
+
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),       //
+                          Pair(TSN(10), State::kAbandoned),  //
+                          Pair(TSN(11), State::kAbandoned),
+                          Pair(TSN(12), State::kAbandoned)));
+
+  EXPECT_TRUE(buf_.ShouldSendForwardTsn());
+  ForwardTsnChunk chunk = buf_.CreateForwardTsn();
+  EXPECT_EQ(chunk.new_cumulative_tsn(), TSN(12));
+}
+
+TEST_F(OutstandingDataTest, AckWithGapBlocksFromRFC4960Section334) {
+  buf_.Insert(gen_.Ordered({1}, "B"), kNow);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow);
+  buf_.Insert(gen_.Ordered({1}, ""), kNow);
+  buf_.Insert(gen_.Ordered({1}, "E"), kNow);
+
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              testing::ElementsAre(Pair(TSN(9), State::kAcked),      //
+                                   Pair(TSN(10), State::kInFlight),  //
+                                   Pair(TSN(11), State::kInFlight),  //
+                                   Pair(TSN(12), State::kInFlight),  //
+                                   Pair(TSN(13), State::kInFlight),  //
+                                   Pair(TSN(14), State::kInFlight),  //
+                                   Pair(TSN(15), State::kInFlight),  //
+                                   Pair(TSN(16), State::kInFlight),  //
+                                   Pair(TSN(17), State::kInFlight)));
+
+  std::vector<SackChunk::GapAckBlock> gab = {SackChunk::GapAckBlock(2, 3),
+                                             SackChunk::GapAckBlock(5, 5)};
+  buf_.HandleSack(unwrapper_.Unwrap(TSN(12)), gab, false);
+
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(12), State::kAcked),   //
+                          Pair(TSN(13), State::kNacked),  //
+                          Pair(TSN(14), State::kAcked),   //
+                          Pair(TSN(15), State::kAcked),   //
+                          Pair(TSN(16), State::kNacked),  //
+                          Pair(TSN(17), State::kAcked)));
+}
+
+TEST_F(OutstandingDataTest, MeasureRTT) {
+  buf_.Insert(gen_.Ordered({1}, "BE"), kNow);
+  buf_.Insert(gen_.Ordered({1}, "BE"), kNow + DurationMs(1));
+  buf_.Insert(gen_.Ordered({1}, "BE"), kNow + DurationMs(2));
+
+  static constexpr DurationMs kDuration(123);
+  ASSERT_HAS_VALUE_AND_ASSIGN(
+      DurationMs duration,
+      buf_.MeasureRTT(kNow + kDuration, unwrapper_.Unwrap(TSN(11))));
+
+  EXPECT_EQ(duration, kDuration - DurationMs(1));
+}
+
+TEST_F(OutstandingDataTest, MustRetransmitBeforeGettingNackedAgain) {
+  // This test case verifies that a chunk that has been nacked, and scheduled to
+  // be retransmitted, doesn't get nacked again until it has been actually sent
+  // on the wire.
+
+  static constexpr MaxRetransmits kOneRetransmission(1);
+  for (int tsn = 10; tsn <= 20; ++tsn) {
+    buf_.Insert(gen_.Ordered({1}, tsn == 10   ? "B"
+                                  : tsn == 20 ? "E"
+                                              : ""),
+                kNow, kOneRetransmission);
+  }
+
+  std::vector<SackChunk::GapAckBlock> gab1 = {SackChunk::GapAckBlock(2, 2)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab1, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab2 = {SackChunk::GapAckBlock(2, 3)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab2, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab3 = {SackChunk::GapAckBlock(2, 4)};
+  OutstandingData::AckInfo ack =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab3, false);
+  EXPECT_TRUE(ack.has_packet_loss);
+  EXPECT_TRUE(buf_.has_data_to_be_retransmitted());
+
+  // Don't call GetChunksToBeRetransmitted yet - simulate that the congestion
+  // window doesn't allow it to be retransmitted yet. It does however get more
+  // SACKs indicating packet loss.
+
+  std::vector<SackChunk::GapAckBlock> gab4 = {SackChunk::GapAckBlock(2, 5)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab4, false).has_packet_loss);
+  EXPECT_TRUE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab5 = {SackChunk::GapAckBlock(2, 6)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab5, false).has_packet_loss);
+  EXPECT_TRUE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab6 = {SackChunk::GapAckBlock(2, 7)};
+  OutstandingData::AckInfo ack2 =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab6, false);
+
+  EXPECT_FALSE(ack2.has_packet_loss);
+  EXPECT_TRUE(buf_.has_data_to_be_retransmitted());
+
+  // Now it's retransmitted.
+  EXPECT_THAT(buf_.GetChunksToBeFastRetransmitted(1000),
+              ElementsAre(Pair(TSN(10), _)));
+  EXPECT_THAT(buf_.GetChunksToBeRetransmitted(1000), IsEmpty());
+
+  // And obviously lost, as it will get NACKed and abandoned.
+  std::vector<SackChunk::GapAckBlock> gab7 = {SackChunk::GapAckBlock(2, 8)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab7, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab8 = {SackChunk::GapAckBlock(2, 9)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab8, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  EXPECT_CALL(on_discard_, Call(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(false));
+
+  std::vector<SackChunk::GapAckBlock> gab9 = {SackChunk::GapAckBlock(2, 10)};
+  OutstandingData::AckInfo ack3 =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab9, false);
+
+  EXPECT_TRUE(ack3.has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+}
+
+TEST_F(OutstandingDataTest, CanAbandonChunksMarkedForFastRetransmit) {
+  // This test is a bit convoluted, and can't really happen with a well behaving
+  // client, but this was found by fuzzers. This test will verify that a message
+  // that was both marked as "to be fast retransmitted" and "abandoned" at the
+  // same time doesn't cause any consistency issues.
+
+  // Add chunks 10-14, but chunk 11 has zero retransmissions. When chunk 10 and
+  // 11 are NACKed three times, chunk 10 will be marked for retransmission, but
+  // chunk 11 will be abandoned, which also abandons chunk 10, as it's part of
+  // the same message.
+  buf_.Insert(gen_.Ordered({1}, "B"), kNow);                    // 10
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, MaxRetransmits(0));  // 11
+  buf_.Insert(gen_.Ordered({1}, ""), kNow);                     // 12
+  buf_.Insert(gen_.Ordered({1}, ""), kNow);                     // 13
+  buf_.Insert(gen_.Ordered({1}, "E"), kNow);                    // 14
+
+  // ACK 9, 12
+  std::vector<SackChunk::GapAckBlock> gab1 = {SackChunk::GapAckBlock(3, 3)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab1, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  // ACK 9, 12, 13
+  std::vector<SackChunk::GapAckBlock> gab2 = {SackChunk::GapAckBlock(3, 4)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab2, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  EXPECT_CALL(on_discard_, Call(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(false));
+
+  // ACK 9, 12, 13, 14
+  std::vector<SackChunk::GapAckBlock> gab3 = {SackChunk::GapAckBlock(3, 5)};
+  OutstandingData::AckInfo ack =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab3, false);
+  EXPECT_TRUE(ack.has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+  EXPECT_THAT(buf_.GetChunksToBeFastRetransmitted(1000), IsEmpty());
+  EXPECT_THAT(buf_.GetChunksToBeRetransmitted(1000), IsEmpty());
+}
+
+TEST_F(OutstandingDataTest, LifecyleReturnsAckedItemsInAckInfo) {
+  buf_.Insert(gen_.Ordered({1}, "BE"), kNow, MaxRetransmits::NoLimit(),
+              TimeMs::InfiniteFuture(), LifecycleId(42));
+  buf_.Insert(gen_.Ordered({1}, "BE"), kNow, MaxRetransmits::NoLimit(),
+              TimeMs::InfiniteFuture(), LifecycleId(43));
+  buf_.Insert(gen_.Ordered({1}, "BE"), kNow, MaxRetransmits::NoLimit(),
+              TimeMs::InfiniteFuture(), LifecycleId(44));
+
+  OutstandingData::AckInfo ack1 =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(11)), {}, false);
+
+  EXPECT_THAT(ack1.acked_lifecycle_ids,
+              ElementsAre(LifecycleId(42), LifecycleId(43)));
+
+  OutstandingData::AckInfo ack2 =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(12)), {}, false);
+
+  EXPECT_THAT(ack2.acked_lifecycle_ids, ElementsAre(LifecycleId(44)));
+}
+
+TEST_F(OutstandingDataTest, LifecycleReturnsAbandonedNackedThreeTimes) {
+  buf_.Insert(gen_.Ordered({1}, "B"), kNow, MaxRetransmits(0));
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, MaxRetransmits(0));
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, MaxRetransmits(0));
+  buf_.Insert(gen_.Ordered({1}, "E"), kNow, MaxRetransmits(0),
+              TimeMs::InfiniteFuture(), LifecycleId(42));
+
+  std::vector<SackChunk::GapAckBlock> gab1 = {SackChunk::GapAckBlock(2, 2)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab1, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab2 = {SackChunk::GapAckBlock(2, 3)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab2, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  std::vector<SackChunk::GapAckBlock> gab3 = {SackChunk::GapAckBlock(2, 4)};
+  EXPECT_CALL(on_discard_, Call(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(false));
+  OutstandingData::AckInfo ack1 =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab3, false);
+  EXPECT_TRUE(ack1.has_packet_loss);
+  EXPECT_THAT(ack1.abandoned_lifecycle_ids, IsEmpty());
+
+  // This will generate a FORWARD-TSN, which is acked
+  EXPECT_TRUE(buf_.ShouldSendForwardTsn());
+  ForwardTsnChunk chunk = buf_.CreateForwardTsn();
+  EXPECT_EQ(chunk.new_cumulative_tsn(), TSN(13));
+
+  OutstandingData::AckInfo ack2 =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(13)), {}, false);
+  EXPECT_FALSE(ack2.has_packet_loss);
+  EXPECT_THAT(ack2.abandoned_lifecycle_ids, ElementsAre(LifecycleId(42)));
+}
+
+TEST_F(OutstandingDataTest, LifecycleReturnsAbandonedAfterT3rtxExpired) {
+  buf_.Insert(gen_.Ordered({1}, "B"), kNow, MaxRetransmits(0));
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, MaxRetransmits(0));
+  buf_.Insert(gen_.Ordered({1}, ""), kNow, MaxRetransmits(0));
+  buf_.Insert(gen_.Ordered({1}, "E"), kNow, MaxRetransmits(0),
+              TimeMs::InfiniteFuture(), LifecycleId(42));
+
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              testing::ElementsAre(Pair(TSN(9), State::kAcked),      //
+                                   Pair(TSN(10), State::kInFlight),  //
+                                   Pair(TSN(11), State::kInFlight),  //
+                                   Pair(TSN(12), State::kInFlight),  //
+                                   Pair(TSN(13), State::kInFlight)));
+
+  std::vector<SackChunk::GapAckBlock> gab1 = {SackChunk::GapAckBlock(2, 4)};
+  EXPECT_FALSE(
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(9)), gab1, false).has_packet_loss);
+  EXPECT_FALSE(buf_.has_data_to_be_retransmitted());
+
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              testing::ElementsAre(Pair(TSN(9), State::kAcked),    //
+                                   Pair(TSN(10), State::kNacked),  //
+                                   Pair(TSN(11), State::kAcked),   //
+                                   Pair(TSN(12), State::kAcked),   //
+                                   Pair(TSN(13), State::kAcked)));
+
+  // T3-rtx triggered.
+  EXPECT_CALL(on_discard_, Call(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(false));
+  buf_.NackAll();
+
+  EXPECT_THAT(buf_.GetChunkStatesForTesting(),
+              testing::ElementsAre(Pair(TSN(9), State::kAcked),       //
+                                   Pair(TSN(10), State::kAbandoned),  //
+                                   Pair(TSN(11), State::kAbandoned),  //
+                                   Pair(TSN(12), State::kAbandoned),  //
+                                   Pair(TSN(13), State::kAbandoned)));
+
+  // This will generate a FORWARD-TSN, which is acked
+  EXPECT_TRUE(buf_.ShouldSendForwardTsn());
+  ForwardTsnChunk chunk = buf_.CreateForwardTsn();
+  EXPECT_EQ(chunk.new_cumulative_tsn(), TSN(13));
+
+  OutstandingData::AckInfo ack2 =
+      buf_.HandleSack(unwrapper_.Unwrap(TSN(13)), {}, false);
+  EXPECT_FALSE(ack2.has_packet_loss);
+  EXPECT_THAT(ack2.abandoned_lifecycle_ids, ElementsAre(LifecycleId(42)));
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/retransmission_error_counter.cc b/third_party/libwebrtc/net/dcsctp/tx/retransmission_error_counter.cc
new file mode 100644
index 0000000000..44b20ba2c2
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/retransmission_error_counter.cc
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/retransmission_error_counter.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+bool RetransmissionErrorCounter::Increment(absl::string_view reason) {
+  ++counter_;
+  if (limit_.has_value() && counter_ > limit_.value()) {
+    RTC_DLOG(LS_INFO) << log_prefix_ << reason
+                      << ", too many retransmissions, counter=" << counter_;
+    return false;
+  }
+
+  RTC_DLOG(LS_VERBOSE) << log_prefix_ << reason << ", new counter=" << counter_
+                       << ", max=" << limit_.value_or(-1);
+  return true;
+}
+
+void RetransmissionErrorCounter::Clear() {
+  if (counter_ > 0) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix_
+                         << "recovered from counter=" << counter_;
+    counter_ = 0;
+  }
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/retransmission_error_counter.h b/third_party/libwebrtc/net/dcsctp/tx/retransmission_error_counter.h
new file mode 100644
index 0000000000..18af3d3c4f
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/retransmission_error_counter.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TX_RETRANSMISSION_ERROR_COUNTER_H_
+#define NET_DCSCTP_TX_RETRANSMISSION_ERROR_COUNTER_H_
+
+#include <functional>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+
+namespace dcsctp {
+
+// The RetransmissionErrorCounter is a simple counter with a limit, and when
+// the limit is exceeded, the counter is exhausted and the connection will
+// be closed. It's incremented on retransmission errors, such as the T3-RTX
+// timer expiring, but also missing heartbeats and stream reset requests.
+class RetransmissionErrorCounter {
+ public:
+  RetransmissionErrorCounter(absl::string_view log_prefix,
+                             const DcSctpOptions& options)
+      : log_prefix_(std::string(log_prefix) + "rtx-errors: "),
+        limit_(options.max_retransmissions) {}
+
+  // Increments the retransmission timer. If the maximum error count has been
+  // reached, `false` will be returned.
+  bool Increment(absl::string_view reason);
+  bool IsExhausted() const { return limit_.has_value() && counter_ > *limit_; }
+
+  // Clears the retransmission errors.
+  void Clear();
+
+  // Returns its current value
+  int value() const { return counter_; }
+
+ private:
+  const std::string log_prefix_;
+  const absl::optional<int> limit_;
+  int counter_ = 0;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TX_RETRANSMISSION_ERROR_COUNTER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/tx/retransmission_error_counter_test.cc b/third_party/libwebrtc/net/dcsctp/tx/retransmission_error_counter_test.cc
new file mode 100644
index 0000000000..67bbc0bec5
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/retransmission_error_counter_test.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/retransmission_error_counter.h"
+
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+TEST(RetransmissionErrorCounterTest, HasInitialValue) {
+  DcSctpOptions options;
+  RetransmissionErrorCounter counter("log: ", options);
+  EXPECT_EQ(counter.value(), 0);
+}
+
+TEST(RetransmissionErrorCounterTest, ReturnsFalseAtMaximumValue) {
+  DcSctpOptions options;
+  options.max_retransmissions = 5;
+  RetransmissionErrorCounter counter("log: ", options);
+  EXPECT_TRUE(counter.Increment("test"));   // 1
+  EXPECT_TRUE(counter.Increment("test"));   // 2
+  EXPECT_TRUE(counter.Increment("test"));   // 3
+  EXPECT_TRUE(counter.Increment("test"));   // 4
+  EXPECT_TRUE(counter.Increment("test"));   // 5
+  EXPECT_FALSE(counter.Increment("test"));  // Too many retransmissions
+}
+
+TEST(RetransmissionErrorCounterTest, CanHandleZeroRetransmission) {
+  DcSctpOptions options;
+  options.max_retransmissions = 0;
+  RetransmissionErrorCounter counter("log: ", options);
+  EXPECT_FALSE(counter.Increment("test"));  // One is too many.
+}
+
+TEST(RetransmissionErrorCounterTest, IsExhaustedAtMaximum) {
+  DcSctpOptions options;
+  options.max_retransmissions = 3;
+  RetransmissionErrorCounter counter("log: ", options);
+  EXPECT_TRUE(counter.Increment("test"));  // 1
+  EXPECT_FALSE(counter.IsExhausted());
+  EXPECT_TRUE(counter.Increment("test"));  // 2
+  EXPECT_FALSE(counter.IsExhausted());
+  EXPECT_TRUE(counter.Increment("test"));  // 3
+  EXPECT_FALSE(counter.IsExhausted());
+  EXPECT_FALSE(counter.Increment("test"));  // Too many retransmissions
+  EXPECT_TRUE(counter.IsExhausted());
+  EXPECT_FALSE(counter.Increment("test"));  // One after too many
+  EXPECT_TRUE(counter.IsExhausted());
+}
+
+TEST(RetransmissionErrorCounterTest, ClearingCounter) {
+  DcSctpOptions options;
+  options.max_retransmissions = 3;
+  RetransmissionErrorCounter counter("log: ", options);
+  EXPECT_TRUE(counter.Increment("test"));  // 1
+  EXPECT_TRUE(counter.Increment("test"));  // 2
+  counter.Clear();
+  EXPECT_TRUE(counter.Increment("test"));  // 1
+  EXPECT_TRUE(counter.Increment("test"));  // 2
+  EXPECT_TRUE(counter.Increment("test"));  // 3
+  EXPECT_FALSE(counter.IsExhausted());
+  EXPECT_FALSE(counter.Increment("test"));  // Too many retransmissions
+  EXPECT_TRUE(counter.IsExhausted());
+}
+
+TEST(RetransmissionErrorCounterTest, CanBeLimitless) {
+  DcSctpOptions options;
+  options.max_retransmissions = absl::nullopt;
+  RetransmissionErrorCounter counter("log: ", options);
+  for (int i = 0; i < 100; ++i) {
+    EXPECT_TRUE(counter.Increment("test"));
+    EXPECT_FALSE(counter.IsExhausted());
+  }
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue.cc b/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue.cc
new file mode 100644
index 0000000000..36e2a859ba
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue.cc
@@ -0,0 +1,611 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/retransmission_queue.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <functional>
+#include <iterator>
+#include <map>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/math.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/common/str_join.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/outstanding_data.h"
+#include "net/dcsctp/tx/send_queue.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+namespace {
+
+// Allow sending only slightly less than an MTU, to account for headers.
+constexpr float kMinBytesRequiredToSendFactor = 0.9;
+}  // namespace
+
+RetransmissionQueue::RetransmissionQueue(
+    absl::string_view log_prefix,
+    DcSctpSocketCallbacks* callbacks,
+    TSN my_initial_tsn,
+    size_t a_rwnd,
+    SendQueue& send_queue,
+    std::function<void(DurationMs rtt)> on_new_rtt,
+    std::function<void()> on_clear_retransmission_counter,
+    Timer& t3_rtx,
+    const DcSctpOptions& options,
+    bool supports_partial_reliability,
+    bool use_message_interleaving)
+    : callbacks_(*callbacks),
+      options_(options),
+      min_bytes_required_to_send_(options.mtu * kMinBytesRequiredToSendFactor),
+      partial_reliability_(supports_partial_reliability),
+      log_prefix_(std::string(log_prefix) + "tx: "),
+      data_chunk_header_size_(use_message_interleaving
+                                  ? IDataChunk::kHeaderSize
+                                  : DataChunk::kHeaderSize),
+      on_new_rtt_(std::move(on_new_rtt)),
+      on_clear_retransmission_counter_(
+          std::move(on_clear_retransmission_counter)),
+      t3_rtx_(t3_rtx),
+      cwnd_(options_.cwnd_mtus_initial * options_.mtu),
+      rwnd_(a_rwnd),
+      // https://tools.ietf.org/html/rfc4960#section-7.2.1
+      // "The initial value of ssthresh MAY be arbitrarily high (for
+      // example, implementations MAY use the size of the receiver advertised
+      // window).""
+      ssthresh_(rwnd_),
+      partial_bytes_acked_(0),
+      send_queue_(send_queue),
+      outstanding_data_(
+          data_chunk_header_size_,
+          tsn_unwrapper_.Unwrap(my_initial_tsn),
+          tsn_unwrapper_.Unwrap(TSN(*my_initial_tsn - 1)),
+          [this](IsUnordered unordered, StreamID stream_id, MID message_id) {
+            return send_queue_.Discard(unordered, stream_id, message_id);
+          }) {}
+
+bool RetransmissionQueue::IsConsistent() const {
+  return true;
+}
+
+// Returns how large a chunk will be, serialized, carrying the data
+size_t RetransmissionQueue::GetSerializedChunkSize(const Data& data) const {
+  return RoundUpTo4(data_chunk_header_size_ + data.size());
+}
+
+void RetransmissionQueue::MaybeExitFastRecovery(
+    UnwrappedTSN cumulative_tsn_ack) {
+  // https://tools.ietf.org/html/rfc4960#section-7.2.4
+  // "When a SACK acknowledges all TSNs up to and including this [fast
+  // recovery] exit point, Fast Recovery is exited."
+  if (fast_recovery_exit_tsn_.has_value() &&
+      cumulative_tsn_ack >= *fast_recovery_exit_tsn_) {
+    RTC_DLOG(LS_VERBOSE) << log_prefix_
+                         << "exit_point=" << *fast_recovery_exit_tsn_->Wrap()
+                         << " reached - exiting fast recovery";
+    fast_recovery_exit_tsn_ = absl::nullopt;
+  }
+}
+
+void RetransmissionQueue::HandleIncreasedCumulativeTsnAck(
+    size_t outstanding_bytes,
+    size_t total_bytes_acked) {
+  // Allow some margin for classifying as fully utilized, due to e.g. that too
+  // small packets (less than kMinimumFragmentedPayload) are not sent +
+  // overhead.
+  bool is_fully_utilized = outstanding_bytes + options_.mtu >= cwnd_;
+  size_t old_cwnd = cwnd_;
+  if (phase() == CongestionAlgorithmPhase::kSlowStart) {
+    if (is_fully_utilized && !is_in_fast_recovery()) {
+      // https://tools.ietf.org/html/rfc4960#section-7.2.1
+      // "Only when these three conditions are met can the cwnd be
+      // increased; otherwise, the cwnd MUST not be increased. If these
+      // conditions are met, then cwnd MUST be increased by, at most, the
+      // lesser of 1) the total size of the previously outstanding DATA
+      // chunk(s) acknowledged, and 2) the destination's path MTU."
+      cwnd_ += std::min(total_bytes_acked, options_.mtu);
+      RTC_DLOG(LS_VERBOSE) << log_prefix_ << "SS increase cwnd=" << cwnd_
+                           << " (" << old_cwnd << ")";
+    }
+  } else if (phase() == CongestionAlgorithmPhase::kCongestionAvoidance) {
+    // https://tools.ietf.org/html/rfc4960#section-7.2.2
+    // "Whenever cwnd is greater than ssthresh, upon each SACK arrival
+    // that advances the Cumulative TSN Ack Point, increase
+    // partial_bytes_acked by the total number of bytes of all new chunks
+    // acknowledged in that SACK including chunks acknowledged by the new
+    // Cumulative TSN Ack and by Gap Ack Blocks."
+    size_t old_pba = partial_bytes_acked_;
+    partial_bytes_acked_ += total_bytes_acked;
+
+    if (partial_bytes_acked_ >= cwnd_ && is_fully_utilized) {
+      // https://tools.ietf.org/html/rfc4960#section-7.2.2
+      // "When partial_bytes_acked is equal to or greater than cwnd and
+      // before the arrival of the SACK the sender had cwnd or more bytes of
+      // data outstanding (i.e., before arrival of the SACK, flightsize was
+      // greater than or equal to cwnd), increase cwnd by MTU, and reset
+      // partial_bytes_acked to (partial_bytes_acked - cwnd)."
+
+      // Errata: https://datatracker.ietf.org/doc/html/rfc8540#section-3.12
+      partial_bytes_acked_ -= cwnd_;
+      cwnd_ += options_.mtu;
+      RTC_DLOG(LS_VERBOSE) << log_prefix_ << "CA increase cwnd=" << cwnd_
+                           << " (" << old_cwnd << ") ssthresh=" << ssthresh_
+                           << ", pba=" << partial_bytes_acked_ << " ("
+                           << old_pba << ")";
+    } else {
+      RTC_DLOG(LS_VERBOSE) << log_prefix_ << "CA unchanged cwnd=" << cwnd_
+                           << " (" << old_cwnd << ") ssthresh=" << ssthresh_
+                           << ", pba=" << partial_bytes_acked_ << " ("
+                           << old_pba << ")";
+    }
+  }
+}
+
+void RetransmissionQueue::HandlePacketLoss(UnwrappedTSN highest_tsn_acked) {
+  if (!is_in_fast_recovery()) {
+    // https://tools.ietf.org/html/rfc4960#section-7.2.4
+    // "If not in Fast Recovery, adjust the ssthresh and cwnd of the
+    // destination address(es) to which the missing DATA chunks were last
+    // sent, according to the formula described in Section 7.2.3."
+    size_t old_cwnd = cwnd_;
+    size_t old_pba = partial_bytes_acked_;
+    ssthresh_ = std::max(cwnd_ / 2, options_.cwnd_mtus_min * options_.mtu);
+    cwnd_ = ssthresh_;
+    partial_bytes_acked_ = 0;
+
+    RTC_DLOG(LS_VERBOSE) << log_prefix_
+                         << "packet loss detected (not fast recovery). cwnd="
+                         << cwnd_ << " (" << old_cwnd
+                         << "), ssthresh=" << ssthresh_
+                         << ", pba=" << partial_bytes_acked_ << " (" << old_pba
+                         << ")";
+
+    // https://tools.ietf.org/html/rfc4960#section-7.2.4
+    // "If not in Fast Recovery, enter Fast Recovery and mark the highest
+    // outstanding TSN as the Fast Recovery exit point."
+    fast_recovery_exit_tsn_ = outstanding_data_.highest_outstanding_tsn();
+    RTC_DLOG(LS_VERBOSE) << log_prefix_
+                         << "fast recovery initiated with exit_point="
+                         << *fast_recovery_exit_tsn_->Wrap();
+  } else {
+    // https://tools.ietf.org/html/rfc4960#section-7.2.4
+    // "While in Fast Recovery, the ssthresh and cwnd SHOULD NOT change for
+    // any destinations due to a subsequent Fast Recovery event (i.e., one
+    // SHOULD NOT reduce the cwnd further due to a subsequent Fast Retransmit)."
+    RTC_DLOG(LS_VERBOSE) << log_prefix_
+                         << "packet loss detected (fast recovery). No changes.";
+  }
+}
+
+void RetransmissionQueue::UpdateReceiverWindow(uint32_t a_rwnd) {
+  rwnd_ = outstanding_data_.outstanding_bytes() >= a_rwnd
+              ? 0
+              : a_rwnd - outstanding_data_.outstanding_bytes();
+}
+
+void RetransmissionQueue::StartT3RtxTimerIfOutstandingData() {
+  // Note: Can't use `outstanding_bytes()` as that one doesn't count chunks to
+  // be retransmitted.
+  if (outstanding_data_.empty()) {
+    // https://tools.ietf.org/html/rfc4960#section-6.3.2
+    // "Whenever all outstanding data sent to an address have been
+    // acknowledged, turn off the T3-rtx timer of that address.
+    // Note: Already stopped in `StopT3RtxTimerOnIncreasedCumulativeTsnAck`."
+  } else {
+    // https://tools.ietf.org/html/rfc4960#section-6.3.2
+    // "Whenever a SACK is received that acknowledges the DATA chunk
+    // with the earliest outstanding TSN for that address, restart the T3-rtx
+    // timer for that address with its current RTO (if there is still
+    // outstanding data on that address)."
+    // "Whenever a SACK is received missing a TSN that was previously
+    // acknowledged via a Gap Ack Block, start the T3-rtx for the destination
+    // address to which the DATA chunk was originally transmitted if it is not
+    // already running."
+    if (!t3_rtx_.is_running()) {
+      t3_rtx_.Start();
+    }
+  }
+}
+
+bool RetransmissionQueue::IsSackValid(const SackChunk& sack) const {
+  // https://tools.ietf.org/html/rfc4960#section-6.2.1
+  // "If Cumulative TSN Ack is less than the Cumulative TSN Ack Point,
+  // then drop the SACK.  Since Cumulative TSN Ack is monotonically increasing,
+  // a SACK whose Cumulative TSN Ack is less than the Cumulative TSN Ack Point
+  // indicates an out-of- order SACK."
+  //
+  // Note: Important not to drop SACKs with identical TSN to that previously
+  // received, as the gap ack blocks or dup tsn fields may have changed.
+  UnwrappedTSN cumulative_tsn_ack =
+      tsn_unwrapper_.PeekUnwrap(sack.cumulative_tsn_ack());
+  if (cumulative_tsn_ack < outstanding_data_.last_cumulative_tsn_ack()) {
+    // https://tools.ietf.org/html/rfc4960#section-6.2.1
+    // "If Cumulative TSN Ack is less than the Cumulative TSN Ack Point,
+    // then drop the SACK.  Since Cumulative TSN Ack is monotonically
+    // increasing, a SACK whose Cumulative TSN Ack is less than the Cumulative
+    // TSN Ack Point indicates an out-of- order SACK."
+    return false;
+  } else if (cumulative_tsn_ack > outstanding_data_.highest_outstanding_tsn()) {
+    return false;
+  }
+  return true;
+}
+
+bool RetransmissionQueue::HandleSack(TimeMs now, const SackChunk& sack) {
+  if (!IsSackValid(sack)) {
+    return false;
+  }
+
+  UnwrappedTSN old_last_cumulative_tsn_ack =
+      outstanding_data_.last_cumulative_tsn_ack();
+  size_t old_outstanding_bytes = outstanding_data_.outstanding_bytes();
+  size_t old_rwnd = rwnd_;
+  UnwrappedTSN cumulative_tsn_ack =
+      tsn_unwrapper_.Unwrap(sack.cumulative_tsn_ack());
+
+  if (sack.gap_ack_blocks().empty()) {
+    UpdateRTT(now, cumulative_tsn_ack);
+  }
+
+  // Exit fast recovery before continuing processing, in case it needs to go
+  // into fast recovery again due to new reported packet loss.
+  MaybeExitFastRecovery(cumulative_tsn_ack);
+
+  OutstandingData::AckInfo ack_info = outstanding_data_.HandleSack(
+      cumulative_tsn_ack, sack.gap_ack_blocks(), is_in_fast_recovery());
+
+  // Add lifecycle events for delivered messages.
+  for (LifecycleId lifecycle_id : ack_info.acked_lifecycle_ids) {
+    RTC_DLOG(LS_VERBOSE) << "Triggering OnLifecycleMessageDelivered("
+                         << lifecycle_id.value() << ")";
+    callbacks_.OnLifecycleMessageDelivered(lifecycle_id);
+    callbacks_.OnLifecycleEnd(lifecycle_id);
+  }
+  for (LifecycleId lifecycle_id : ack_info.abandoned_lifecycle_ids) {
+    RTC_DLOG(LS_VERBOSE) << "Triggering OnLifecycleMessageExpired("
+                         << lifecycle_id.value() << ", true)";
+    callbacks_.OnLifecycleMessageExpired(lifecycle_id,
+                                         /*maybe_delivered=*/true);
+    callbacks_.OnLifecycleEnd(lifecycle_id);
+  }
+
+  // Update of outstanding_data_ is now done. Congestion control remains.
+  UpdateReceiverWindow(sack.a_rwnd());
+
+  RTC_DLOG(LS_VERBOSE) << log_prefix_ << "Received SACK, cum_tsn_ack="
+                       << *cumulative_tsn_ack.Wrap() << " ("
+                       << *old_last_cumulative_tsn_ack.Wrap()
+                       << "), outstanding_bytes="
+                       << outstanding_data_.outstanding_bytes() << " ("
+                       << old_outstanding_bytes << "), rwnd=" << rwnd_ << " ("
+                       << old_rwnd << ")";
+
+  if (cumulative_tsn_ack > old_last_cumulative_tsn_ack) {
+    // https://tools.ietf.org/html/rfc4960#section-6.3.2
+    // "Whenever a SACK is received that acknowledges the DATA chunk
+    // with the earliest outstanding TSN for that address, restart the T3-rtx
+    // timer for that address with its current RTO (if there is still
+    // outstanding data on that address)."
+    // Note: It may be started again in a bit further down.
+    t3_rtx_.Stop();
+
+    HandleIncreasedCumulativeTsnAck(old_outstanding_bytes,
+                                    ack_info.bytes_acked);
+  }
+
+  if (ack_info.has_packet_loss) {
+    HandlePacketLoss(ack_info.highest_tsn_acked);
+  }
+
+  // https://tools.ietf.org/html/rfc4960#section-8.2
+  // "When an outstanding TSN is acknowledged [...] the endpoint shall clear
+  // the error counter ..."
+  if (ack_info.bytes_acked > 0) {
+    on_clear_retransmission_counter_();
+  }
+
+  StartT3RtxTimerIfOutstandingData();
+  RTC_DCHECK(IsConsistent());
+  return true;
+}
+
+void RetransmissionQueue::UpdateRTT(TimeMs now,
+                                    UnwrappedTSN cumulative_tsn_ack) {
+  // RTT updating is flawed in SCTP, as explained in e.g. Pedersen J, Griwodz C,
+  // Halvorsen P (2006) Considerations of SCTP retransmission delays for thin
+  // streams.
+  // Due to delayed acknowledgement, the SACK may be sent much later which
+  // increases the calculated RTT.
+  // TODO(boivie): Consider occasionally sending DATA chunks with I-bit set and
+  // use only those packets for measurement.
+
+  absl::optional<DurationMs> rtt =
+      outstanding_data_.MeasureRTT(now, cumulative_tsn_ack);
+
+  if (rtt.has_value()) {
+    on_new_rtt_(*rtt);
+  }
+}
+
+void RetransmissionQueue::HandleT3RtxTimerExpiry() {
+  size_t old_cwnd = cwnd_;
+  size_t old_outstanding_bytes = outstanding_bytes();
+  // https://tools.ietf.org/html/rfc4960#section-6.3.3
+  // "For the destination address for which the timer expires, adjust
+  // its ssthresh with rules defined in Section 7.2.3 and set the cwnd <- MTU."
+  ssthresh_ = std::max(cwnd_ / 2, 4 * options_.mtu);
+  cwnd_ = 1 * options_.mtu;
+  // Errata: https://datatracker.ietf.org/doc/html/rfc8540#section-3.11
+  partial_bytes_acked_ = 0;
+
+  // https://tools.ietf.org/html/rfc4960#section-6.3.3
+  // "For the destination address for which the timer expires, set RTO
+  // <- RTO * 2 ("back off the timer").  The maximum value discussed in rule C7
+  // above (RTO.max) may be used to provide an upper bound to this doubling
+  // operation."
+
+  // Already done by the Timer implementation.
+
+  // https://tools.ietf.org/html/rfc4960#section-6.3.3
+  // "Determine how many of the earliest (i.e., lowest TSN) outstanding
+  // DATA chunks for the address for which the T3-rtx has expired will fit into
+  // a single packet"
+
+  // https://tools.ietf.org/html/rfc4960#section-6.3.3
+  // "Note: Any DATA chunks that were sent to the address for which the
+  // T3-rtx timer expired but did not fit in one MTU (rule E3 above) should be
+  // marked for retransmission and sent as soon as cwnd allows (normally, when a
+  // SACK arrives)."
+  outstanding_data_.NackAll();
+
+  // https://tools.ietf.org/html/rfc4960#section-6.3.3
+  // "Start the retransmission timer T3-rtx on the destination address
+  // to which the retransmission is sent, if rule R1 above indicates to do so."
+
+  // Already done by the Timer implementation.
+
+  RTC_DLOG(LS_INFO) << log_prefix_ << "t3-rtx expired. new cwnd=" << cwnd_
+                    << " (" << old_cwnd << "), ssthresh=" << ssthresh_
+                    << ", outstanding_bytes " << outstanding_bytes() << " ("
+                    << old_outstanding_bytes << ")";
+  RTC_DCHECK(IsConsistent());
+}
+
+std::vector<std::pair<TSN, Data>>
+RetransmissionQueue::GetChunksForFastRetransmit(size_t bytes_in_packet) {
+  RTC_DCHECK(outstanding_data_.has_data_to_be_fast_retransmitted());
+  RTC_DCHECK(IsDivisibleBy4(bytes_in_packet));
+  std::vector<std::pair<TSN, Data>> to_be_sent;
+  size_t old_outstanding_bytes = outstanding_bytes();
+
+  to_be_sent =
+      outstanding_data_.GetChunksToBeFastRetransmitted(bytes_in_packet);
+  RTC_DCHECK(!to_be_sent.empty());
+
+  // https://tools.ietf.org/html/rfc4960#section-7.2.4
+  // "4)  Restart the T3-rtx timer only if ... the endpoint is retransmitting
+  // the first outstanding DATA chunk sent to that address."
+  if (to_be_sent[0].first ==
+      outstanding_data_.last_cumulative_tsn_ack().next_value().Wrap()) {
+    RTC_DLOG(LS_VERBOSE)
+        << log_prefix_
+        << "First outstanding DATA to be retransmitted - restarting T3-RTX";
+    t3_rtx_.Stop();
+  }
+
+  // https://tools.ietf.org/html/rfc4960#section-6.3.2
+  // "Every time a DATA chunk is sent to any address (including a
+  // retransmission), if the T3-rtx timer of that address is not running,
+  // start it running so that it will expire after the RTO of that address."
+  if (!t3_rtx_.is_running()) {
+    t3_rtx_.Start();
+  }
+  RTC_DLOG(LS_VERBOSE) << log_prefix_ << "Fast-retransmitting TSN "
+                       << StrJoin(to_be_sent, ",",
+                                  [&](rtc::StringBuilder& sb,
+                                      const std::pair<TSN, Data>& c) {
+                                    sb << *c.first;
+                                  })
+                       << " - "
+                       << absl::c_accumulate(
+                              to_be_sent, 0,
+                              [&](size_t r, const std::pair<TSN, Data>& d) {
+                                return r + GetSerializedChunkSize(d.second);
+                              })
+                       << " bytes. outstanding_bytes=" << outstanding_bytes()
+                       << " (" << old_outstanding_bytes << ")";
+
+  RTC_DCHECK(IsConsistent());
+  return to_be_sent;
+}
+
+std::vector<std::pair<TSN, Data>> RetransmissionQueue::GetChunksToSend(
+    TimeMs now,
+    size_t bytes_remaining_in_packet) {
+  // Chunks are always padded to even divisible by four.
+  RTC_DCHECK(IsDivisibleBy4(bytes_remaining_in_packet));
+
+  std::vector<std::pair<TSN, Data>> to_be_sent;
+  size_t old_outstanding_bytes = outstanding_bytes();
+  size_t old_rwnd = rwnd_;
+
+  // Calculate the bandwidth budget (how many bytes that is
+  // allowed to be sent), and fill that up first with chunks that are
+  // scheduled to be retransmitted. If there is still budget, send new chunks
+  // (which will have their TSN assigned here.)
+  size_t max_bytes =
+      RoundDownTo4(std::min(max_bytes_to_send(), bytes_remaining_in_packet));
+
+  to_be_sent = outstanding_data_.GetChunksToBeRetransmitted(max_bytes);
+  max_bytes -= absl::c_accumulate(to_be_sent, 0,
+                                  [&](size_t r, const std::pair<TSN, Data>& d) {
+                                    return r + GetSerializedChunkSize(d.second);
+                                  });
+
+  while (max_bytes > data_chunk_header_size_) {
+    RTC_DCHECK(IsDivisibleBy4(max_bytes));
+    absl::optional<SendQueue::DataToSend> chunk_opt =
+        send_queue_.Produce(now, max_bytes - data_chunk_header_size_);
+    if (!chunk_opt.has_value()) {
+      break;
+    }
+
+    size_t chunk_size = GetSerializedChunkSize(chunk_opt->data);
+    max_bytes -= chunk_size;
+    rwnd_ -= chunk_size;
+
+    absl::optional<UnwrappedTSN> tsn = outstanding_data_.Insert(
+        chunk_opt->data, now,
+        partial_reliability_ ? chunk_opt->max_retransmissions
+                             : MaxRetransmits::NoLimit(),
+        partial_reliability_ ? chunk_opt->expires_at : TimeMs::InfiniteFuture(),
+        chunk_opt->lifecycle_id);
+
+    if (tsn.has_value()) {
+      if (chunk_opt->lifecycle_id.IsSet()) {
+        RTC_DCHECK(chunk_opt->data.is_end);
+        callbacks_.OnLifecycleMessageFullySent(chunk_opt->lifecycle_id);
+      }
+      to_be_sent.emplace_back(tsn->Wrap(), std::move(chunk_opt->data));
+    }
+  }
+
+  if (!to_be_sent.empty()) {
+    // https://tools.ietf.org/html/rfc4960#section-6.3.2
+    // "Every time a DATA chunk is sent to any address (including a
+    // retransmission), if the T3-rtx timer of that address is not running,
+    // start it running so that it will expire after the RTO of that address."
+    if (!t3_rtx_.is_running()) {
+      t3_rtx_.Start();
+    }
+    RTC_DLOG(LS_VERBOSE) << log_prefix_ << "Sending TSN "
+                         << StrJoin(to_be_sent, ",",
+                                    [&](rtc::StringBuilder& sb,
+                                        const std::pair<TSN, Data>& c) {
+                                      sb << *c.first;
+                                    })
+                         << " - "
+                         << absl::c_accumulate(
+                                to_be_sent, 0,
+                                [&](size_t r, const std::pair<TSN, Data>& d) {
+                                  return r + GetSerializedChunkSize(d.second);
+                                })
+                         << " bytes. outstanding_bytes=" << outstanding_bytes()
+                         << " (" << old_outstanding_bytes << "), cwnd=" << cwnd_
+                         << ", rwnd=" << rwnd_ << " (" << old_rwnd << ")";
+  }
+  RTC_DCHECK(IsConsistent());
+  return to_be_sent;
+}
+
+bool RetransmissionQueue::can_send_data() const {
+  return cwnd_ < options_.avoid_fragmentation_cwnd_mtus * options_.mtu ||
+         max_bytes_to_send() >= min_bytes_required_to_send_;
+}
+
+bool RetransmissionQueue::ShouldSendForwardTsn(TimeMs now) {
+  if (!partial_reliability_) {
+    return false;
+  }
+  outstanding_data_.ExpireOutstandingChunks(now);
+  bool ret = outstanding_data_.ShouldSendForwardTsn();
+  RTC_DCHECK(IsConsistent());
+  return ret;
+}
+
+size_t RetransmissionQueue::max_bytes_to_send() const {
+  size_t left = outstanding_bytes() >= cwnd_ ? 0 : cwnd_ - outstanding_bytes();
+
+  if (outstanding_bytes() == 0) {
+    // https://datatracker.ietf.org/doc/html/rfc4960#section-6.1
+    // ... However, regardless of the value of rwnd (including if it is 0), the
+    // data sender can always have one DATA chunk in flight to the receiver if
+    // allowed by cwnd (see rule B, below).
+    return left;
+  }
+
+  return std::min(rwnd(), left);
+}
+
+void RetransmissionQueue::PrepareResetStream(StreamID stream_id) {
+  // TODO(boivie): These calls are now only affecting the send queue. The
+  // packet buffer can also change behavior - for example draining the chunk
+  // producer and eagerly assign TSNs so that an "Outgoing SSN Reset Request"
+  // can be sent quickly, with a known `sender_last_assigned_tsn`.
+  send_queue_.PrepareResetStream(stream_id);
+}
+bool RetransmissionQueue::HasStreamsReadyToBeReset() const {
+  return send_queue_.HasStreamsReadyToBeReset();
+}
+void RetransmissionQueue::CommitResetStreams() {
+  send_queue_.CommitResetStreams();
+}
+void RetransmissionQueue::RollbackResetStreams() {
+  send_queue_.RollbackResetStreams();
+}
+
+HandoverReadinessStatus RetransmissionQueue::GetHandoverReadiness() const {
+  HandoverReadinessStatus status;
+  if (!outstanding_data_.empty()) {
+    status.Add(HandoverUnreadinessReason::kRetransmissionQueueOutstandingData);
+  }
+  if (fast_recovery_exit_tsn_.has_value()) {
+    status.Add(HandoverUnreadinessReason::kRetransmissionQueueFastRecovery);
+  }
+  if (outstanding_data_.has_data_to_be_retransmitted()) {
+    status.Add(HandoverUnreadinessReason::kRetransmissionQueueNotEmpty);
+  }
+  return status;
+}
+
+void RetransmissionQueue::AddHandoverState(DcSctpSocketHandoverState& state) {
+  state.tx.next_tsn = next_tsn().value();
+  state.tx.rwnd = rwnd_;
+  state.tx.cwnd = cwnd_;
+  state.tx.ssthresh = ssthresh_;
+  state.tx.partial_bytes_acked = partial_bytes_acked_;
+}
+
+void RetransmissionQueue::RestoreFromState(
+    const DcSctpSocketHandoverState& state) {
+  // Validate that the component is in pristine state.
+  RTC_DCHECK(outstanding_data_.empty());
+  RTC_DCHECK(!t3_rtx_.is_running());
+  RTC_DCHECK(partial_bytes_acked_ == 0);
+
+  cwnd_ = state.tx.cwnd;
+  rwnd_ = state.tx.rwnd;
+  ssthresh_ = state.tx.ssthresh;
+  partial_bytes_acked_ = state.tx.partial_bytes_acked;
+
+  outstanding_data_.ResetSequenceNumbers(
+      tsn_unwrapper_.Unwrap(TSN(state.tx.next_tsn)),
+      tsn_unwrapper_.Unwrap(TSN(state.tx.next_tsn - 1)));
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue.h b/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue.h
new file mode 100644
index 0000000000..830c0b346d
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue.h
@@ -0,0 +1,257 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TX_RETRANSMISSION_QUEUE_H_
+#define NET_DCSCTP_TX_RETRANSMISSION_QUEUE_H_
+
+#include <cstdint>
+#include <functional>
+#include <map>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/dcsctp_handover_state.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/outstanding_data.h"
+#include "net/dcsctp/tx/retransmission_timeout.h"
+#include "net/dcsctp/tx/send_queue.h"
+
+namespace dcsctp {
+
+// The RetransmissionQueue manages all DATA/I-DATA chunks that are in-flight and
+// schedules them to be retransmitted if necessary. Chunks are retransmitted
+// when they have been lost for a number of consecutive SACKs, or when the
+// retransmission timer, `t3_rtx` expires.
+//
+// As congestion control is tightly connected with the state of transmitted
+// packets, that's also managed here to limit the amount of data that is
+// in-flight (sent, but not yet acknowledged).
+class RetransmissionQueue {
+ public:
+  static constexpr size_t kMinimumFragmentedPayload = 10;
+  using State = OutstandingData::State;
+  // Creates a RetransmissionQueue which will send data using `my_initial_tsn`
+  // (or a value from `DcSctpSocketHandoverState` if given) as the first TSN
+  // to use for sent fragments. It will poll data from `send_queue`. When SACKs
+  // are received, it will estimate the RTT, and call `on_new_rtt`. When an
+  // outstanding chunk has been ACKed, it will call
+  // `on_clear_retransmission_counter` and will also use `t3_rtx`, which is the
+  // SCTP retransmission timer to manage retransmissions.
+  RetransmissionQueue(absl::string_view log_prefix,
+                      DcSctpSocketCallbacks* callbacks,
+                      TSN my_initial_tsn,
+                      size_t a_rwnd,
+                      SendQueue& send_queue,
+                      std::function<void(DurationMs rtt)> on_new_rtt,
+                      std::function<void()> on_clear_retransmission_counter,
+                      Timer& t3_rtx,
+                      const DcSctpOptions& options,
+                      bool supports_partial_reliability = true,
+                      bool use_message_interleaving = false);
+
+  // Handles a received SACK. Returns true if the `sack` was processed and
+  // false if it was discarded due to received out-of-order and not relevant.
+  bool HandleSack(TimeMs now, const SackChunk& sack);
+
+  // Handles an expired retransmission timer.
+  void HandleT3RtxTimerExpiry();
+
+  bool has_data_to_be_fast_retransmitted() const {
+    return outstanding_data_.has_data_to_be_fast_retransmitted();
+  }
+
+  // Returns a list of chunks to "fast retransmit" that would fit in one SCTP
+  // packet with `bytes_in_packet` bytes available. The current value
+  // of `cwnd` is ignored.
+  std::vector<std::pair<TSN, Data>> GetChunksForFastRetransmit(
+      size_t bytes_in_packet);
+
+  // Returns a list of chunks to send that would fit in one SCTP packet with
+  // `bytes_remaining_in_packet` bytes available. This may be further limited by
+  // the congestion control windows. Note that `ShouldSendForwardTSN` must be
+  // called prior to this method, to abandon expired chunks, as this method will
+  // not expire any chunks.
+  std::vector<std::pair<TSN, Data>> GetChunksToSend(
+      TimeMs now,
+      size_t bytes_remaining_in_packet);
+
+  // Returns the internal state of all queued chunks. This is only used in
+  // unit-tests.
+  std::vector<std::pair<TSN, OutstandingData::State>> GetChunkStatesForTesting()
+      const {
+    return outstanding_data_.GetChunkStatesForTesting();
+  }
+
+  // Returns the next TSN that will be allocated for sent DATA chunks.
+  TSN next_tsn() const { return outstanding_data_.next_tsn().Wrap(); }
+
+  // Returns the size of the congestion window, in bytes. This is the number of
+  // bytes that may be in-flight.
+  size_t cwnd() const { return cwnd_; }
+
+  // Overrides the current congestion window size.
+  void set_cwnd(size_t cwnd) { cwnd_ = cwnd; }
+
+  // Returns the current receiver window size.
+  size_t rwnd() const { return rwnd_; }
+
+  // Returns the number of bytes of packets that are in-flight.
+  size_t outstanding_bytes() const {
+    return outstanding_data_.outstanding_bytes();
+  }
+
+  // Returns the number of DATA chunks that are in-flight.
+  size_t outstanding_items() const {
+    return outstanding_data_.outstanding_items();
+  }
+
+  // Indicates if the congestion control algorithm allows data to be sent.
+  bool can_send_data() const;
+
+  // Given the current time `now`, it will evaluate if there are chunks that
+  // have expired and that need to be discarded. It returns true if a
+  // FORWARD-TSN should be sent.
+  bool ShouldSendForwardTsn(TimeMs now);
+
+  // Creates a FORWARD-TSN chunk.
+  ForwardTsnChunk CreateForwardTsn() const {
+    return outstanding_data_.CreateForwardTsn();
+  }
+
+  // Creates an I-FORWARD-TSN chunk.
+  IForwardTsnChunk CreateIForwardTsn() const {
+    return outstanding_data_.CreateIForwardTsn();
+  }
+
+  // See the SendQueue for a longer description of these methods related
+  // to stream resetting.
+  void PrepareResetStream(StreamID stream_id);
+  bool HasStreamsReadyToBeReset() const;
+  std::vector<StreamID> GetStreamsReadyToBeReset() const {
+    return send_queue_.GetStreamsReadyToBeReset();
+  }
+  void CommitResetStreams();
+  void RollbackResetStreams();
+
+  HandoverReadinessStatus GetHandoverReadiness() const;
+
+  void AddHandoverState(DcSctpSocketHandoverState& state);
+  void RestoreFromState(const DcSctpSocketHandoverState& state);
+
+ private:
+  enum class CongestionAlgorithmPhase {
+    kSlowStart,
+    kCongestionAvoidance,
+  };
+
+  bool IsConsistent() const;
+
+  // Returns how large a chunk will be, serialized, carrying the data
+  size_t GetSerializedChunkSize(const Data& data) const;
+
+  // Indicates if the congestion control algorithm is in "fast recovery".
+  bool is_in_fast_recovery() const {
+    return fast_recovery_exit_tsn_.has_value();
+  }
+
+  // Indicates if the provided SACK is valid given what has previously been
+  // received. If it returns false, the SACK is most likely a duplicate of
+  // something already seen, so this returning false doesn't necessarily mean
+  // that the SACK is illegal.
+  bool IsSackValid(const SackChunk& sack) const;
+
+  // When a SACK chunk is received, this method will be called which _may_ call
+  // into the `RetransmissionTimeout` to update the RTO.
+  void UpdateRTT(TimeMs now, UnwrappedTSN cumulative_tsn_ack);
+
+  // If the congestion control is in "fast recovery mode", this may be exited
+  // now.
+  void MaybeExitFastRecovery(UnwrappedTSN cumulative_tsn_ack);
+
+  // If chunks have been ACKed, stop the retransmission timer.
+  void StopT3RtxTimerOnIncreasedCumulativeTsnAck(
+      UnwrappedTSN cumulative_tsn_ack);
+
+  // Update the congestion control algorithm given as the cumulative ack TSN
+  // value has increased, as reported in an incoming SACK chunk.
+  void HandleIncreasedCumulativeTsnAck(size_t outstanding_bytes,
+                                       size_t total_bytes_acked);
+  // Update the congestion control algorithm, given as packet loss has been
+  // detected, as reported in an incoming SACK chunk.
+  void HandlePacketLoss(UnwrappedTSN highest_tsn_acked);
+  // Update the view of the receiver window size.
+  void UpdateReceiverWindow(uint32_t a_rwnd);
+  // If there is data sent and not ACKED, ensure that the retransmission timer
+  // is running.
+  void StartT3RtxTimerIfOutstandingData();
+
+  // Returns the current congestion control algorithm phase.
+  CongestionAlgorithmPhase phase() const {
+    return (cwnd_ <= ssthresh_)
+               ? CongestionAlgorithmPhase::kSlowStart
+               : CongestionAlgorithmPhase::kCongestionAvoidance;
+  }
+
+  // Returns the number of bytes that may be sent in a single packet according
+  // to the congestion control algorithm.
+  size_t max_bytes_to_send() const;
+
+  DcSctpSocketCallbacks& callbacks_;
+  const DcSctpOptions options_;
+  // The minimum bytes required to be available in the congestion window to
+  // allow packets to be sent - to avoid sending too small packets.
+  const size_t min_bytes_required_to_send_;
+  // If the peer supports RFC3758 - SCTP Partial Reliability Extension.
+  const bool partial_reliability_;
+  const std::string log_prefix_;
+  // The size of the data chunk (DATA/I-DATA) header that is used.
+  const size_t data_chunk_header_size_;
+  // Called when a new RTT measurement has been done
+  const std::function<void(DurationMs rtt)> on_new_rtt_;
+  // Called when a SACK has been seen that cleared the retransmission counter.
+  const std::function<void()> on_clear_retransmission_counter_;
+  // The retransmission counter.
+  Timer& t3_rtx_;
+  // Unwraps TSNs
+  UnwrappedTSN::Unwrapper tsn_unwrapper_;
+
+  // Congestion Window. Number of bytes that may be in-flight (sent, not acked).
+  size_t cwnd_;
+  // Receive Window. Number of bytes available in the receiver's RX buffer.
+  size_t rwnd_;
+  // Slow Start Threshold. See RFC4960.
+  size_t ssthresh_;
+  // Partial Bytes Acked. See RFC4960.
+  size_t partial_bytes_acked_;
+  // If set, fast recovery is enabled until this TSN has been cumulative
+  // acked.
+  absl::optional<UnwrappedTSN> fast_recovery_exit_tsn_ = absl::nullopt;
+
+  // The send queue.
+  SendQueue& send_queue_;
+  // All the outstanding data chunks that are in-flight and that have not been
+  // cumulative acked. Note that it also contains chunks that have been acked in
+  // gap ack blocks.
+  OutstandingData outstanding_data_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TX_RETRANSMISSION_QUEUE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue_test.cc b/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue_test.cc
new file mode 100644
index 0000000000..e62c030bfa
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue_test.cc
@@ -0,0 +1,1593 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/retransmission_queue.h"
+
+#include <cstddef>
+#include <cstdint>
+#include <functional>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/common/handover_testing.h"
+#include "net/dcsctp/common/math.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/socket/mock_dcsctp_socket_callbacks.h"
+#include "net/dcsctp/testing/data_generator.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "net/dcsctp/timer/fake_timeout.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/mock_send_queue.h"
+#include "net/dcsctp/tx/send_queue.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::MockFunction;
+using State = ::dcsctp::RetransmissionQueue::State;
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+using ::testing::NiceMock;
+using ::testing::Pair;
+using ::testing::Return;
+using ::testing::SizeIs;
+using ::testing::UnorderedElementsAre;
+
+constexpr uint32_t kArwnd = 100000;
+constexpr uint32_t kMaxMtu = 1191;
+
+DcSctpOptions MakeOptions() {
+  DcSctpOptions options;
+  options.mtu = kMaxMtu;
+  return options;
+}
+
+class RetransmissionQueueTest : public testing::Test {
+ protected:
+  RetransmissionQueueTest()
+      : options_(MakeOptions()),
+        gen_(MID(42)),
+        timeout_manager_([this]() { return now_; }),
+        timer_manager_([this](webrtc::TaskQueueBase::DelayPrecision precision) {
+          return timeout_manager_.CreateTimeout(precision);
+        }),
+        timer_(timer_manager_.CreateTimer(
+            "test/t3_rtx",
+            []() { return absl::nullopt; },
+            TimerOptions(options_.rto_initial))) {}
+
+  std::function<SendQueue::DataToSend(TimeMs, size_t)> CreateChunk() {
+    return [this](TimeMs now, size_t max_size) {
+      return SendQueue::DataToSend(gen_.Ordered({1, 2, 3, 4}, "BE"));
+    };
+  }
+
+  std::vector<TSN> GetTSNsForFastRetransmit(RetransmissionQueue& queue) {
+    std::vector<TSN> tsns;
+    for (const auto& elem : queue.GetChunksForFastRetransmit(10000)) {
+      tsns.push_back(elem.first);
+    }
+    return tsns;
+  }
+
+  std::vector<TSN> GetSentPacketTSNs(RetransmissionQueue& queue) {
+    std::vector<TSN> tsns;
+    for (const auto& elem : queue.GetChunksToSend(now_, 10000)) {
+      tsns.push_back(elem.first);
+    }
+    return tsns;
+  }
+
+  RetransmissionQueue CreateQueue(bool supports_partial_reliability = true,
+                                  bool use_message_interleaving = false) {
+    return RetransmissionQueue(
+        "", &callbacks_, TSN(10), kArwnd, producer_, on_rtt_.AsStdFunction(),
+        on_clear_retransmission_counter_.AsStdFunction(), *timer_, options_,
+        supports_partial_reliability, use_message_interleaving);
+  }
+
+  std::unique_ptr<RetransmissionQueue> CreateQueueByHandover(
+      RetransmissionQueue& queue) {
+    EXPECT_EQ(queue.GetHandoverReadiness(), HandoverReadinessStatus());
+    DcSctpSocketHandoverState state;
+    queue.AddHandoverState(state);
+    g_handover_state_transformer_for_test(&state);
+    auto queue2 = std::make_unique<RetransmissionQueue>(
+        "", &callbacks_, TSN(10), kArwnd, producer_, on_rtt_.AsStdFunction(),
+        on_clear_retransmission_counter_.AsStdFunction(), *timer_, options_,
+        /*supports_partial_reliability=*/true,
+        /*use_message_interleaving=*/false);
+    queue2->RestoreFromState(state);
+    return queue2;
+  }
+
+  MockDcSctpSocketCallbacks callbacks_;
+  DcSctpOptions options_;
+  DataGenerator gen_;
+  TimeMs now_ = TimeMs(0);
+  FakeTimeoutManager timeout_manager_;
+  TimerManager timer_manager_;
+  NiceMock<MockFunction<void(DurationMs rtt_ms)>> on_rtt_;
+  NiceMock<MockFunction<void()>> on_clear_retransmission_counter_;
+  NiceMock<MockSendQueue> producer_;
+  std::unique_ptr<Timer> timer_;
+};
+
+TEST_F(RetransmissionQueueTest, InitialAckedPrevTsn) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked)));
+}
+
+TEST_F(RetransmissionQueueTest, SendOneChunk) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(10)));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kInFlight)));
+}
+
+TEST_F(RetransmissionQueueTest, SendOneChunkAndAck) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(10)));
+
+  queue.HandleSack(now_, SackChunk(TSN(10), kArwnd, {}, {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(10), State::kAcked)));
+}
+
+TEST_F(RetransmissionQueueTest, SendThreeChunksAndAckTwo) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_THAT(GetSentPacketTSNs(queue),
+              testing::ElementsAre(TSN(10), TSN(11), TSN(12)));
+
+  queue.HandleSack(now_, SackChunk(TSN(11), kArwnd, {}, {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(11), State::kAcked),  //
+                          Pair(TSN(12), State::kInFlight)));
+}
+
+TEST_F(RetransmissionQueueTest, AckWithGapBlocksFromRFC4960Section334) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_THAT(GetSentPacketTSNs(queue),
+              testing::ElementsAre(TSN(10), TSN(11), TSN(12), TSN(13), TSN(14),
+                                   TSN(15), TSN(16), TSN(17)));
+
+  queue.HandleSack(now_, SackChunk(TSN(12), kArwnd,
+                                   {SackChunk::GapAckBlock(2, 3),
+                                    SackChunk::GapAckBlock(5, 5)},
+                                   {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(12), State::kAcked),   //
+                          Pair(TSN(13), State::kNacked),  //
+                          Pair(TSN(14), State::kAcked),   //
+                          Pair(TSN(15), State::kAcked),   //
+                          Pair(TSN(16), State::kNacked),  //
+                          Pair(TSN(17), State::kAcked)));
+}
+
+TEST_F(RetransmissionQueueTest, ResendPacketsWhenNackedThreeTimes) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_THAT(GetSentPacketTSNs(queue),
+              testing::ElementsAre(TSN(10), TSN(11), TSN(12), TSN(13), TSN(14),
+                                   TSN(15), TSN(16), TSN(17)));
+
+  // Send more chunks, but leave some as gaps to force retransmission after
+  // three NACKs.
+
+  // Send 18
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(18)));
+
+  // Ack 12, 14-15, 17-18
+  queue.HandleSack(now_, SackChunk(TSN(12), kArwnd,
+                                   {SackChunk::GapAckBlock(2, 3),
+                                    SackChunk::GapAckBlock(5, 6)},
+                                   {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(12), State::kAcked),   //
+                          Pair(TSN(13), State::kNacked),  //
+                          Pair(TSN(14), State::kAcked),   //
+                          Pair(TSN(15), State::kAcked),   //
+                          Pair(TSN(16), State::kNacked),  //
+                          Pair(TSN(17), State::kAcked),   //
+                          Pair(TSN(18), State::kAcked)));
+
+  // Send 19
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(19)));
+
+  // Ack 12, 14-15, 17-19
+  queue.HandleSack(now_, SackChunk(TSN(12), kArwnd,
+                                   {SackChunk::GapAckBlock(2, 3),
+                                    SackChunk::GapAckBlock(5, 7)},
+                                   {}));
+
+  // Send 20
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(20)));
+
+  // Ack 12, 14-15, 17-20
+  queue.HandleSack(now_, SackChunk(TSN(12), kArwnd,
+                                   {SackChunk::GapAckBlock(2, 3),
+                                    SackChunk::GapAckBlock(5, 8)},
+                                   {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(12), State::kAcked),              //
+                          Pair(TSN(13), State::kToBeRetransmitted),  //
+                          Pair(TSN(14), State::kAcked),              //
+                          Pair(TSN(15), State::kAcked),              //
+                          Pair(TSN(16), State::kToBeRetransmitted),  //
+                          Pair(TSN(17), State::kAcked),              //
+                          Pair(TSN(18), State::kAcked),              //
+                          Pair(TSN(19), State::kAcked),              //
+                          Pair(TSN(20), State::kAcked)));
+
+  // This will trigger "fast retransmit" mode and only chunks 13 and 16 will be
+  // resent right now. The send queue will not even be queried.
+  EXPECT_CALL(producer_, Produce).Times(0);
+
+  EXPECT_THAT(GetTSNsForFastRetransmit(queue),
+              testing::ElementsAre(TSN(13), TSN(16)));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(12), State::kAcked),     //
+                          Pair(TSN(13), State::kInFlight),  //
+                          Pair(TSN(14), State::kAcked),     //
+                          Pair(TSN(15), State::kAcked),     //
+                          Pair(TSN(16), State::kInFlight),  //
+                          Pair(TSN(17), State::kAcked),     //
+                          Pair(TSN(18), State::kAcked),     //
+                          Pair(TSN(19), State::kAcked),     //
+                          Pair(TSN(20), State::kAcked)));
+}
+
+TEST_F(RetransmissionQueueTest, RestartsT3RtxOnRetransmitFirstOutstandingTSN) {
+  // Verifies that if fast retransmit is retransmitting the first outstanding
+  // TSN, it will also restart T3-RTX.
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  static constexpr TimeMs kStartTime(100000);
+  now_ = kStartTime;
+
+  EXPECT_THAT(GetSentPacketTSNs(queue),
+              testing::ElementsAre(TSN(10), TSN(11), TSN(12)));
+
+  // Ack 10, 12, after 100ms.
+  now_ += DurationMs(100);
+  queue.HandleSack(
+      now_, SackChunk(TSN(10), kArwnd, {SackChunk::GapAckBlock(2, 2)}, {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(10), State::kAcked),   //
+                          Pair(TSN(11), State::kNacked),  //
+                          Pair(TSN(12), State::kAcked)));
+
+  // Send 13
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(13)));
+
+  // Ack 10, 12-13, after 100ms.
+  now_ += DurationMs(100);
+  queue.HandleSack(
+      now_, SackChunk(TSN(10), kArwnd, {SackChunk::GapAckBlock(2, 3)}, {}));
+
+  // Send 14
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(14)));
+
+  // Ack 10, 12-14, after 100 ms.
+  now_ += DurationMs(100);
+  queue.HandleSack(
+      now_, SackChunk(TSN(10), kArwnd, {SackChunk::GapAckBlock(2, 4)}, {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(10), State::kAcked),              //
+                          Pair(TSN(11), State::kToBeRetransmitted),  //
+                          Pair(TSN(12), State::kAcked),              //
+                          Pair(TSN(13), State::kAcked),              //
+                          Pair(TSN(14), State::kAcked)));
+
+  // This will trigger "fast retransmit" mode and only chunks 13 and 16 will be
+  // resent right now. The send queue will not even be queried.
+  EXPECT_CALL(producer_, Produce).Times(0);
+
+  EXPECT_THAT(GetTSNsForFastRetransmit(queue), testing::ElementsAre(TSN(11)));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(10), State::kAcked),     //
+                          Pair(TSN(11), State::kInFlight),  //
+                          Pair(TSN(12), State::kAcked),     //
+                          Pair(TSN(13), State::kAcked),     //
+                          Pair(TSN(14), State::kAcked)));
+
+  // Verify that the timer was really restarted when fast-retransmitting. The
+  // timeout is `options_.rto_initial`, so advance the time just before that.
+  now_ += options_.rto_initial - DurationMs(1);
+  EXPECT_FALSE(timeout_manager_.GetNextExpiredTimeout().has_value());
+
+  // And ensure it really is running.
+  now_ += DurationMs(1);
+  ASSERT_HAS_VALUE_AND_ASSIGN(TimeoutID timeout,
+                              timeout_manager_.GetNextExpiredTimeout());
+  // An expired timeout has to be handled (asserts validate this).
+  timer_manager_.HandleTimeout(timeout);
+}
+
+TEST_F(RetransmissionQueueTest, CanOnlyProduceTwoPacketsButWantsToSendThree) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        return SendQueue::DataToSend(gen_.Ordered({1, 2, 3, 4}, "BE"));
+      })
+      .WillOnce([this](TimeMs, size_t) {
+        return SendQueue::DataToSend(gen_.Ordered({1, 2, 3, 4}, "BE"));
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _), Pair(TSN(11), _)));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kInFlight),  //
+                          Pair(TSN(11), State::kInFlight)));
+}
+
+TEST_F(RetransmissionQueueTest, RetransmitsOnT3Expiry) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        return SendQueue::DataToSend(gen_.Ordered({1, 2, 3, 4}, "BE"));
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kInFlight)));
+
+  // Will force chunks to be retransmitted
+  queue.HandleT3RtxTimerExpiry();
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kToBeRetransmitted)));
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kToBeRetransmitted)));
+
+  std::vector<std::pair<TSN, Data>> chunks_to_rtx =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_rtx, ElementsAre(Pair(TSN(10), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kInFlight)));
+}
+
+TEST_F(RetransmissionQueueTest, LimitedRetransmissionOnlyWithRfc3758Support) {
+  RetransmissionQueue queue =
+      CreateQueue(/*supports_partial_reliability=*/false);
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "BE"));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kInFlight)));
+
+  // Will force chunks to be retransmitted
+  queue.HandleT3RtxTimerExpiry();
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kToBeRetransmitted)));
+
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .Times(0);
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+}  // namespace dcsctp
+
+TEST_F(RetransmissionQueueTest, LimitsRetransmissionsAsUdp) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "BE"));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kInFlight)));
+
+  // Will force chunks to be retransmitted
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .Times(1);
+
+  queue.HandleT3RtxTimerExpiry();
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kAbandoned)));
+
+  EXPECT_TRUE(queue.ShouldSendForwardTsn(now_));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kAbandoned)));
+
+  std::vector<std::pair<TSN, Data>> chunks_to_rtx =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_rtx, testing::IsEmpty());
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kAbandoned)));
+}
+
+TEST_F(RetransmissionQueueTest, LimitsRetransmissionsToThreeSends) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "BE"));
+        dts.max_retransmissions = MaxRetransmits(3);
+        return dts;
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kInFlight)));
+
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .Times(0);
+
+  // Retransmission 1
+  queue.HandleT3RtxTimerExpiry();
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+  EXPECT_THAT(queue.GetChunksToSend(now_, 1000), SizeIs(1));
+
+  // Retransmission 2
+  queue.HandleT3RtxTimerExpiry();
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+  EXPECT_THAT(queue.GetChunksToSend(now_, 1000), SizeIs(1));
+
+  // Retransmission 3
+  queue.HandleT3RtxTimerExpiry();
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+  EXPECT_THAT(queue.GetChunksToSend(now_, 1000), SizeIs(1));
+
+  // Retransmission 4 - not allowed.
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .Times(1);
+  queue.HandleT3RtxTimerExpiry();
+  EXPECT_TRUE(queue.ShouldSendForwardTsn(now_));
+  EXPECT_THAT(queue.GetChunksToSend(now_, 1000), IsEmpty());
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kAbandoned)));
+}
+
+TEST_F(RetransmissionQueueTest, RetransmitsWhenSendBufferIsFullT3Expiry) {
+  RetransmissionQueue queue = CreateQueue();
+  static constexpr size_t kCwnd = 1200;
+  queue.set_cwnd(kCwnd);
+  EXPECT_EQ(queue.cwnd(), kCwnd);
+  EXPECT_EQ(queue.outstanding_bytes(), 0u);
+  EXPECT_EQ(queue.outstanding_items(), 0u);
+
+  std::vector<uint8_t> payload(1000);
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this, payload](TimeMs, size_t) {
+        return SendQueue::DataToSend(gen_.Ordered(payload, "BE"));
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1500);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kInFlight)));
+  EXPECT_EQ(queue.outstanding_bytes(), payload.size() + DataChunk::kHeaderSize);
+  EXPECT_EQ(queue.outstanding_items(), 1u);
+
+  // Will force chunks to be retransmitted
+  queue.HandleT3RtxTimerExpiry();
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kToBeRetransmitted)));
+  EXPECT_EQ(queue.outstanding_bytes(), 0u);
+  EXPECT_EQ(queue.outstanding_items(), 0u);
+
+  std::vector<std::pair<TSN, Data>> chunks_to_rtx =
+      queue.GetChunksToSend(now_, 1500);
+  EXPECT_THAT(chunks_to_rtx, ElementsAre(Pair(TSN(10), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),  //
+                          Pair(TSN(10), State::kInFlight)));
+  EXPECT_EQ(queue.outstanding_bytes(), payload.size() + DataChunk::kHeaderSize);
+  EXPECT_EQ(queue.outstanding_items(), 1u);
+}
+
+TEST_F(RetransmissionQueueTest, ProducesValidForwardTsn) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "B"));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({5, 6, 7, 8}, ""));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({9, 10, 11, 12}, ""));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  // Send and ack first chunk (TSN 10)
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _), Pair(TSN(11), _),
+                                          Pair(TSN(12), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kInFlight),  //
+                          Pair(TSN(11), State::kInFlight),  //
+                          Pair(TSN(12), State::kInFlight)));
+
+  // Chunk 10 is acked, but the remaining are lost
+  queue.HandleSack(now_, SackChunk(TSN(10), kArwnd, {}, {}));
+
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(true));
+
+  queue.HandleT3RtxTimerExpiry();
+
+  // NOTE: The TSN=13 represents the end fragment.
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(10), State::kAcked),      //
+                          Pair(TSN(11), State::kAbandoned),  //
+                          Pair(TSN(12), State::kAbandoned),  //
+                          Pair(TSN(13), State::kAbandoned)));
+
+  EXPECT_TRUE(queue.ShouldSendForwardTsn(now_));
+
+  ForwardTsnChunk forward_tsn = queue.CreateForwardTsn();
+  EXPECT_EQ(forward_tsn.new_cumulative_tsn(), TSN(13));
+  EXPECT_THAT(forward_tsn.skipped_streams(),
+              UnorderedElementsAre(
+                  ForwardTsnChunk::SkippedStream(StreamID(1), SSN(42))));
+}
+
+TEST_F(RetransmissionQueueTest, ProducesValidForwardTsnWhenFullySent) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "B"));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({5, 6, 7, 8}, ""));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({9, 10, 11, 12}, "E"));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  // Send and ack first chunk (TSN 10)
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _), Pair(TSN(11), _),
+                                          Pair(TSN(12), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kInFlight),  //
+                          Pair(TSN(11), State::kInFlight),  //
+                          Pair(TSN(12), State::kInFlight)));
+
+  // Chunk 10 is acked, but the remaining are lost
+  queue.HandleSack(now_, SackChunk(TSN(10), kArwnd, {}, {}));
+
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(false));
+
+  queue.HandleT3RtxTimerExpiry();
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(10), State::kAcked),      //
+                          Pair(TSN(11), State::kAbandoned),  //
+                          Pair(TSN(12), State::kAbandoned)));
+
+  EXPECT_TRUE(queue.ShouldSendForwardTsn(now_));
+
+  ForwardTsnChunk forward_tsn = queue.CreateForwardTsn();
+  EXPECT_EQ(forward_tsn.new_cumulative_tsn(), TSN(12));
+  EXPECT_THAT(forward_tsn.skipped_streams(),
+              UnorderedElementsAre(
+                  ForwardTsnChunk::SkippedStream(StreamID(1), SSN(42))));
+}
+
+TEST_F(RetransmissionQueueTest, ProducesValidIForwardTsn) {
+  RetransmissionQueue queue = CreateQueue(/*use_message_interleaving=*/true);
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        DataGeneratorOptions opts;
+        opts.stream_id = StreamID(1);
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "B", opts));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillOnce([this](TimeMs, size_t) {
+        DataGeneratorOptions opts;
+        opts.stream_id = StreamID(2);
+        SendQueue::DataToSend dts(gen_.Unordered({1, 2, 3, 4}, "B", opts));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillOnce([this](TimeMs, size_t) {
+        DataGeneratorOptions opts;
+        opts.stream_id = StreamID(3);
+        SendQueue::DataToSend dts(gen_.Ordered({9, 10, 11, 12}, "B", opts));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillOnce([this](TimeMs, size_t) {
+        DataGeneratorOptions opts;
+        opts.stream_id = StreamID(4);
+        SendQueue::DataToSend dts(gen_.Ordered({13, 14, 15, 16}, "B", opts));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _), Pair(TSN(11), _),
+                                          Pair(TSN(12), _), Pair(TSN(13), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kInFlight),  //
+                          Pair(TSN(11), State::kInFlight),  //
+                          Pair(TSN(12), State::kInFlight),  //
+                          Pair(TSN(13), State::kInFlight)));
+
+  // Chunk 13 is acked, but the remaining are lost
+  queue.HandleSack(
+      now_, SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(4, 4)}, {}));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),    //
+                          Pair(TSN(10), State::kNacked),  //
+                          Pair(TSN(11), State::kNacked),  //
+                          Pair(TSN(12), State::kNacked),  //
+                          Pair(TSN(13), State::kAcked)));
+
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(true));
+  EXPECT_CALL(producer_, Discard(IsUnordered(true), StreamID(2), MID(42)))
+      .WillOnce(Return(true));
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(3), MID(42)))
+      .WillOnce(Return(true));
+
+  queue.HandleT3RtxTimerExpiry();
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),       //
+                          Pair(TSN(10), State::kAbandoned),  //
+                          Pair(TSN(11), State::kAbandoned),  //
+                          Pair(TSN(12), State::kAbandoned),  //
+                          Pair(TSN(13), State::kAcked),
+                          // Representing end fragments of stream 1-3
+                          Pair(TSN(14), State::kAbandoned),  //
+                          Pair(TSN(15), State::kAbandoned),  //
+                          Pair(TSN(16), State::kAbandoned)));
+
+  EXPECT_TRUE(queue.ShouldSendForwardTsn(now_));
+
+  IForwardTsnChunk forward_tsn1 = queue.CreateIForwardTsn();
+  EXPECT_EQ(forward_tsn1.new_cumulative_tsn(), TSN(12));
+  EXPECT_THAT(
+      forward_tsn1.skipped_streams(),
+      UnorderedElementsAre(IForwardTsnChunk::SkippedStream(
+                               IsUnordered(false), StreamID(1), MID(42)),
+                           IForwardTsnChunk::SkippedStream(
+                               IsUnordered(true), StreamID(2), MID(42)),
+                           IForwardTsnChunk::SkippedStream(
+                               IsUnordered(false), StreamID(3), MID(42))));
+
+  // When TSN 13 is acked, the placeholder "end fragments" must be skipped as
+  // well.
+
+  // A receiver is more likely to ack TSN 13, but do it incrementally.
+  queue.HandleSack(now_, SackChunk(TSN(12), kArwnd, {}, {}));
+
+  EXPECT_CALL(producer_, Discard).Times(0);
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+
+  queue.HandleSack(now_, SackChunk(TSN(13), kArwnd, {}, {}));
+  EXPECT_TRUE(queue.ShouldSendForwardTsn(now_));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(13), State::kAcked),      //
+                          Pair(TSN(14), State::kAbandoned),  //
+                          Pair(TSN(15), State::kAbandoned),  //
+                          Pair(TSN(16), State::kAbandoned)));
+
+  IForwardTsnChunk forward_tsn2 = queue.CreateIForwardTsn();
+  EXPECT_EQ(forward_tsn2.new_cumulative_tsn(), TSN(16));
+  EXPECT_THAT(
+      forward_tsn2.skipped_streams(),
+      UnorderedElementsAre(IForwardTsnChunk::SkippedStream(
+                               IsUnordered(false), StreamID(1), MID(42)),
+                           IForwardTsnChunk::SkippedStream(
+                               IsUnordered(true), StreamID(2), MID(42)),
+                           IForwardTsnChunk::SkippedStream(
+                               IsUnordered(false), StreamID(3), MID(42))));
+}
+
+TEST_F(RetransmissionQueueTest, MeasureRTT) {
+  RetransmissionQueue queue = CreateQueue(/*use_message_interleaving=*/true);
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "B"));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _)));
+
+  now_ = now_ + DurationMs(123);
+
+  EXPECT_CALL(on_rtt_, Call(DurationMs(123))).Times(1);
+  queue.HandleSack(now_, SackChunk(TSN(10), kArwnd, {}, {}));
+}
+
+TEST_F(RetransmissionQueueTest, ValidateCumTsnAtRest) {
+  RetransmissionQueue queue = CreateQueue(/*use_message_interleaving=*/true);
+
+  EXPECT_FALSE(queue.HandleSack(now_, SackChunk(TSN(8), kArwnd, {}, {})));
+  EXPECT_TRUE(queue.HandleSack(now_, SackChunk(TSN(9), kArwnd, {}, {})));
+  EXPECT_FALSE(queue.HandleSack(now_, SackChunk(TSN(10), kArwnd, {}, {})));
+}
+
+TEST_F(RetransmissionQueueTest, ValidateCumTsnAckOnInflightData) {
+  RetransmissionQueue queue = CreateQueue();
+
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_THAT(GetSentPacketTSNs(queue),
+              testing::ElementsAre(TSN(10), TSN(11), TSN(12), TSN(13), TSN(14),
+                                   TSN(15), TSN(16), TSN(17)));
+
+  EXPECT_FALSE(queue.HandleSack(now_, SackChunk(TSN(8), kArwnd, {}, {})));
+  EXPECT_TRUE(queue.HandleSack(now_, SackChunk(TSN(9), kArwnd, {}, {})));
+  EXPECT_TRUE(queue.HandleSack(now_, SackChunk(TSN(10), kArwnd, {}, {})));
+  EXPECT_TRUE(queue.HandleSack(now_, SackChunk(TSN(11), kArwnd, {}, {})));
+  EXPECT_TRUE(queue.HandleSack(now_, SackChunk(TSN(12), kArwnd, {}, {})));
+  EXPECT_TRUE(queue.HandleSack(now_, SackChunk(TSN(13), kArwnd, {}, {})));
+  EXPECT_TRUE(queue.HandleSack(now_, SackChunk(TSN(14), kArwnd, {}, {})));
+  EXPECT_TRUE(queue.HandleSack(now_, SackChunk(TSN(15), kArwnd, {}, {})));
+  EXPECT_TRUE(queue.HandleSack(now_, SackChunk(TSN(16), kArwnd, {}, {})));
+  EXPECT_TRUE(queue.HandleSack(now_, SackChunk(TSN(17), kArwnd, {}, {})));
+  EXPECT_FALSE(queue.HandleSack(now_, SackChunk(TSN(18), kArwnd, {}, {})));
+}
+
+TEST_F(RetransmissionQueueTest, HandleGapAckBlocksMatchingNoInflightData) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_THAT(GetSentPacketTSNs(queue),
+              testing::ElementsAre(TSN(10), TSN(11), TSN(12), TSN(13), TSN(14),
+                                   TSN(15), TSN(16), TSN(17)));
+
+  // Ack 9, 20-25. This is an invalid SACK, but should still be handled.
+  queue.HandleSack(
+      now_, SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(11, 16)}, {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kInFlight),  //
+                          Pair(TSN(11), State::kInFlight),  //
+                          Pair(TSN(12), State::kInFlight),  //
+                          Pair(TSN(13), State::kInFlight),  //
+                          Pair(TSN(14), State::kInFlight),  //
+                          Pair(TSN(15), State::kInFlight),  //
+                          Pair(TSN(16), State::kInFlight),  //
+                          Pair(TSN(17), State::kInFlight)));
+}
+
+TEST_F(RetransmissionQueueTest, HandleInvalidGapAckBlocks) {
+  RetransmissionQueue queue = CreateQueue();
+
+  // Nothing produced - nothing in retransmission queue
+
+  // Ack 9, 12-13
+  queue.HandleSack(
+      now_, SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(3, 4)}, {}));
+
+  // Gap ack blocks are just ignore.
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked)));
+}
+
+TEST_F(RetransmissionQueueTest, GapAckBlocksDoNotMoveCumTsnAck) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_THAT(GetSentPacketTSNs(queue),
+              testing::ElementsAre(TSN(10), TSN(11), TSN(12), TSN(13), TSN(14),
+                                   TSN(15), TSN(16), TSN(17)));
+
+  // Ack 9, 10-14. This is actually an invalid ACK as the first gap can't be
+  // adjacent to the cum-tsn-ack, but it's not strictly forbidden. However, the
+  // cum-tsn-ack should not move, as the gap-ack-blocks are just advisory.
+  queue.HandleSack(
+      now_, SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(1, 5)}, {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kAcked),     //
+                          Pair(TSN(11), State::kAcked),     //
+                          Pair(TSN(12), State::kAcked),     //
+                          Pair(TSN(13), State::kAcked),     //
+                          Pair(TSN(14), State::kAcked),     //
+                          Pair(TSN(15), State::kInFlight),  //
+                          Pair(TSN(16), State::kInFlight),  //
+                          Pair(TSN(17), State::kInFlight)));
+}
+
+TEST_F(RetransmissionQueueTest, StaysWithinAvailableSize) {
+  RetransmissionQueue queue = CreateQueue();
+
+  // See SctpPacketTest::ReturnsCorrectSpaceAvailableToStayWithinMTU for the
+  // magic numbers in this test.
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t size) {
+        EXPECT_EQ(size, 1176 - DataChunk::kHeaderSize);
+
+        std::vector<uint8_t> payload(183);
+        return SendQueue::DataToSend(gen_.Ordered(payload, "BE"));
+      })
+      .WillOnce([this](TimeMs, size_t size) {
+        EXPECT_EQ(size, 976 - DataChunk::kHeaderSize);
+
+        std::vector<uint8_t> payload(957);
+        return SendQueue::DataToSend(gen_.Ordered(payload, "BE"));
+      });
+
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1188 - 12);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _), Pair(TSN(11), _)));
+}
+
+TEST_F(RetransmissionQueueTest, AccountsNackedAbandonedChunksAsNotOutstanding) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "B"));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({5, 6, 7, 8}, ""));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({9, 10, 11, 12}, ""));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  // Send and ack first chunk (TSN 10)
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _), Pair(TSN(11), _),
+                                          Pair(TSN(12), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kInFlight),  //
+                          Pair(TSN(11), State::kInFlight),  //
+                          Pair(TSN(12), State::kInFlight)));
+  EXPECT_EQ(queue.outstanding_bytes(), (16 + 4) * 3u);
+  EXPECT_EQ(queue.outstanding_items(), 3u);
+
+  // Mark the message as lost.
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .Times(1);
+  queue.HandleT3RtxTimerExpiry();
+
+  EXPECT_TRUE(queue.ShouldSendForwardTsn(now_));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),       //
+                          Pair(TSN(10), State::kAbandoned),  //
+                          Pair(TSN(11), State::kAbandoned),  //
+                          Pair(TSN(12), State::kAbandoned)));
+  EXPECT_EQ(queue.outstanding_bytes(), 0u);
+  EXPECT_EQ(queue.outstanding_items(), 0u);
+
+  // Now ACK those, one at a time.
+  queue.HandleSack(now_, SackChunk(TSN(10), kArwnd, {}, {}));
+  EXPECT_EQ(queue.outstanding_bytes(), 0u);
+  EXPECT_EQ(queue.outstanding_items(), 0u);
+
+  queue.HandleSack(now_, SackChunk(TSN(11), kArwnd, {}, {}));
+  EXPECT_EQ(queue.outstanding_bytes(), 0u);
+  EXPECT_EQ(queue.outstanding_items(), 0u);
+
+  queue.HandleSack(now_, SackChunk(TSN(12), kArwnd, {}, {}));
+  EXPECT_EQ(queue.outstanding_bytes(), 0u);
+  EXPECT_EQ(queue.outstanding_items(), 0u);
+}
+
+TEST_F(RetransmissionQueueTest, ExpireFromSendQueueWhenPartiallySent) {
+  RetransmissionQueue queue = CreateQueue();
+  DataGeneratorOptions options;
+  options.stream_id = StreamID(17);
+  options.message_id = MID(42);
+  TimeMs test_start = now_;
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([&](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "B", options));
+        dts.expires_at = TimeMs(test_start + DurationMs(10));
+        return dts;
+      })
+      .WillOnce([&](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({5, 6, 7, 8}, "", options));
+        dts.expires_at = TimeMs(test_start + DurationMs(10));
+        return dts;
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 24);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _)));
+
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(17), MID(42)))
+      .WillOnce(Return(true));
+  now_ += DurationMs(100);
+
+  EXPECT_THAT(queue.GetChunksToSend(now_, 24), IsEmpty());
+
+  EXPECT_THAT(
+      queue.GetChunkStatesForTesting(),
+      ElementsAre(Pair(TSN(9), State::kAcked),         // Initial TSN
+                  Pair(TSN(10), State::kAbandoned),    // Produced
+                  Pair(TSN(11), State::kAbandoned),    // Produced and expired
+                  Pair(TSN(12), State::kAbandoned)));  // Placeholder end
+}
+
+TEST_F(RetransmissionQueueTest, LimitsRetransmissionsOnlyWhenNackedThreeTimes) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "BE"));
+        dts.max_retransmissions = MaxRetransmits(0);
+        return dts;
+      })
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _), Pair(TSN(11), _),
+                                          Pair(TSN(12), _), Pair(TSN(13), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kInFlight),  //
+                          Pair(TSN(11), State::kInFlight),  //
+                          Pair(TSN(12), State::kInFlight),  //
+                          Pair(TSN(13), State::kInFlight)));
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .Times(0);
+
+  queue.HandleSack(
+      now_, SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(2, 2)}, {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kNacked),    //
+                          Pair(TSN(11), State::kAcked),     //
+                          Pair(TSN(12), State::kInFlight),  //
+                          Pair(TSN(13), State::kInFlight)));
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+
+  queue.HandleSack(
+      now_, SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(2, 3)}, {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),    //
+                          Pair(TSN(10), State::kNacked),  //
+                          Pair(TSN(11), State::kAcked),   //
+                          Pair(TSN(12), State::kAcked),   //
+                          Pair(TSN(13), State::kInFlight)));
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(false));
+  queue.HandleSack(
+      now_, SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(2, 4)}, {}));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),       //
+                          Pair(TSN(10), State::kAbandoned),  //
+                          Pair(TSN(11), State::kAcked),      //
+                          Pair(TSN(12), State::kAcked),      //
+                          Pair(TSN(13), State::kAcked)));
+
+  EXPECT_TRUE(queue.ShouldSendForwardTsn(now_));
+}
+
+TEST_F(RetransmissionQueueTest, AbandonsRtxLimit2WhenNackedNineTimes) {
+  // This is a fairly long test.
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this](TimeMs, size_t) {
+        SendQueue::DataToSend dts(gen_.Ordered({1, 2, 3, 4}, "BE"));
+        dts.max_retransmissions = MaxRetransmits(2);
+        return dts;
+      })
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1000);
+  EXPECT_THAT(chunks_to_send,
+              ElementsAre(Pair(TSN(10), _), Pair(TSN(11), _), Pair(TSN(12), _),
+                          Pair(TSN(13), _), Pair(TSN(14), _), Pair(TSN(15), _),
+                          Pair(TSN(16), _), Pair(TSN(17), _), Pair(TSN(18), _),
+                          Pair(TSN(19), _)));
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kInFlight),  //
+                          Pair(TSN(11), State::kInFlight),  //
+                          Pair(TSN(12), State::kInFlight),  //
+                          Pair(TSN(13), State::kInFlight),  //
+                          Pair(TSN(14), State::kInFlight),  //
+                          Pair(TSN(15), State::kInFlight),  //
+                          Pair(TSN(16), State::kInFlight),  //
+                          Pair(TSN(17), State::kInFlight),  //
+                          Pair(TSN(18), State::kInFlight),  //
+                          Pair(TSN(19), State::kInFlight)));
+
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .Times(0);
+
+  // Ack TSN [11 to 13] - three nacks for TSN(10), which will retransmit it.
+  for (int tsn = 11; tsn <= 13; ++tsn) {
+    queue.HandleSack(
+        now_,
+        SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(2, (tsn - 9))}, {}));
+  }
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),               //
+                          Pair(TSN(10), State::kToBeRetransmitted),  //
+                          Pair(TSN(11), State::kAcked),              //
+                          Pair(TSN(12), State::kAcked),              //
+                          Pair(TSN(13), State::kAcked),              //
+                          Pair(TSN(14), State::kInFlight),           //
+                          Pair(TSN(15), State::kInFlight),           //
+                          Pair(TSN(16), State::kInFlight),           //
+                          Pair(TSN(17), State::kInFlight),           //
+                          Pair(TSN(18), State::kInFlight),           //
+                          Pair(TSN(19), State::kInFlight)));
+
+  EXPECT_THAT(queue.GetChunksForFastRetransmit(1000),
+              ElementsAre(Pair(TSN(10), _)));
+
+  // Ack TSN [14 to 16] - three more nacks - second and last retransmission.
+  for (int tsn = 14; tsn <= 16; ++tsn) {
+    queue.HandleSack(
+        now_,
+        SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(2, (tsn - 9))}, {}));
+  }
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),               //
+                          Pair(TSN(10), State::kToBeRetransmitted),  //
+                          Pair(TSN(11), State::kAcked),              //
+                          Pair(TSN(12), State::kAcked),              //
+                          Pair(TSN(13), State::kAcked),              //
+                          Pair(TSN(14), State::kAcked),              //
+                          Pair(TSN(15), State::kAcked),              //
+                          Pair(TSN(16), State::kAcked),              //
+                          Pair(TSN(17), State::kInFlight),           //
+                          Pair(TSN(18), State::kInFlight),           //
+                          Pair(TSN(19), State::kInFlight)));
+
+  EXPECT_THAT(queue.GetChunksToSend(now_, 1000), ElementsAre(Pair(TSN(10), _)));
+
+  // Ack TSN [17 to 18]
+  for (int tsn = 17; tsn <= 18; ++tsn) {
+    queue.HandleSack(
+        now_,
+        SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(2, (tsn - 9))}, {}));
+  }
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),    //
+                          Pair(TSN(10), State::kNacked),  //
+                          Pair(TSN(11), State::kAcked),   //
+                          Pair(TSN(12), State::kAcked),   //
+                          Pair(TSN(13), State::kAcked),   //
+                          Pair(TSN(14), State::kAcked),   //
+                          Pair(TSN(15), State::kAcked),   //
+                          Pair(TSN(16), State::kAcked),   //
+                          Pair(TSN(17), State::kAcked),   //
+                          Pair(TSN(18), State::kAcked),   //
+                          Pair(TSN(19), State::kInFlight)));
+
+  EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
+
+  // Ack TSN 19 - three more nacks for TSN 10, no more retransmissions.
+  EXPECT_CALL(producer_, Discard(IsUnordered(false), StreamID(1), MID(42)))
+      .WillOnce(Return(false));
+  queue.HandleSack(
+      now_, SackChunk(TSN(9), kArwnd, {SackChunk::GapAckBlock(2, 10)}, {}));
+
+  EXPECT_THAT(queue.GetChunksToSend(now_, 1000), IsEmpty());
+
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),       //
+                          Pair(TSN(10), State::kAbandoned),  //
+                          Pair(TSN(11), State::kAcked),      //
+                          Pair(TSN(12), State::kAcked),      //
+                          Pair(TSN(13), State::kAcked),      //
+                          Pair(TSN(14), State::kAcked),      //
+                          Pair(TSN(15), State::kAcked),      //
+                          Pair(TSN(16), State::kAcked),      //
+                          Pair(TSN(17), State::kAcked),      //
+                          Pair(TSN(18), State::kAcked),      //
+                          Pair(TSN(19), State::kAcked)));
+
+  EXPECT_TRUE(queue.ShouldSendForwardTsn(now_));
+}
+
+TEST_F(RetransmissionQueueTest, CwndRecoversWhenAcking) {
+  RetransmissionQueue queue = CreateQueue();
+  static constexpr size_t kCwnd = 1200;
+  queue.set_cwnd(kCwnd);
+  EXPECT_EQ(queue.cwnd(), kCwnd);
+
+  std::vector<uint8_t> payload(1000);
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this, payload](TimeMs, size_t) {
+        return SendQueue::DataToSend(gen_.Ordered(payload, "BE"));
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 1500);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _)));
+  size_t serialized_size = payload.size() + DataChunk::kHeaderSize;
+  EXPECT_EQ(queue.outstanding_bytes(), serialized_size);
+
+  queue.HandleSack(now_, SackChunk(TSN(10), kArwnd, {}, {}));
+
+  EXPECT_EQ(queue.cwnd(), kCwnd + serialized_size);
+}
+
+// Verifies that it doesn't produce tiny packets, when getting close to
+// the full congestion window.
+TEST_F(RetransmissionQueueTest, OnlySendsLargePacketsOnLargeCongestionWindow) {
+  RetransmissionQueue queue = CreateQueue();
+  size_t intial_cwnd = options_.avoid_fragmentation_cwnd_mtus * options_.mtu;
+  queue.set_cwnd(intial_cwnd);
+  EXPECT_EQ(queue.cwnd(), intial_cwnd);
+
+  // Fill the congestion window almost - leaving 500 bytes.
+  size_t chunk_size = intial_cwnd - 500;
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([chunk_size, this](TimeMs, size_t) {
+        return SendQueue::DataToSend(
+            gen_.Ordered(std::vector<uint8_t>(chunk_size), "BE"));
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_TRUE(queue.can_send_data());
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 10000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _)));
+
+  // To little space left - will not send more.
+  EXPECT_FALSE(queue.can_send_data());
+
+  // But when the first chunk is acked, it will continue.
+  queue.HandleSack(now_, SackChunk(TSN(10), kArwnd, {}, {}));
+
+  EXPECT_TRUE(queue.can_send_data());
+  EXPECT_EQ(queue.outstanding_bytes(), 0u);
+  EXPECT_EQ(queue.cwnd(), intial_cwnd + kMaxMtu);
+}
+
+TEST_F(RetransmissionQueueTest, AllowsSmallFragmentsOnSmallCongestionWindow) {
+  RetransmissionQueue queue = CreateQueue();
+  size_t intial_cwnd =
+      options_.avoid_fragmentation_cwnd_mtus * options_.mtu - 1;
+  queue.set_cwnd(intial_cwnd);
+  EXPECT_EQ(queue.cwnd(), intial_cwnd);
+
+  // Fill the congestion window almost - leaving 500 bytes.
+  size_t chunk_size = intial_cwnd - 500;
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([chunk_size, this](TimeMs, size_t) {
+        return SendQueue::DataToSend(
+            gen_.Ordered(std::vector<uint8_t>(chunk_size), "BE"));
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_TRUE(queue.can_send_data());
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 10000);
+  EXPECT_THAT(chunks_to_send, ElementsAre(Pair(TSN(10), _)));
+
+  // With congestion window under limit, allow small packets to be created.
+  EXPECT_TRUE(queue.can_send_data());
+}
+
+TEST_F(RetransmissionQueueTest, ReadyForHandoverWhenHasNoOutstandingData) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(1));
+  EXPECT_EQ(
+      queue.GetHandoverReadiness(),
+      HandoverReadinessStatus(
+          HandoverUnreadinessReason::kRetransmissionQueueOutstandingData));
+
+  queue.HandleSack(now_, SackChunk(TSN(10), kArwnd, {}, {}));
+  EXPECT_EQ(queue.GetHandoverReadiness(), HandoverReadinessStatus());
+}
+
+TEST_F(RetransmissionQueueTest, ReadyForHandoverWhenNothingToRetransmit) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(8));
+  EXPECT_EQ(
+      queue.GetHandoverReadiness(),
+      HandoverReadinessStatus(
+          HandoverUnreadinessReason::kRetransmissionQueueOutstandingData));
+
+  // Send more chunks, but leave some chunks unacked to force retransmission
+  // after three NACKs.
+
+  // Send 18
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(1));
+
+  // Ack 12, 14-15, 17-18
+  queue.HandleSack(now_, SackChunk(TSN(12), kArwnd,
+                                   {SackChunk::GapAckBlock(2, 3),
+                                    SackChunk::GapAckBlock(5, 6)},
+                                   {}));
+
+  // Send 19
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(1));
+
+  // Ack 12, 14-15, 17-19
+  queue.HandleSack(now_, SackChunk(TSN(12), kArwnd,
+                                   {SackChunk::GapAckBlock(2, 3),
+                                    SackChunk::GapAckBlock(5, 7)},
+                                   {}));
+
+  // Send 20
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(1));
+
+  // Ack 12, 14-15, 17-20
+  // This will trigger "fast retransmit" mode and only chunks 13 and 16 will be
+  // resent right now. The send queue will not even be queried.
+  queue.HandleSack(now_, SackChunk(TSN(12), kArwnd,
+                                   {SackChunk::GapAckBlock(2, 3),
+                                    SackChunk::GapAckBlock(5, 8)},
+                                   {}));
+  EXPECT_EQ(
+      queue.GetHandoverReadiness(),
+      HandoverReadinessStatus()
+          .Add(HandoverUnreadinessReason::kRetransmissionQueueOutstandingData)
+          .Add(HandoverUnreadinessReason::kRetransmissionQueueFastRecovery)
+          .Add(HandoverUnreadinessReason::kRetransmissionQueueNotEmpty));
+
+  // Send "fast retransmit" mode chunks
+  EXPECT_CALL(producer_, Produce).Times(0);
+  EXPECT_THAT(GetTSNsForFastRetransmit(queue), SizeIs(2));
+  EXPECT_EQ(
+      queue.GetHandoverReadiness(),
+      HandoverReadinessStatus()
+          .Add(HandoverUnreadinessReason::kRetransmissionQueueOutstandingData)
+          .Add(HandoverUnreadinessReason::kRetransmissionQueueFastRecovery));
+
+  // Ack 20 to confirm the retransmission
+  queue.HandleSack(now_, SackChunk(TSN(20), kArwnd, {}, {}));
+  EXPECT_EQ(queue.GetHandoverReadiness(), HandoverReadinessStatus());
+}
+
+TEST_F(RetransmissionQueueTest, HandoverTest) {
+  RetransmissionQueue queue = CreateQueue();
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(2));
+  queue.HandleSack(now_, SackChunk(TSN(11), kArwnd, {}, {}));
+
+  std::unique_ptr<RetransmissionQueue> handedover_queue =
+      CreateQueueByHandover(queue);
+
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillOnce(CreateChunk())
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+  EXPECT_THAT(GetSentPacketTSNs(*handedover_queue),
+              testing::ElementsAre(TSN(12), TSN(13), TSN(14)));
+
+  handedover_queue->HandleSack(now_, SackChunk(TSN(13), kArwnd, {}, {}));
+  EXPECT_THAT(handedover_queue->GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(13), State::kAcked),  //
+                          Pair(TSN(14), State::kInFlight)));
+}
+
+TEST_F(RetransmissionQueueTest, CanAlwaysSendOnePacket) {
+  RetransmissionQueue queue = CreateQueue();
+
+  // A large payload - enough to not fit two DATA in same packet.
+  size_t mtu = RoundDownTo4(options_.mtu);
+  std::vector<uint8_t> payload(mtu - 100);
+
+  EXPECT_CALL(producer_, Produce)
+      .WillOnce([this, payload](TimeMs, size_t) {
+        return SendQueue::DataToSend(gen_.Ordered(payload, "B"));
+      })
+      .WillOnce([this, payload](TimeMs, size_t) {
+        return SendQueue::DataToSend(gen_.Ordered(payload, ""));
+      })
+      .WillOnce([this, payload](TimeMs, size_t) {
+        return SendQueue::DataToSend(gen_.Ordered(payload, ""));
+      })
+      .WillOnce([this, payload](TimeMs, size_t) {
+        return SendQueue::DataToSend(gen_.Ordered(payload, ""));
+      })
+      .WillOnce([this, payload](TimeMs, size_t) {
+        return SendQueue::DataToSend(gen_.Ordered(payload, "E"));
+      })
+      .WillRepeatedly([](TimeMs, size_t) { return absl::nullopt; });
+
+  // Produce all chunks and put them in the retransmission queue.
+  std::vector<std::pair<TSN, Data>> chunks_to_send =
+      queue.GetChunksToSend(now_, 5 * mtu);
+  EXPECT_THAT(chunks_to_send,
+              ElementsAre(Pair(TSN(10), _), Pair(TSN(11), _), Pair(TSN(12), _),
+                          Pair(TSN(13), _), Pair(TSN(14), _)));
+  EXPECT_THAT(queue.GetChunkStatesForTesting(),
+              ElementsAre(Pair(TSN(9), State::kAcked),      //
+                          Pair(TSN(10), State::kInFlight),  //
+                          Pair(TSN(11), State::kInFlight),  //
+                          Pair(TSN(12), State::kInFlight),
+                          Pair(TSN(13), State::kInFlight),
+                          Pair(TSN(14), State::kInFlight)));
+
+  // Ack 12, and report an empty receiver window (the peer obviously has a
+  // tiny receive window).
+  queue.HandleSack(
+      now_, SackChunk(TSN(9), /*rwnd=*/0, {SackChunk::GapAckBlock(3, 3)}, {}));
+
+  // Force TSN 10 to be retransmitted.
+  queue.HandleT3RtxTimerExpiry();
+
+  // Even if the receiver window is empty, it will allow TSN 10 to be sent.
+  EXPECT_THAT(queue.GetChunksToSend(now_, mtu), ElementsAre(Pair(TSN(10), _)));
+
+  // But not more than that, as there now is outstanding data.
+  EXPECT_THAT(queue.GetChunksToSend(now_, mtu), IsEmpty());
+
+  // Don't ack any new data, and still have receiver window zero.
+  queue.HandleSack(
+      now_, SackChunk(TSN(9), /*rwnd=*/0, {SackChunk::GapAckBlock(3, 3)}, {}));
+
+  // There is in-flight data, so new data should not be allowed to be send since
+  // the receiver window is full.
+  EXPECT_THAT(queue.GetChunksToSend(now_, mtu), IsEmpty());
+
+  // Ack that packet (no more in-flight data), but still report an empty
+  // receiver window.
+  queue.HandleSack(
+      now_, SackChunk(TSN(10), /*rwnd=*/0, {SackChunk::GapAckBlock(2, 2)}, {}));
+
+  // Then TSN 11 can be sent, as there is no in-flight data.
+  EXPECT_THAT(queue.GetChunksToSend(now_, mtu), ElementsAre(Pair(TSN(11), _)));
+  EXPECT_THAT(queue.GetChunksToSend(now_, mtu), IsEmpty());
+
+  // Ack and recover the receiver window
+  queue.HandleSack(now_, SackChunk(TSN(12), /*rwnd=*/5 * mtu, {}, {}));
+
+  // That will unblock sending remaining chunks.
+  EXPECT_THAT(queue.GetChunksToSend(now_, mtu), ElementsAre(Pair(TSN(13), _)));
+  EXPECT_THAT(queue.GetChunksToSend(now_, mtu), ElementsAre(Pair(TSN(14), _)));
+  EXPECT_THAT(queue.GetChunksToSend(now_, mtu), IsEmpty());
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/retransmission_timeout.cc b/third_party/libwebrtc/net/dcsctp/tx/retransmission_timeout.cc
new file mode 100644
index 0000000000..7d8fb9761c
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/retransmission_timeout.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/retransmission_timeout.h"
+
+#include <algorithm>
+#include <cstdint>
+
+#include "net/dcsctp/public/dcsctp_options.h"
+
+namespace dcsctp {
+
+RetransmissionTimeout::RetransmissionTimeout(const DcSctpOptions& options)
+    : min_rto_(*options.rto_min),
+      max_rto_(*options.rto_max),
+      max_rtt_(*options.rtt_max),
+      min_rtt_variance_(*options.min_rtt_variance),
+      scaled_srtt_(*options.rto_initial << kRttShift),
+      rto_(*options.rto_initial) {}
+
+void RetransmissionTimeout::ObserveRTT(DurationMs measured_rtt) {
+  const int32_t rtt = *measured_rtt;
+
+  // Unrealistic values will be skipped. If a wrongly measured (or otherwise
+  // corrupt) value was processed, it could change the state in a way that would
+  // take a very long time to recover.
+  if (rtt < 0 || rtt > max_rtt_) {
+    return;
+  }
+
+  // From https://tools.ietf.org/html/rfc4960#section-6.3.1, but avoiding
+  // floating point math by implementing algorithm from "V. Jacobson: Congestion
+  // avoidance and control", but adapted for SCTP.
+  if (first_measurement_) {
+    scaled_srtt_ = rtt << kRttShift;
+    scaled_rtt_var_ = (rtt / 2) << kRttVarShift;
+    first_measurement_ = false;
+  } else {
+    int32_t rtt_diff = rtt - (scaled_srtt_ >> kRttShift);
+    scaled_srtt_ += rtt_diff;
+    if (rtt_diff < 0) {
+      rtt_diff = -rtt_diff;
+    }
+    rtt_diff -= (scaled_rtt_var_ >> kRttVarShift);
+    scaled_rtt_var_ += rtt_diff;
+  }
+
+  if (scaled_rtt_var_ < min_rtt_variance_) {
+    scaled_rtt_var_ = min_rtt_variance_;
+  }
+
+  rto_ = (scaled_srtt_ >> kRttShift) + scaled_rtt_var_;
+
+  // Clamp RTO between min and max.
+  rto_ = std::min(std::max(rto_, min_rto_), max_rto_);
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/retransmission_timeout.h b/third_party/libwebrtc/net/dcsctp/tx/retransmission_timeout.h
new file mode 100644
index 0000000000..01530cb3b5
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/retransmission_timeout.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TX_RETRANSMISSION_TIMEOUT_H_
+#define NET_DCSCTP_TX_RETRANSMISSION_TIMEOUT_H_
+
+#include <cstdint>
+#include <functional>
+
+#include "net/dcsctp/public/dcsctp_options.h"
+
+namespace dcsctp {
+
+// Manages updating of the Retransmission Timeout (RTO) SCTP variable, which is
+// used directly as the base timeout for T3-RTX and for other timers, such as
+// delayed ack.
+//
+// When a round-trip-time (RTT) is calculated (outside this class), `Observe`
+// is called, which calculates the retransmission timeout (RTO) value. The RTO
+// value will become larger if the RTT is high and/or the RTT values are varying
+// a lot, which is an indicator of a bad connection.
+class RetransmissionTimeout {
+ public:
+  static constexpr int kRttShift = 3;
+  static constexpr int kRttVarShift = 2;
+  explicit RetransmissionTimeout(const DcSctpOptions& options);
+
+  // To be called when a RTT has been measured, to update the RTO value.
+  void ObserveRTT(DurationMs measured_rtt);
+
+  // Returns the Retransmission Timeout (RTO) value, in milliseconds.
+  DurationMs rto() const { return DurationMs(rto_); }
+
+  // Returns the smoothed RTT value, in milliseconds.
+  DurationMs srtt() const { return DurationMs(scaled_srtt_ >> kRttShift); }
+
+ private:
+  const int32_t min_rto_;
+  const int32_t max_rto_;
+  const int32_t max_rtt_;
+  const int32_t min_rtt_variance_;
+  // If this is the first measurement
+  bool first_measurement_ = true;
+  // Smoothed Round-Trip Time, shifted by kRttShift
+  int32_t scaled_srtt_;
+  // Round-Trip Time Variation, shifted by kRttVarShift
+  int32_t scaled_rtt_var_ = 0;
+  // Retransmission Timeout
+  int32_t rto_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TX_RETRANSMISSION_TIMEOUT_H_
diff --git a/third_party/libwebrtc/net/dcsctp/tx/retransmission_timeout_test.cc b/third_party/libwebrtc/net/dcsctp/tx/retransmission_timeout_test.cc
new file mode 100644
index 0000000000..b901995e97
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/retransmission_timeout_test.cc
@@ -0,0 +1,180 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/retransmission_timeout.h"
+
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+
+constexpr DurationMs kMaxRtt = DurationMs(8'000);
+constexpr DurationMs kInitialRto = DurationMs(200);
+constexpr DurationMs kMaxRto = DurationMs(800);
+constexpr DurationMs kMinRto = DurationMs(120);
+constexpr DurationMs kMinRttVariance = DurationMs(220);
+
+DcSctpOptions MakeOptions() {
+  DcSctpOptions options;
+  options.rtt_max = kMaxRtt;
+  options.rto_initial = kInitialRto;
+  options.rto_max = kMaxRto;
+  options.rto_min = kMinRto;
+  options.min_rtt_variance = kMinRttVariance;
+  return options;
+}
+
+TEST(RetransmissionTimeoutTest, HasValidInitialRto) {
+  RetransmissionTimeout rto_(MakeOptions());
+  EXPECT_EQ(rto_.rto(), kInitialRto);
+}
+
+TEST(RetransmissionTimeoutTest, HasValidInitialSrtt) {
+  RetransmissionTimeout rto_(MakeOptions());
+  EXPECT_EQ(rto_.srtt(), kInitialRto);
+}
+
+TEST(RetransmissionTimeoutTest, NegativeValuesDoNotAffectRTO) {
+  RetransmissionTimeout rto_(MakeOptions());
+  // Initial negative value
+  rto_.ObserveRTT(DurationMs(-10));
+  EXPECT_EQ(rto_.rto(), kInitialRto);
+  rto_.ObserveRTT(DurationMs(124));
+  EXPECT_EQ(*rto_.rto(), 372);
+  // Subsequent negative value
+  rto_.ObserveRTT(DurationMs(-10));
+  EXPECT_EQ(*rto_.rto(), 372);
+}
+
+TEST(RetransmissionTimeoutTest, TooLargeValuesDoNotAffectRTO) {
+  RetransmissionTimeout rto_(MakeOptions());
+  // Initial too large value
+  rto_.ObserveRTT(kMaxRtt + DurationMs(100));
+  EXPECT_EQ(rto_.rto(), kInitialRto);
+  rto_.ObserveRTT(DurationMs(124));
+  EXPECT_EQ(*rto_.rto(), 372);
+  // Subsequent too large value
+  rto_.ObserveRTT(kMaxRtt + DurationMs(100));
+  EXPECT_EQ(*rto_.rto(), 372);
+}
+
+TEST(RetransmissionTimeoutTest, WillNeverGoBelowMinimumRto) {
+  RetransmissionTimeout rto_(MakeOptions());
+  for (int i = 0; i < 1000; ++i) {
+    rto_.ObserveRTT(DurationMs(1));
+  }
+  EXPECT_GE(rto_.rto(), kMinRto);
+}
+
+TEST(RetransmissionTimeoutTest, WillNeverGoAboveMaximumRto) {
+  RetransmissionTimeout rto_(MakeOptions());
+  for (int i = 0; i < 1000; ++i) {
+    rto_.ObserveRTT(kMaxRtt - DurationMs(1));
+    // Adding jitter, which would make it RTO be well above RTT.
+    rto_.ObserveRTT(kMaxRtt - DurationMs(100));
+  }
+  EXPECT_LE(rto_.rto(), kMaxRto);
+}
+
+TEST(RetransmissionTimeoutTest, CalculatesRtoForStableRtt) {
+  RetransmissionTimeout rto_(MakeOptions());
+  rto_.ObserveRTT(DurationMs(124));
+  EXPECT_EQ(*rto_.rto(), 372);
+  rto_.ObserveRTT(DurationMs(128));
+  EXPECT_EQ(*rto_.rto(), 344);
+  rto_.ObserveRTT(DurationMs(123));
+  EXPECT_EQ(*rto_.rto(), 344);
+  rto_.ObserveRTT(DurationMs(125));
+  EXPECT_EQ(*rto_.rto(), 344);
+  rto_.ObserveRTT(DurationMs(127));
+  EXPECT_EQ(*rto_.rto(), 344);
+}
+
+TEST(RetransmissionTimeoutTest, CalculatesRtoForUnstableRtt) {
+  RetransmissionTimeout rto_(MakeOptions());
+  rto_.ObserveRTT(DurationMs(124));
+  EXPECT_EQ(*rto_.rto(), 372);
+  rto_.ObserveRTT(DurationMs(402));
+  EXPECT_EQ(*rto_.rto(), 622);
+  rto_.ObserveRTT(DurationMs(728));
+  EXPECT_EQ(*rto_.rto(), 800);
+  rto_.ObserveRTT(DurationMs(89));
+  EXPECT_EQ(*rto_.rto(), 800);
+  rto_.ObserveRTT(DurationMs(126));
+  EXPECT_EQ(*rto_.rto(), 800);
+}
+
+TEST(RetransmissionTimeoutTest, WillStabilizeAfterAWhile) {
+  RetransmissionTimeout rto_(MakeOptions());
+  rto_.ObserveRTT(DurationMs(124));
+  rto_.ObserveRTT(DurationMs(402));
+  rto_.ObserveRTT(DurationMs(728));
+  rto_.ObserveRTT(DurationMs(89));
+  rto_.ObserveRTT(DurationMs(126));
+  EXPECT_EQ(*rto_.rto(), 800);
+  rto_.ObserveRTT(DurationMs(124));
+  EXPECT_EQ(*rto_.rto(), 800);
+  rto_.ObserveRTT(DurationMs(122));
+  EXPECT_EQ(*rto_.rto(), 710);
+  rto_.ObserveRTT(DurationMs(123));
+  EXPECT_EQ(*rto_.rto(), 631);
+  rto_.ObserveRTT(DurationMs(124));
+  EXPECT_EQ(*rto_.rto(), 562);
+  rto_.ObserveRTT(DurationMs(122));
+  EXPECT_EQ(*rto_.rto(), 505);
+  rto_.ObserveRTT(DurationMs(124));
+  EXPECT_EQ(*rto_.rto(), 454);
+  rto_.ObserveRTT(DurationMs(124));
+  EXPECT_EQ(*rto_.rto(), 410);
+  rto_.ObserveRTT(DurationMs(124));
+  EXPECT_EQ(*rto_.rto(), 372);
+  rto_.ObserveRTT(DurationMs(124));
+  EXPECT_EQ(*rto_.rto(), 367);
+}
+
+TEST(RetransmissionTimeoutTest, WillAlwaysStayAboveRTT) {
+  // In simulations, it's quite common to have a very stable RTT, and having an
+  // RTO at the same value will cause issues as expiry timers will be scheduled
+  // to be expire exactly when a packet is supposed to arrive. The RTO must be
+  // larger than the RTT. In non-simulated environments, this is a non-issue as
+  // any jitter will increase the RTO.
+  RetransmissionTimeout rto_(MakeOptions());
+
+  for (int i = 0; i < 1000; ++i) {
+    rto_.ObserveRTT(DurationMs(124));
+  }
+  EXPECT_EQ(*rto_.rto(), 344);
+}
+
+TEST(RetransmissionTimeoutTest, CanSpecifySmallerMinimumRttVariance) {
+  DcSctpOptions options = MakeOptions();
+  options.min_rtt_variance = kMinRttVariance - DurationMs(100);
+  RetransmissionTimeout rto_(options);
+
+  for (int i = 0; i < 1000; ++i) {
+    rto_.ObserveRTT(DurationMs(124));
+  }
+  EXPECT_EQ(*rto_.rto(), 244);
+}
+
+TEST(RetransmissionTimeoutTest, CanSpecifyLargerMinimumRttVariance) {
+  DcSctpOptions options = MakeOptions();
+  options.min_rtt_variance = kMinRttVariance + DurationMs(100);
+  RetransmissionTimeout rto_(options);
+
+  for (int i = 0; i < 1000; ++i) {
+    rto_.ObserveRTT(DurationMs(124));
+  }
+  EXPECT_EQ(*rto_.rto(), 444);
+}
+
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/rr_send_queue.cc b/third_party/libwebrtc/net/dcsctp/tx/rr_send_queue.cc
new file mode 100644
index 0000000000..b1812f0f8a
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/rr_send_queue.cc
@@ -0,0 +1,542 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/rr_send_queue.h"
+
+#include <cstdint>
+#include <deque>
+#include <limits>
+#include <map>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/str_join.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/tx/send_queue.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+
+RRSendQueue::RRSendQueue(absl::string_view log_prefix,
+                         DcSctpSocketCallbacks* callbacks,
+                         size_t buffer_size,
+                         size_t mtu,
+                         StreamPriority default_priority,
+                         size_t total_buffered_amount_low_threshold)
+    : log_prefix_(std::string(log_prefix) + "fcfs: "),
+      callbacks_(*callbacks),
+      buffer_size_(buffer_size),
+      default_priority_(default_priority),
+      scheduler_(mtu),
+      total_buffered_amount_(
+          [this]() { callbacks_.OnTotalBufferedAmountLow(); }) {
+  total_buffered_amount_.SetLowThreshold(total_buffered_amount_low_threshold);
+}
+
+size_t RRSendQueue::OutgoingStream::bytes_to_send_in_next_message() const {
+  if (pause_state_ == PauseState::kPaused ||
+      pause_state_ == PauseState::kResetting) {
+    // The stream has paused (and there is no partially sent message).
+    return 0;
+  }
+
+  if (items_.empty()) {
+    return 0;
+  }
+
+  return items_.front().remaining_size;
+}
+
+void RRSendQueue::OutgoingStream::AddHandoverState(
+    DcSctpSocketHandoverState::OutgoingStream& state) const {
+  state.next_ssn = next_ssn_.value();
+  state.next_ordered_mid = next_ordered_mid_.value();
+  state.next_unordered_mid = next_unordered_mid_.value();
+  state.priority = *scheduler_stream_->priority();
+}
+
+bool RRSendQueue::IsConsistent() const {
+  std::set<StreamID> expected_active_streams;
+  std::set<StreamID> actual_active_streams =
+      scheduler_.ActiveStreamsForTesting();
+
+  size_t total_buffered_amount = 0;
+  for (const auto& [stream_id, stream] : streams_) {
+    total_buffered_amount += stream.buffered_amount().value();
+    if (stream.bytes_to_send_in_next_message() > 0) {
+      expected_active_streams.emplace(stream_id);
+    }
+  }
+  if (expected_active_streams != actual_active_streams) {
+    auto fn = [&](rtc::StringBuilder& sb, const auto& p) { sb << *p; };
+    RTC_DLOG(LS_ERROR) << "Active streams mismatch, is=["
+                       << StrJoin(actual_active_streams, ",", fn)
+                       << "], expected=["
+                       << StrJoin(expected_active_streams, ",", fn) << "]";
+    return false;
+  }
+
+  return total_buffered_amount == total_buffered_amount_.value();
+}
+
+bool RRSendQueue::OutgoingStream::IsConsistent() const {
+  size_t bytes = 0;
+  for (const auto& item : items_) {
+    bytes += item.remaining_size;
+  }
+  return bytes == buffered_amount_.value();
+}
+
+void RRSendQueue::ThresholdWatcher::Decrease(size_t bytes) {
+  RTC_DCHECK(bytes <= value_);
+  size_t old_value = value_;
+  value_ -= bytes;
+
+  if (old_value > low_threshold_ && value_ <= low_threshold_) {
+    on_threshold_reached_();
+  }
+}
+
+void RRSendQueue::ThresholdWatcher::SetLowThreshold(size_t low_threshold) {
+  // Betting on https://github.com/w3c/webrtc-pc/issues/2654 being accepted.
+  if (low_threshold_ < value_ && low_threshold >= value_) {
+    on_threshold_reached_();
+  }
+  low_threshold_ = low_threshold;
+}
+
+void RRSendQueue::OutgoingStream::Add(DcSctpMessage message,
+                                      MessageAttributes attributes) {
+  bool was_active = bytes_to_send_in_next_message() > 0;
+  buffered_amount_.Increase(message.payload().size());
+  parent_.total_buffered_amount_.Increase(message.payload().size());
+  items_.emplace_back(std::move(message), std::move(attributes));
+
+  if (!was_active) {
+    scheduler_stream_->MaybeMakeActive();
+  }
+
+  RTC_DCHECK(IsConsistent());
+}
+
+absl::optional<SendQueue::DataToSend> RRSendQueue::OutgoingStream::Produce(
+    TimeMs now,
+    size_t max_size) {
+  RTC_DCHECK(pause_state_ != PauseState::kPaused &&
+             pause_state_ != PauseState::kResetting);
+
+  while (!items_.empty()) {
+    Item& item = items_.front();
+    DcSctpMessage& message = item.message;
+
+    // Allocate Message ID and SSN when the first fragment is sent.
+    if (!item.message_id.has_value()) {
+      // Oops, this entire message has already expired. Try the next one.
+      if (item.attributes.expires_at <= now) {
+        HandleMessageExpired(item);
+        items_.pop_front();
+        continue;
+      }
+
+      MID& mid =
+          item.attributes.unordered ? next_unordered_mid_ : next_ordered_mid_;
+      item.message_id = mid;
+      mid = MID(*mid + 1);
+    }
+    if (!item.attributes.unordered && !item.ssn.has_value()) {
+      item.ssn = next_ssn_;
+      next_ssn_ = SSN(*next_ssn_ + 1);
+    }
+
+    // Grab the next `max_size` fragment from this message and calculate flags.
+    rtc::ArrayView<const uint8_t> chunk_payload =
+        item.message.payload().subview(item.remaining_offset, max_size);
+    rtc::ArrayView<const uint8_t> message_payload = message.payload();
+    Data::IsBeginning is_beginning(chunk_payload.data() ==
+                                   message_payload.data());
+    Data::IsEnd is_end((chunk_payload.data() + chunk_payload.size()) ==
+                       (message_payload.data() + message_payload.size()));
+
+    StreamID stream_id = message.stream_id();
+    PPID ppid = message.ppid();
+
+    // Zero-copy the payload if the message fits in a single chunk.
+    std::vector<uint8_t> payload =
+        is_beginning && is_end
+            ? std::move(message).ReleasePayload()
+            : std::vector<uint8_t>(chunk_payload.begin(), chunk_payload.end());
+
+    FSN fsn(item.current_fsn);
+    item.current_fsn = FSN(*item.current_fsn + 1);
+    buffered_amount_.Decrease(payload.size());
+    parent_.total_buffered_amount_.Decrease(payload.size());
+
+    SendQueue::DataToSend chunk(Data(stream_id, item.ssn.value_or(SSN(0)),
+                                     item.message_id.value(), fsn, ppid,
+                                     std::move(payload), is_beginning, is_end,
+                                     item.attributes.unordered));
+    chunk.max_retransmissions = item.attributes.max_retransmissions;
+    chunk.expires_at = item.attributes.expires_at;
+    chunk.lifecycle_id =
+        is_end ? item.attributes.lifecycle_id : LifecycleId::NotSet();
+
+    if (is_end) {
+      // The entire message has been sent, and its last data copied to `chunk`,
+      // so it can safely be discarded.
+      items_.pop_front();
+
+      if (pause_state_ == PauseState::kPending) {
+        RTC_DLOG(LS_VERBOSE) << "Pause state on " << *stream_id
+                             << " is moving from pending to paused";
+        pause_state_ = PauseState::kPaused;
+      }
+    } else {
+      item.remaining_offset += chunk_payload.size();
+      item.remaining_size -= chunk_payload.size();
+      RTC_DCHECK(item.remaining_offset + item.remaining_size ==
+                 item.message.payload().size());
+      RTC_DCHECK(item.remaining_size > 0);
+    }
+    RTC_DCHECK(IsConsistent());
+    return chunk;
+  }
+  RTC_DCHECK(IsConsistent());
+  return absl::nullopt;
+}
+
+void RRSendQueue::OutgoingStream::HandleMessageExpired(
+    OutgoingStream::Item& item) {
+  buffered_amount_.Decrease(item.remaining_size);
+  parent_.total_buffered_amount_.Decrease(item.remaining_size);
+  if (item.attributes.lifecycle_id.IsSet()) {
+    RTC_DLOG(LS_VERBOSE) << "Triggering OnLifecycleMessageExpired("
+                         << item.attributes.lifecycle_id.value() << ", false)";
+
+    parent_.callbacks_.OnLifecycleMessageExpired(item.attributes.lifecycle_id,
+                                                 /*maybe_delivered=*/false);
+    parent_.callbacks_.OnLifecycleEnd(item.attributes.lifecycle_id);
+  }
+}
+
+bool RRSendQueue::OutgoingStream::Discard(IsUnordered unordered,
+                                          MID message_id) {
+  bool result = false;
+  if (!items_.empty()) {
+    Item& item = items_.front();
+    if (item.attributes.unordered == unordered && item.message_id.has_value() &&
+        *item.message_id == message_id) {
+      HandleMessageExpired(item);
+      items_.pop_front();
+
+      // Only partially sent messages are discarded, so if a message was
+      // discarded, then it was the currently sent message.
+      scheduler_stream_->ForceReschedule();
+
+      if (pause_state_ == PauseState::kPending) {
+        pause_state_ = PauseState::kPaused;
+        scheduler_stream_->MakeInactive();
+      } else if (bytes_to_send_in_next_message() == 0) {
+        scheduler_stream_->MakeInactive();
+      }
+
+      // As the item still existed, it had unsent data.
+      result = true;
+    }
+  }
+  RTC_DCHECK(IsConsistent());
+  return result;
+}
+
+void RRSendQueue::OutgoingStream::Pause() {
+  if (pause_state_ != PauseState::kNotPaused) {
+    // Already in progress.
+    return;
+  }
+
+  bool had_pending_items = !items_.empty();
+
+  // https://datatracker.ietf.org/doc/html/rfc8831#section-6.7
+  // "Closing of a data channel MUST be signaled by resetting the corresponding
+  // outgoing streams [RFC6525].  This means that if one side decides to close
+  // the data channel, it resets the corresponding outgoing stream."
+  // ... "[RFC6525] also guarantees that all the messages are delivered (or
+  // abandoned) before the stream is reset."
+
+  // A stream is paused when it's about to be reset. In this implementation,
+  // it will throw away all non-partially send messages - they will be abandoned
+  // as noted above. This is subject to change. It will however not discard any
+  // partially sent messages - only whole messages. Partially delivered messages
+  // (at the time of receiving a Stream Reset command) will always deliver all
+  // the fragments before actually resetting the stream.
+  for (auto it = items_.begin(); it != items_.end();) {
+    if (it->remaining_offset == 0) {
+      HandleMessageExpired(*it);
+      it = items_.erase(it);
+    } else {
+      ++it;
+    }
+  }
+
+  pause_state_ = (items_.empty() || items_.front().remaining_offset == 0)
+                     ? PauseState::kPaused
+                     : PauseState::kPending;
+
+  if (had_pending_items && pause_state_ == PauseState::kPaused) {
+    RTC_DLOG(LS_VERBOSE) << "Stream " << *stream_id()
+                         << " was previously active, but is now paused.";
+    scheduler_stream_->MakeInactive();
+  }
+
+  RTC_DCHECK(IsConsistent());
+}
+
+void RRSendQueue::OutgoingStream::Resume() {
+  RTC_DCHECK(pause_state_ == PauseState::kResetting);
+  pause_state_ = PauseState::kNotPaused;
+  scheduler_stream_->MaybeMakeActive();
+  RTC_DCHECK(IsConsistent());
+}
+
+void RRSendQueue::OutgoingStream::Reset() {
+  // This can be called both when an outgoing stream reset has been responded
+  // to, or when the entire SendQueue is reset due to detecting the peer having
+  // restarted. The stream may be in any state at this time.
+  PauseState old_pause_state = pause_state_;
+  pause_state_ = PauseState::kNotPaused;
+  next_ordered_mid_ = MID(0);
+  next_unordered_mid_ = MID(0);
+  next_ssn_ = SSN(0);
+  if (!items_.empty()) {
+    // If this message has been partially sent, reset it so that it will be
+    // re-sent.
+    auto& item = items_.front();
+    buffered_amount_.Increase(item.message.payload().size() -
+                              item.remaining_size);
+    parent_.total_buffered_amount_.Increase(item.message.payload().size() -
+                                            item.remaining_size);
+    item.remaining_offset = 0;
+    item.remaining_size = item.message.payload().size();
+    item.message_id = absl::nullopt;
+    item.ssn = absl::nullopt;
+    item.current_fsn = FSN(0);
+    if (old_pause_state == PauseState::kPaused ||
+        old_pause_state == PauseState::kResetting) {
+      scheduler_stream_->MaybeMakeActive();
+    }
+  }
+  RTC_DCHECK(IsConsistent());
+}
+
+bool RRSendQueue::OutgoingStream::has_partially_sent_message() const {
+  if (items_.empty()) {
+    return false;
+  }
+  return items_.front().message_id.has_value();
+}
+
+void RRSendQueue::Add(TimeMs now,
+                      DcSctpMessage message,
+                      const SendOptions& send_options) {
+  RTC_DCHECK(!message.payload().empty());
+  // Any limited lifetime should start counting from now - when the message
+  // has been added to the queue.
+
+  // `expires_at` is the time when it expires. Which is slightly larger than the
+  // message's lifetime, as the message is alive during its entire lifetime
+  // (which may be zero).
+  MessageAttributes attributes = {
+      .unordered = send_options.unordered,
+      .max_retransmissions =
+          send_options.max_retransmissions.has_value()
+              ? MaxRetransmits(send_options.max_retransmissions.value())
+              : MaxRetransmits::NoLimit(),
+      .expires_at = send_options.lifetime.has_value()
+                        ? now + *send_options.lifetime + DurationMs(1)
+                        : TimeMs::InfiniteFuture(),
+      .lifecycle_id = send_options.lifecycle_id,
+  };
+  GetOrCreateStreamInfo(message.stream_id())
+      .Add(std::move(message), std::move(attributes));
+  RTC_DCHECK(IsConsistent());
+}
+
+bool RRSendQueue::IsFull() const {
+  return total_buffered_amount() >= buffer_size_;
+}
+
+bool RRSendQueue::IsEmpty() const {
+  return total_buffered_amount() == 0;
+}
+
+absl::optional<SendQueue::DataToSend> RRSendQueue::Produce(TimeMs now,
+                                                           size_t max_size) {
+  return scheduler_.Produce(now, max_size);
+}
+
+bool RRSendQueue::Discard(IsUnordered unordered,
+                          StreamID stream_id,
+                          MID message_id) {
+  bool has_discarded =
+      GetOrCreateStreamInfo(stream_id).Discard(unordered, message_id);
+
+  RTC_DCHECK(IsConsistent());
+  return has_discarded;
+}
+
+void RRSendQueue::PrepareResetStream(StreamID stream_id) {
+  GetOrCreateStreamInfo(stream_id).Pause();
+  RTC_DCHECK(IsConsistent());
+}
+
+bool RRSendQueue::HasStreamsReadyToBeReset() const {
+  for (auto& [unused, stream] : streams_) {
+    if (stream.IsReadyToBeReset()) {
+      return true;
+    }
+  }
+  return false;
+}
+std::vector<StreamID> RRSendQueue::GetStreamsReadyToBeReset() {
+  RTC_DCHECK(absl::c_count_if(streams_, [](const auto& p) {
+               return p.second.IsResetting();
+             }) == 0);
+  std::vector<StreamID> ready;
+  for (auto& [stream_id, stream] : streams_) {
+    if (stream.IsReadyToBeReset()) {
+      stream.SetAsResetting();
+      ready.push_back(stream_id);
+    }
+  }
+  return ready;
+}
+
+void RRSendQueue::CommitResetStreams() {
+  RTC_DCHECK(absl::c_count_if(streams_, [](const auto& p) {
+               return p.second.IsResetting();
+             }) > 0);
+  for (auto& [unused, stream] : streams_) {
+    if (stream.IsResetting()) {
+      stream.Reset();
+    }
+  }
+  RTC_DCHECK(IsConsistent());
+}
+
+void RRSendQueue::RollbackResetStreams() {
+  RTC_DCHECK(absl::c_count_if(streams_, [](const auto& p) {
+               return p.second.IsResetting();
+             }) > 0);
+  for (auto& [unused, stream] : streams_) {
+    if (stream.IsResetting()) {
+      stream.Resume();
+    }
+  }
+  RTC_DCHECK(IsConsistent());
+}
+
+void RRSendQueue::Reset() {
+  // Recalculate buffered amount, as partially sent messages may have been put
+  // fully back in the queue.
+  for (auto& [unused, stream] : streams_) {
+    stream.Reset();
+  }
+  scheduler_.ForceReschedule();
+}
+
+size_t RRSendQueue::buffered_amount(StreamID stream_id) const {
+  auto it = streams_.find(stream_id);
+  if (it == streams_.end()) {
+    return 0;
+  }
+  return it->second.buffered_amount().value();
+}
+
+size_t RRSendQueue::buffered_amount_low_threshold(StreamID stream_id) const {
+  auto it = streams_.find(stream_id);
+  if (it == streams_.end()) {
+    return 0;
+  }
+  return it->second.buffered_amount().low_threshold();
+}
+
+void RRSendQueue::SetBufferedAmountLowThreshold(StreamID stream_id,
+                                                size_t bytes) {
+  GetOrCreateStreamInfo(stream_id).buffered_amount().SetLowThreshold(bytes);
+}
+
+RRSendQueue::OutgoingStream& RRSendQueue::GetOrCreateStreamInfo(
+    StreamID stream_id) {
+  auto it = streams_.find(stream_id);
+  if (it != streams_.end()) {
+    return it->second;
+  }
+
+  return streams_
+      .emplace(
+          std::piecewise_construct, std::forward_as_tuple(stream_id),
+          std::forward_as_tuple(this, &scheduler_, stream_id, default_priority_,
+                                [this, stream_id]() {
+                                  callbacks_.OnBufferedAmountLow(stream_id);
+                                }))
+      .first->second;
+}
+
+void RRSendQueue::SetStreamPriority(StreamID stream_id,
+                                    StreamPriority priority) {
+  OutgoingStream& stream = GetOrCreateStreamInfo(stream_id);
+
+  stream.SetPriority(priority);
+  RTC_DCHECK(IsConsistent());
+}
+
+StreamPriority RRSendQueue::GetStreamPriority(StreamID stream_id) const {
+  auto stream_it = streams_.find(stream_id);
+  if (stream_it == streams_.end()) {
+    return default_priority_;
+  }
+  return stream_it->second.priority();
+}
+
+HandoverReadinessStatus RRSendQueue::GetHandoverReadiness() const {
+  HandoverReadinessStatus status;
+  if (!IsEmpty()) {
+    status.Add(HandoverUnreadinessReason::kSendQueueNotEmpty);
+  }
+  return status;
+}
+
+void RRSendQueue::AddHandoverState(DcSctpSocketHandoverState& state) {
+  for (const auto& [stream_id, stream] : streams_) {
+    DcSctpSocketHandoverState::OutgoingStream state_stream;
+    state_stream.id = stream_id.value();
+    stream.AddHandoverState(state_stream);
+    state.tx.streams.push_back(std::move(state_stream));
+  }
+}
+
+void RRSendQueue::RestoreFromState(const DcSctpSocketHandoverState& state) {
+  for (const DcSctpSocketHandoverState::OutgoingStream& state_stream :
+       state.tx.streams) {
+    StreamID stream_id(state_stream.id);
+    streams_.emplace(
+        std::piecewise_construct, std::forward_as_tuple(stream_id),
+        std::forward_as_tuple(
+            this, &scheduler_, stream_id, StreamPriority(state_stream.priority),
+            [this, stream_id]() { callbacks_.OnBufferedAmountLow(stream_id); },
+            &state_stream));
+  }
+}
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/rr_send_queue.h b/third_party/libwebrtc/net/dcsctp/tx/rr_send_queue.h
new file mode 100644
index 0000000000..e9b8cd2081
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/rr_send_queue.h
@@ -0,0 +1,282 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TX_RR_SEND_QUEUE_H_
+#define NET_DCSCTP_TX_RR_SEND_QUEUE_H_
+
+#include <cstdint>
+#include <deque>
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/tx/send_queue.h"
+#include "net/dcsctp/tx/stream_scheduler.h"
+
+namespace dcsctp {
+
+// The Round Robin SendQueue holds all messages that the client wants to send,
+// but that haven't yet been split into chunks and fully sent on the wire.
+//
+// As defined in https://datatracker.ietf.org/doc/html/rfc8260#section-3.2,
+// it will cycle to send messages from different streams. It will send all
+// fragments from one message before continuing with a different message on
+// possibly a different stream, until support for message interleaving has been
+// implemented.
+//
+// As messages can be (requested to be) sent before the connection is properly
+// established, this send queue is always present - even for closed connections.
+//
+// The send queue may trigger callbacks:
+//  * `OnBufferedAmountLow`, `OnTotalBufferedAmountLow`
+//    These will be triggered as defined in their documentation.
+//  * `OnLifecycleMessageExpired(/*maybe_delivered=*/false)`, `OnLifecycleEnd`
+//    These will be triggered when messages have been expired, abandoned or
+//    discarded from the send queue. If a message is fully produced, meaning
+//    that the last fragment has been produced, the responsibility to send
+//    lifecycle events is then transferred to the retransmission queue, which
+//    is the one asking to produce the message.
+class RRSendQueue : public SendQueue {
+ public:
+  RRSendQueue(absl::string_view log_prefix,
+              DcSctpSocketCallbacks* callbacks,
+              size_t buffer_size,
+              size_t mtu,
+              StreamPriority default_priority,
+              size_t total_buffered_amount_low_threshold);
+
+  // Indicates if the buffer is full. Note that it's up to the caller to ensure
+  // that the buffer is not full prior to adding new items to it.
+  bool IsFull() const;
+  // Indicates if the buffer is empty.
+  bool IsEmpty() const;
+
+  // Adds the message to be sent using the `send_options` provided. The current
+  // time should be in `now`. Note that it's the responsibility of the caller to
+  // ensure that the buffer is not full (by calling `IsFull`) before adding
+  // messages to it.
+  void Add(TimeMs now,
+           DcSctpMessage message,
+           const SendOptions& send_options = {});
+
+  // Implementation of `SendQueue`.
+  absl::optional<DataToSend> Produce(TimeMs now, size_t max_size) override;
+  bool Discard(IsUnordered unordered,
+               StreamID stream_id,
+               MID message_id) override;
+  void PrepareResetStream(StreamID streams) override;
+  bool HasStreamsReadyToBeReset() const override;
+  std::vector<StreamID> GetStreamsReadyToBeReset() override;
+  void CommitResetStreams() override;
+  void RollbackResetStreams() override;
+  void Reset() override;
+  size_t buffered_amount(StreamID stream_id) const override;
+  size_t total_buffered_amount() const override {
+    return total_buffered_amount_.value();
+  }
+  size_t buffered_amount_low_threshold(StreamID stream_id) const override;
+  void SetBufferedAmountLowThreshold(StreamID stream_id, size_t bytes) override;
+  void EnableMessageInterleaving(bool enabled) override {
+    scheduler_.EnableMessageInterleaving(enabled);
+  }
+
+  void SetStreamPriority(StreamID stream_id, StreamPriority priority);
+  StreamPriority GetStreamPriority(StreamID stream_id) const;
+  HandoverReadinessStatus GetHandoverReadiness() const;
+  void AddHandoverState(DcSctpSocketHandoverState& state);
+  void RestoreFromState(const DcSctpSocketHandoverState& state);
+
+ private:
+  struct MessageAttributes {
+    IsUnordered unordered;
+    MaxRetransmits max_retransmissions;
+    TimeMs expires_at;
+    LifecycleId lifecycle_id;
+  };
+
+  // Represents a value and a "low threshold" that when the value reaches or
+  // goes under the "low threshold", will trigger `on_threshold_reached`
+  // callback.
+  class ThresholdWatcher {
+   public:
+    explicit ThresholdWatcher(std::function<void()> on_threshold_reached)
+        : on_threshold_reached_(std::move(on_threshold_reached)) {}
+    // Increases the value.
+    void Increase(size_t bytes) { value_ += bytes; }
+    // Decreases the value and triggers `on_threshold_reached` if it's at or
+    // below `low_threshold()`.
+    void Decrease(size_t bytes);
+
+    size_t value() const { return value_; }
+    size_t low_threshold() const { return low_threshold_; }
+    void SetLowThreshold(size_t low_threshold);
+
+   private:
+    const std::function<void()> on_threshold_reached_;
+    size_t value_ = 0;
+    size_t low_threshold_ = 0;
+  };
+
+  // Per-stream information.
+  class OutgoingStream : public StreamScheduler::StreamProducer {
+   public:
+    OutgoingStream(
+        RRSendQueue* parent,
+        StreamScheduler* scheduler,
+        StreamID stream_id,
+        StreamPriority priority,
+        std::function<void()> on_buffered_amount_low,
+        const DcSctpSocketHandoverState::OutgoingStream* state = nullptr)
+        : parent_(*parent),
+          scheduler_stream_(scheduler->CreateStream(this, stream_id, priority)),
+          next_unordered_mid_(MID(state ? state->next_unordered_mid : 0)),
+          next_ordered_mid_(MID(state ? state->next_ordered_mid : 0)),
+          next_ssn_(SSN(state ? state->next_ssn : 0)),
+          buffered_amount_(std::move(on_buffered_amount_low)) {}
+
+    StreamID stream_id() const { return scheduler_stream_->stream_id(); }
+
+    // Enqueues a message to this stream.
+    void Add(DcSctpMessage message, MessageAttributes attributes);
+
+    // Implementing `StreamScheduler::StreamProducer`.
+    absl::optional<SendQueue::DataToSend> Produce(TimeMs now,
+                                                  size_t max_size) override;
+    size_t bytes_to_send_in_next_message() const override;
+
+    const ThresholdWatcher& buffered_amount() const { return buffered_amount_; }
+    ThresholdWatcher& buffered_amount() { return buffered_amount_; }
+
+    // Discards a partially sent message, see `SendQueue::Discard`.
+    bool Discard(IsUnordered unordered, MID message_id);
+
+    // Pauses this stream, which is used before resetting it.
+    void Pause();
+
+    // Resumes a paused stream.
+    void Resume();
+
+    bool IsReadyToBeReset() const {
+      return pause_state_ == PauseState::kPaused;
+    }
+
+    bool IsResetting() const { return pause_state_ == PauseState::kResetting; }
+
+    void SetAsResetting() {
+      RTC_DCHECK(pause_state_ == PauseState::kPaused);
+      pause_state_ = PauseState::kResetting;
+    }
+
+    // Resets this stream, meaning MIDs and SSNs are set to zero.
+    void Reset();
+
+    // Indicates if this stream has a partially sent message in it.
+    bool has_partially_sent_message() const;
+
+    StreamPriority priority() const { return scheduler_stream_->priority(); }
+    void SetPriority(StreamPriority priority) {
+      scheduler_stream_->SetPriority(priority);
+    }
+
+    void AddHandoverState(
+        DcSctpSocketHandoverState::OutgoingStream& state) const;
+
+   private:
+    // Streams are paused before they can be reset. To reset a stream, the
+    // socket sends an outgoing stream reset command with the TSN of the last
+    // fragment of the last message, so that receivers and senders can agree on
+    // when it stopped. And if the send queue is in the middle of sending a
+    // message, and without fragments not yet sent and without TSNs allocated to
+    // them, it will keep sending data until that message has ended.
+    enum class PauseState {
+      // The stream is not paused, and not scheduled to be reset.
+      kNotPaused,
+      // The stream has requested to be reset/paused but is still producing
+      // fragments of a message that hasn't ended yet. When it does, it will
+      // transition to the `kPaused` state.
+      kPending,
+      // The stream is fully paused and can be reset.
+      kPaused,
+      // The stream has been added to an outgoing stream reset request and a
+      // response from the peer hasn't been received yet.
+      kResetting,
+    };
+
+    // An enqueued message and metadata.
+    struct Item {
+      explicit Item(DcSctpMessage msg, MessageAttributes attributes)
+          : message(std::move(msg)),
+            attributes(std::move(attributes)),
+            remaining_offset(0),
+            remaining_size(message.payload().size()) {}
+      DcSctpMessage message;
+      MessageAttributes attributes;
+      // The remaining payload (offset and size) to be sent, when it has been
+      // fragmented.
+      size_t remaining_offset;
+      size_t remaining_size;
+      // If set, an allocated Message ID and SSN. Will be allocated when the
+      // first fragment is sent.
+      absl::optional<MID> message_id = absl::nullopt;
+      absl::optional<SSN> ssn = absl::nullopt;
+      // The current Fragment Sequence Number, incremented for each fragment.
+      FSN current_fsn = FSN(0);
+    };
+
+    bool IsConsistent() const;
+    void HandleMessageExpired(OutgoingStream::Item& item);
+
+    RRSendQueue& parent_;
+
+    const std::unique_ptr<StreamScheduler::Stream> scheduler_stream_;
+
+    PauseState pause_state_ = PauseState::kNotPaused;
+    // MIDs are different for unordered and ordered messages sent on a stream.
+    MID next_unordered_mid_;
+    MID next_ordered_mid_;
+
+    SSN next_ssn_;
+    // Enqueued messages, and metadata.
+    std::deque<Item> items_;
+
+    // The current amount of buffered data.
+    ThresholdWatcher buffered_amount_;
+  };
+
+  bool IsConsistent() const;
+  OutgoingStream& GetOrCreateStreamInfo(StreamID stream_id);
+  absl::optional<DataToSend> Produce(
+      std::map<StreamID, OutgoingStream>::iterator it,
+      TimeMs now,
+      size_t max_size);
+
+  const std::string log_prefix_;
+  DcSctpSocketCallbacks& callbacks_;
+  const size_t buffer_size_;
+  const StreamPriority default_priority_;
+  StreamScheduler scheduler_;
+
+  // The total amount of buffer data, for all streams.
+  ThresholdWatcher total_buffered_amount_;
+
+  // All streams, and messages added to those.
+  std::map<StreamID, OutgoingStream> streams_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TX_RR_SEND_QUEUE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/tx/rr_send_queue_test.cc b/third_party/libwebrtc/net/dcsctp/tx/rr_send_queue_test.cc
new file mode 100644
index 0000000000..95416b193a
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/rr_send_queue_test.cc
@@ -0,0 +1,866 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/rr_send_queue.h"
+
+#include <cstdint>
+#include <type_traits>
+#include <vector>
+
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/socket/mock_dcsctp_socket_callbacks.h"
+#include "net/dcsctp/testing/testing_macros.h"
+#include "net/dcsctp/tx/send_queue.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::SizeIs;
+using ::testing::UnorderedElementsAre;
+
+constexpr TimeMs kNow = TimeMs(0);
+constexpr StreamID kStreamID(1);
+constexpr PPID kPPID(53);
+constexpr size_t kMaxQueueSize = 1000;
+constexpr StreamPriority kDefaultPriority(10);
+constexpr size_t kBufferedAmountLowThreshold = 500;
+constexpr size_t kOneFragmentPacketSize = 100;
+constexpr size_t kTwoFragmentPacketSize = 101;
+constexpr size_t kMtu = 1100;
+
+class RRSendQueueTest : public testing::Test {
+ protected:
+  RRSendQueueTest()
+      : buf_("log: ",
+             &callbacks_,
+             kMaxQueueSize,
+             kMtu,
+             kDefaultPriority,
+             kBufferedAmountLowThreshold) {}
+
+  testing::NiceMock<MockDcSctpSocketCallbacks> callbacks_;
+  const DcSctpOptions options_;
+  RRSendQueue buf_;
+};
+
+TEST_F(RRSendQueueTest, EmptyBuffer) {
+  EXPECT_TRUE(buf_.IsEmpty());
+  EXPECT_FALSE(buf_.Produce(kNow, kOneFragmentPacketSize).has_value());
+  EXPECT_FALSE(buf_.IsFull());
+}
+
+TEST_F(RRSendQueueTest, AddAndGetSingleChunk) {
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, {1, 2, 4, 5, 6}));
+
+  EXPECT_FALSE(buf_.IsEmpty());
+  EXPECT_FALSE(buf_.IsFull());
+  absl::optional<SendQueue::DataToSend> chunk_opt =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_opt.has_value());
+  EXPECT_TRUE(chunk_opt->data.is_beginning);
+  EXPECT_TRUE(chunk_opt->data.is_end);
+}
+
+TEST_F(RRSendQueueTest, CarveOutBeginningMiddleAndEnd) {
+  std::vector<uint8_t> payload(60);
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+
+  absl::optional<SendQueue::DataToSend> chunk_beg =
+      buf_.Produce(kNow, /*max_size=*/20);
+  ASSERT_TRUE(chunk_beg.has_value());
+  EXPECT_TRUE(chunk_beg->data.is_beginning);
+  EXPECT_FALSE(chunk_beg->data.is_end);
+
+  absl::optional<SendQueue::DataToSend> chunk_mid =
+      buf_.Produce(kNow, /*max_size=*/20);
+  ASSERT_TRUE(chunk_mid.has_value());
+  EXPECT_FALSE(chunk_mid->data.is_beginning);
+  EXPECT_FALSE(chunk_mid->data.is_end);
+
+  absl::optional<SendQueue::DataToSend> chunk_end =
+      buf_.Produce(kNow, /*max_size=*/20);
+  ASSERT_TRUE(chunk_end.has_value());
+  EXPECT_FALSE(chunk_end->data.is_beginning);
+  EXPECT_TRUE(chunk_end->data.is_end);
+
+  EXPECT_FALSE(buf_.Produce(kNow, kOneFragmentPacketSize).has_value());
+}
+
+TEST_F(RRSendQueueTest, GetChunksFromTwoMessages) {
+  std::vector<uint8_t> payload(60);
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  buf_.Add(kNow, DcSctpMessage(StreamID(3), PPID(54), payload));
+
+  absl::optional<SendQueue::DataToSend> chunk_one =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
+  EXPECT_EQ(chunk_one->data.ppid, kPPID);
+  EXPECT_TRUE(chunk_one->data.is_beginning);
+  EXPECT_TRUE(chunk_one->data.is_end);
+
+  absl::optional<SendQueue::DataToSend> chunk_two =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_two.has_value());
+  EXPECT_EQ(chunk_two->data.stream_id, StreamID(3));
+  EXPECT_EQ(chunk_two->data.ppid, PPID(54));
+  EXPECT_TRUE(chunk_two->data.is_beginning);
+  EXPECT_TRUE(chunk_two->data.is_end);
+}
+
+TEST_F(RRSendQueueTest, BufferBecomesFullAndEmptied) {
+  std::vector<uint8_t> payload(600);
+  EXPECT_FALSE(buf_.IsFull());
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  EXPECT_FALSE(buf_.IsFull());
+  buf_.Add(kNow, DcSctpMessage(StreamID(3), PPID(54), payload));
+  EXPECT_TRUE(buf_.IsFull());
+  // However, it's still possible to add messages. It's a soft limit, and it
+  // might be necessary to forcefully add messages due to e.g. external
+  // fragmentation.
+  buf_.Add(kNow, DcSctpMessage(StreamID(5), PPID(55), payload));
+  EXPECT_TRUE(buf_.IsFull());
+
+  absl::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 1000);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
+  EXPECT_EQ(chunk_one->data.ppid, kPPID);
+
+  EXPECT_TRUE(buf_.IsFull());
+
+  absl::optional<SendQueue::DataToSend> chunk_two = buf_.Produce(kNow, 1000);
+  ASSERT_TRUE(chunk_two.has_value());
+  EXPECT_EQ(chunk_two->data.stream_id, StreamID(3));
+  EXPECT_EQ(chunk_two->data.ppid, PPID(54));
+
+  EXPECT_FALSE(buf_.IsFull());
+  EXPECT_FALSE(buf_.IsEmpty());
+
+  absl::optional<SendQueue::DataToSend> chunk_three = buf_.Produce(kNow, 1000);
+  ASSERT_TRUE(chunk_three.has_value());
+  EXPECT_EQ(chunk_three->data.stream_id, StreamID(5));
+  EXPECT_EQ(chunk_three->data.ppid, PPID(55));
+
+  EXPECT_FALSE(buf_.IsFull());
+  EXPECT_TRUE(buf_.IsEmpty());
+}
+
+TEST_F(RRSendQueueTest, DefaultsToOrderedSend) {
+  std::vector<uint8_t> payload(20);
+
+  // Default is ordered
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  absl::optional<SendQueue::DataToSend> chunk_one =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_FALSE(chunk_one->data.is_unordered);
+
+  // Explicitly unordered.
+  SendOptions opts;
+  opts.unordered = IsUnordered(true);
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload), opts);
+  absl::optional<SendQueue::DataToSend> chunk_two =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_two.has_value());
+  EXPECT_TRUE(chunk_two->data.is_unordered);
+}
+
+TEST_F(RRSendQueueTest, ProduceWithLifetimeExpiry) {
+  std::vector<uint8_t> payload(20);
+
+  // Default is no expiry
+  TimeMs now = kNow;
+  buf_.Add(now, DcSctpMessage(kStreamID, kPPID, payload));
+  now += DurationMs(1000000);
+  ASSERT_TRUE(buf_.Produce(now, kOneFragmentPacketSize));
+
+  SendOptions expires_2_seconds;
+  expires_2_seconds.lifetime = DurationMs(2000);
+
+  // Add and consume within lifetime
+  buf_.Add(now, DcSctpMessage(kStreamID, kPPID, payload), expires_2_seconds);
+  now += DurationMs(2000);
+  ASSERT_TRUE(buf_.Produce(now, kOneFragmentPacketSize));
+
+  // Add and consume just outside lifetime
+  buf_.Add(now, DcSctpMessage(kStreamID, kPPID, payload), expires_2_seconds);
+  now += DurationMs(2001);
+  ASSERT_FALSE(buf_.Produce(now, kOneFragmentPacketSize));
+
+  // A long time after expiry
+  buf_.Add(now, DcSctpMessage(kStreamID, kPPID, payload), expires_2_seconds);
+  now += DurationMs(1000000);
+  ASSERT_FALSE(buf_.Produce(now, kOneFragmentPacketSize));
+
+  // Expire one message, but produce the second that is not expired.
+  buf_.Add(now, DcSctpMessage(kStreamID, kPPID, payload), expires_2_seconds);
+
+  SendOptions expires_4_seconds;
+  expires_4_seconds.lifetime = DurationMs(4000);
+
+  buf_.Add(now, DcSctpMessage(kStreamID, kPPID, payload), expires_4_seconds);
+  now += DurationMs(2001);
+
+  ASSERT_TRUE(buf_.Produce(now, kOneFragmentPacketSize));
+  ASSERT_FALSE(buf_.Produce(now, kOneFragmentPacketSize));
+}
+
+TEST_F(RRSendQueueTest, DiscardPartialPackets) {
+  std::vector<uint8_t> payload(120);
+
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), PPID(54), payload));
+
+  absl::optional<SendQueue::DataToSend> chunk_one =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_FALSE(chunk_one->data.is_end);
+  EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
+  buf_.Discard(IsUnordered(false), chunk_one->data.stream_id,
+               chunk_one->data.message_id);
+
+  absl::optional<SendQueue::DataToSend> chunk_two =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_two.has_value());
+  EXPECT_FALSE(chunk_two->data.is_end);
+  EXPECT_EQ(chunk_two->data.stream_id, StreamID(2));
+
+  absl::optional<SendQueue::DataToSend> chunk_three =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_three.has_value());
+  EXPECT_TRUE(chunk_three->data.is_end);
+  EXPECT_EQ(chunk_three->data.stream_id, StreamID(2));
+  ASSERT_FALSE(buf_.Produce(kNow, kOneFragmentPacketSize));
+
+  // Calling it again shouldn't cause issues.
+  buf_.Discard(IsUnordered(false), chunk_one->data.stream_id,
+               chunk_one->data.message_id);
+  ASSERT_FALSE(buf_.Produce(kNow, kOneFragmentPacketSize));
+}
+
+TEST_F(RRSendQueueTest, PrepareResetStreamsDiscardsStream) {
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, {1, 2, 3}));
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), PPID(54), {1, 2, 3, 4, 5}));
+  EXPECT_EQ(buf_.total_buffered_amount(), 8u);
+
+  buf_.PrepareResetStream(StreamID(1));
+  EXPECT_EQ(buf_.total_buffered_amount(), 5u);
+
+  EXPECT_THAT(buf_.GetStreamsReadyToBeReset(),
+              UnorderedElementsAre(StreamID(1)));
+  buf_.CommitResetStreams();
+  buf_.PrepareResetStream(StreamID(2));
+  EXPECT_EQ(buf_.total_buffered_amount(), 0u);
+}
+
+TEST_F(RRSendQueueTest, PrepareResetStreamsNotPartialPackets) {
+  std::vector<uint8_t> payload(120);
+
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+
+  absl::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 50);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
+  EXPECT_EQ(buf_.total_buffered_amount(), 2 * payload.size() - 50);
+
+  buf_.PrepareResetStream(StreamID(1));
+  EXPECT_EQ(buf_.total_buffered_amount(), payload.size() - 50);
+}
+
+TEST_F(RRSendQueueTest, EnqueuedItemsArePausedDuringStreamReset) {
+  std::vector<uint8_t> payload(50);
+
+  buf_.PrepareResetStream(StreamID(1));
+  EXPECT_EQ(buf_.total_buffered_amount(), 0u);
+
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  EXPECT_EQ(buf_.total_buffered_amount(), payload.size());
+
+  EXPECT_FALSE(buf_.Produce(kNow, kOneFragmentPacketSize).has_value());
+
+  EXPECT_TRUE(buf_.HasStreamsReadyToBeReset());
+  EXPECT_THAT(buf_.GetStreamsReadyToBeReset(),
+              UnorderedElementsAre(StreamID(1)));
+
+  EXPECT_FALSE(buf_.Produce(kNow, kOneFragmentPacketSize).has_value());
+
+  buf_.CommitResetStreams();
+  EXPECT_EQ(buf_.total_buffered_amount(), payload.size());
+
+  absl::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 50);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
+  EXPECT_EQ(buf_.total_buffered_amount(), 0u);
+}
+
+TEST_F(RRSendQueueTest, PausedStreamsStillSendPartialMessagesUntilEnd) {
+  constexpr size_t kPayloadSize = 100;
+  constexpr size_t kFragmentSize = 50;
+  std::vector<uint8_t> payload(kPayloadSize);
+
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+
+  absl::optional<SendQueue::DataToSend> chunk_one =
+      buf_.Produce(kNow, kFragmentSize);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
+  EXPECT_EQ(buf_.total_buffered_amount(), 2 * kPayloadSize - kFragmentSize);
+
+  // This will stop the second message from being sent.
+  buf_.PrepareResetStream(StreamID(1));
+  EXPECT_EQ(buf_.total_buffered_amount(), 1 * kPayloadSize - kFragmentSize);
+
+  // Should still produce fragments until end of message.
+  absl::optional<SendQueue::DataToSend> chunk_two =
+      buf_.Produce(kNow, kFragmentSize);
+  ASSERT_TRUE(chunk_two.has_value());
+  EXPECT_EQ(chunk_two->data.stream_id, kStreamID);
+  EXPECT_EQ(buf_.total_buffered_amount(), 0ul);
+
+  // But shouldn't produce any more messages as the stream is paused.
+  EXPECT_FALSE(buf_.Produce(kNow, kFragmentSize).has_value());
+}
+
+TEST_F(RRSendQueueTest, CommittingResetsSSN) {
+  std::vector<uint8_t> payload(50);
+
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+
+  absl::optional<SendQueue::DataToSend> chunk_one =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_EQ(chunk_one->data.ssn, SSN(0));
+
+  absl::optional<SendQueue::DataToSend> chunk_two =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_two.has_value());
+  EXPECT_EQ(chunk_two->data.ssn, SSN(1));
+
+  buf_.PrepareResetStream(StreamID(1));
+
+  // Buffered
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+
+  EXPECT_TRUE(buf_.HasStreamsReadyToBeReset());
+  EXPECT_THAT(buf_.GetStreamsReadyToBeReset(),
+              UnorderedElementsAre(StreamID(1)));
+  buf_.CommitResetStreams();
+
+  absl::optional<SendQueue::DataToSend> chunk_three =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_three.has_value());
+  EXPECT_EQ(chunk_three->data.ssn, SSN(0));
+}
+
+TEST_F(RRSendQueueTest, CommittingResetsSSNForPausedStreamsOnly) {
+  std::vector<uint8_t> payload(50);
+
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, payload));
+  buf_.Add(kNow, DcSctpMessage(StreamID(3), kPPID, payload));
+
+  absl::optional<SendQueue::DataToSend> chunk_one =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_EQ(chunk_one->data.stream_id, StreamID(1));
+  EXPECT_EQ(chunk_one->data.ssn, SSN(0));
+
+  absl::optional<SendQueue::DataToSend> chunk_two =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_two.has_value());
+  EXPECT_EQ(chunk_two->data.stream_id, StreamID(3));
+  EXPECT_EQ(chunk_two->data.ssn, SSN(0));
+
+  buf_.PrepareResetStream(StreamID(3));
+
+  // Send two more messages - SID 3 will buffer, SID 1 will send.
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, payload));
+  buf_.Add(kNow, DcSctpMessage(StreamID(3), kPPID, payload));
+
+  EXPECT_TRUE(buf_.HasStreamsReadyToBeReset());
+  EXPECT_THAT(buf_.GetStreamsReadyToBeReset(),
+              UnorderedElementsAre(StreamID(3)));
+
+  buf_.CommitResetStreams();
+
+  absl::optional<SendQueue::DataToSend> chunk_three =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_three.has_value());
+  EXPECT_EQ(chunk_three->data.stream_id, StreamID(1));
+  EXPECT_EQ(chunk_three->data.ssn, SSN(1));
+
+  absl::optional<SendQueue::DataToSend> chunk_four =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_four.has_value());
+  EXPECT_EQ(chunk_four->data.stream_id, StreamID(3));
+  EXPECT_EQ(chunk_four->data.ssn, SSN(0));
+}
+
+TEST_F(RRSendQueueTest, RollBackResumesSSN) {
+  std::vector<uint8_t> payload(50);
+
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+
+  absl::optional<SendQueue::DataToSend> chunk_one =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_EQ(chunk_one->data.ssn, SSN(0));
+
+  absl::optional<SendQueue::DataToSend> chunk_two =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_two.has_value());
+  EXPECT_EQ(chunk_two->data.ssn, SSN(1));
+
+  buf_.PrepareResetStream(StreamID(1));
+
+  // Buffered
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+
+  EXPECT_TRUE(buf_.HasStreamsReadyToBeReset());
+  EXPECT_THAT(buf_.GetStreamsReadyToBeReset(),
+              UnorderedElementsAre(StreamID(1)));
+  buf_.RollbackResetStreams();
+
+  absl::optional<SendQueue::DataToSend> chunk_three =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_three.has_value());
+  EXPECT_EQ(chunk_three->data.ssn, SSN(2));
+}
+
+TEST_F(RRSendQueueTest, ReturnsFragmentsForOneMessageBeforeMovingToNext) {
+  std::vector<uint8_t> payload(200);
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, payload));
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), kPPID, payload));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk1,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk1.data.stream_id, StreamID(1));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk2,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk2.data.stream_id, StreamID(1));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk3,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk3.data.stream_id, StreamID(2));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk4,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk4.data.stream_id, StreamID(2));
+}
+
+TEST_F(RRSendQueueTest, ReturnsAlsoSmallFragmentsBeforeMovingToNext) {
+  std::vector<uint8_t> payload(kTwoFragmentPacketSize);
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, payload));
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), kPPID, payload));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk1,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk1.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk1.data.payload, SizeIs(kOneFragmentPacketSize));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk2,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk2.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk2.data.payload,
+              SizeIs(kTwoFragmentPacketSize - kOneFragmentPacketSize));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk3,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk3.data.stream_id, StreamID(2));
+  EXPECT_THAT(chunk3.data.payload, SizeIs(kOneFragmentPacketSize));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk4,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk4.data.stream_id, StreamID(2));
+  EXPECT_THAT(chunk4.data.payload,
+              SizeIs(kTwoFragmentPacketSize - kOneFragmentPacketSize));
+}
+
+TEST_F(RRSendQueueTest, WillCycleInRoundRobinFashionBetweenStreams) {
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(1)));
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(2)));
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), kPPID, std::vector<uint8_t>(3)));
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), kPPID, std::vector<uint8_t>(4)));
+  buf_.Add(kNow, DcSctpMessage(StreamID(3), kPPID, std::vector<uint8_t>(5)));
+  buf_.Add(kNow, DcSctpMessage(StreamID(3), kPPID, std::vector<uint8_t>(6)));
+  buf_.Add(kNow, DcSctpMessage(StreamID(4), kPPID, std::vector<uint8_t>(7)));
+  buf_.Add(kNow, DcSctpMessage(StreamID(4), kPPID, std::vector<uint8_t>(8)));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk1,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk1.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk1.data.payload, SizeIs(1));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk2,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk2.data.stream_id, StreamID(2));
+  EXPECT_THAT(chunk2.data.payload, SizeIs(3));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk3,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk3.data.stream_id, StreamID(3));
+  EXPECT_THAT(chunk3.data.payload, SizeIs(5));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk4,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk4.data.stream_id, StreamID(4));
+  EXPECT_THAT(chunk4.data.payload, SizeIs(7));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk5,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk5.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk5.data.payload, SizeIs(2));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk6,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk6.data.stream_id, StreamID(2));
+  EXPECT_THAT(chunk6.data.payload, SizeIs(4));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk7,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk7.data.stream_id, StreamID(3));
+  EXPECT_THAT(chunk7.data.payload, SizeIs(6));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk8,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk8.data.stream_id, StreamID(4));
+  EXPECT_THAT(chunk8.data.payload, SizeIs(8));
+}
+
+TEST_F(RRSendQueueTest, DoesntTriggerOnBufferedAmountLowWhenSetToZero) {
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow).Times(0);
+  buf_.SetBufferedAmountLowThreshold(StreamID(1), 0u);
+}
+
+TEST_F(RRSendQueueTest, TriggersOnBufferedAmountAtZeroLowWhenSent) {
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(1)));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 1u);
+
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow(StreamID(1)));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk1,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk1.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk1.data.payload, SizeIs(1));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 0u);
+}
+
+TEST_F(RRSendQueueTest, WillRetriggerOnBufferedAmountLowIfAddingMore) {
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(1)));
+
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow(StreamID(1)));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk1,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk1.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk1.data.payload, SizeIs(1));
+
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow).Times(0);
+
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(1)));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 1u);
+
+  // Should now trigger again, as buffer_amount went above the threshold.
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow(StreamID(1)));
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk2,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk2.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk2.data.payload, SizeIs(1));
+}
+
+TEST_F(RRSendQueueTest, OnlyTriggersWhenTransitioningFromAboveToBelowOrEqual) {
+  buf_.SetBufferedAmountLowThreshold(StreamID(1), 1000);
+
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(10)));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 10u);
+
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow).Times(0);
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk1,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk1.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk1.data.payload, SizeIs(10));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 0u);
+
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(20)));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 20u);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk2,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk2.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk2.data.payload, SizeIs(20));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 0u);
+}
+
+TEST_F(RRSendQueueTest, WillTriggerOnBufferedAmountLowSetAboveZero) {
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow).Times(0);
+
+  buf_.SetBufferedAmountLowThreshold(StreamID(1), 700);
+
+  std::vector<uint8_t> payload(1000);
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, payload));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk1,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk1.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk1.data.payload, SizeIs(kOneFragmentPacketSize));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 900u);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk2,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk2.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk2.data.payload, SizeIs(kOneFragmentPacketSize));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 800u);
+
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow(StreamID(1)));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk3,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk3.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk3.data.payload, SizeIs(kOneFragmentPacketSize));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 700u);
+
+  // Doesn't trigger when reducing even further.
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow).Times(0);
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk4,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk3.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk3.data.payload, SizeIs(kOneFragmentPacketSize));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 600u);
+}
+
+TEST_F(RRSendQueueTest, WillRetriggerOnBufferedAmountLowSetAboveZero) {
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow).Times(0);
+
+  buf_.SetBufferedAmountLowThreshold(StreamID(1), 700);
+
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(1000)));
+
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow(StreamID(1)));
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk1,
+                              buf_.Produce(kNow, 400));
+  EXPECT_EQ(chunk1.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk1.data.payload, SizeIs(400));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 600u);
+
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow).Times(0);
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(200)));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 800u);
+
+  // Will trigger again, as it went above the limit.
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow(StreamID(1)));
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk2,
+                              buf_.Produce(kNow, 200));
+  EXPECT_EQ(chunk2.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk2.data.payload, SizeIs(200));
+  EXPECT_EQ(buf_.buffered_amount(StreamID(1)), 600u);
+}
+
+TEST_F(RRSendQueueTest, TriggersOnBufferedAmountLowOnThresholdChanged) {
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow).Times(0);
+
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(100)));
+
+  // Modifying the threshold, still under buffered_amount, should not trigger.
+  buf_.SetBufferedAmountLowThreshold(StreamID(1), 50);
+  buf_.SetBufferedAmountLowThreshold(StreamID(1), 99);
+
+  // When the threshold reaches buffered_amount, it will trigger.
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow(StreamID(1)));
+  buf_.SetBufferedAmountLowThreshold(StreamID(1), 100);
+
+  // But not when it's set low again.
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow).Times(0);
+  buf_.SetBufferedAmountLowThreshold(StreamID(1), 50);
+
+  // But it will trigger when it overshoots.
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow(StreamID(1)));
+  buf_.SetBufferedAmountLowThreshold(StreamID(1), 150);
+
+  // But not when it's set low again.
+  EXPECT_CALL(callbacks_, OnBufferedAmountLow).Times(0);
+  buf_.SetBufferedAmountLowThreshold(StreamID(1), 0);
+}
+
+TEST_F(RRSendQueueTest,
+       OnTotalBufferedAmountLowDoesNotTriggerOnBufferFillingUp) {
+  EXPECT_CALL(callbacks_, OnTotalBufferedAmountLow).Times(0);
+  std::vector<uint8_t> payload(kBufferedAmountLowThreshold - 1);
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  EXPECT_EQ(buf_.total_buffered_amount(), payload.size());
+
+  // Will not trigger if going above but never below.
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID,
+                               std::vector<uint8_t>(kOneFragmentPacketSize)));
+}
+
+TEST_F(RRSendQueueTest, TriggersOnTotalBufferedAmountLowWhenCrossing) {
+  EXPECT_CALL(callbacks_, OnTotalBufferedAmountLow).Times(0);
+  std::vector<uint8_t> payload(kBufferedAmountLowThreshold);
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
+  EXPECT_EQ(buf_.total_buffered_amount(), payload.size());
+
+  // Reaches it.
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, std::vector<uint8_t>(1)));
+
+  // Drain it a bit - will trigger.
+  EXPECT_CALL(callbacks_, OnTotalBufferedAmountLow).Times(1);
+  absl::optional<SendQueue::DataToSend> chunk_two =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+}
+
+TEST_F(RRSendQueueTest, WillStayInAStreamAsLongAsThatMessageIsSending) {
+  buf_.Add(kNow, DcSctpMessage(StreamID(5), kPPID, std::vector<uint8_t>(1)));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk1,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk1.data.stream_id, StreamID(5));
+  EXPECT_THAT(chunk1.data.payload, SizeIs(1));
+
+  // Next, it should pick a different stream.
+
+  buf_.Add(kNow,
+           DcSctpMessage(StreamID(1), kPPID,
+                         std::vector<uint8_t>(kOneFragmentPacketSize * 2)));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk2,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk2.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk2.data.payload, SizeIs(kOneFragmentPacketSize));
+
+  // It should still stay on the Stream1 now, even if might be tempted to switch
+  // to this stream, as it's the stream following 5.
+  buf_.Add(kNow, DcSctpMessage(StreamID(6), kPPID, std::vector<uint8_t>(1)));
+
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk3,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk3.data.stream_id, StreamID(1));
+  EXPECT_THAT(chunk3.data.payload, SizeIs(kOneFragmentPacketSize));
+
+  // After stream id 1 is complete, it's time to do stream 6.
+  ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk4,
+                              buf_.Produce(kNow, kOneFragmentPacketSize));
+  EXPECT_EQ(chunk4.data.stream_id, StreamID(6));
+  EXPECT_THAT(chunk4.data.payload, SizeIs(1));
+
+  EXPECT_FALSE(buf_.Produce(kNow, kOneFragmentPacketSize).has_value());
+}
+
+TEST_F(RRSendQueueTest, StreamsHaveInitialPriority) {
+  EXPECT_EQ(buf_.GetStreamPriority(StreamID(1)), kDefaultPriority);
+
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), kPPID, std::vector<uint8_t>(40)));
+  EXPECT_EQ(buf_.GetStreamPriority(StreamID(2)), kDefaultPriority);
+}
+
+TEST_F(RRSendQueueTest, CanChangeStreamPriority) {
+  buf_.SetStreamPriority(StreamID(1), StreamPriority(42));
+  EXPECT_EQ(buf_.GetStreamPriority(StreamID(1)), StreamPriority(42));
+
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), kPPID, std::vector<uint8_t>(40)));
+  buf_.SetStreamPriority(StreamID(2), StreamPriority(42));
+  EXPECT_EQ(buf_.GetStreamPriority(StreamID(2)), StreamPriority(42));
+}
+
+TEST_F(RRSendQueueTest, WillHandoverPriority) {
+  buf_.SetStreamPriority(StreamID(1), StreamPriority(42));
+
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), kPPID, std::vector<uint8_t>(40)));
+  buf_.SetStreamPriority(StreamID(2), StreamPriority(42));
+
+  DcSctpSocketHandoverState state;
+  buf_.AddHandoverState(state);
+
+  RRSendQueue q2("log: ", &callbacks_, kMaxQueueSize, kMtu, kDefaultPriority,
+                 kBufferedAmountLowThreshold);
+  q2.RestoreFromState(state);
+  EXPECT_EQ(q2.GetStreamPriority(StreamID(1)), StreamPriority(42));
+  EXPECT_EQ(q2.GetStreamPriority(StreamID(2)), StreamPriority(42));
+}
+
+TEST_F(RRSendQueueTest, WillSendMessagesByPrio) {
+  buf_.EnableMessageInterleaving(true);
+  buf_.SetStreamPriority(StreamID(1), StreamPriority(10));
+  buf_.SetStreamPriority(StreamID(2), StreamPriority(20));
+  buf_.SetStreamPriority(StreamID(3), StreamPriority(30));
+
+  buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, std::vector<uint8_t>(40)));
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), kPPID, std::vector<uint8_t>(20)));
+  buf_.Add(kNow, DcSctpMessage(StreamID(3), kPPID, std::vector<uint8_t>(10)));
+  std::vector<uint16_t> expected_streams = {3, 2, 2, 1, 1, 1, 1};
+
+  for (uint16_t stream_num : expected_streams) {
+    ASSERT_HAS_VALUE_AND_ASSIGN(SendQueue::DataToSend chunk,
+                                buf_.Produce(kNow, 10));
+    EXPECT_EQ(chunk.data.stream_id, StreamID(stream_num));
+  }
+  EXPECT_FALSE(buf_.Produce(kNow, 1).has_value());
+}
+
+TEST_F(RRSendQueueTest, WillSendLifecycleExpireWhenExpiredInSendQueue) {
+  std::vector<uint8_t> payload(kOneFragmentPacketSize);
+  buf_.Add(kNow, DcSctpMessage(StreamID(2), kPPID, payload),
+           SendOptions{.lifetime = DurationMs(1000),
+                       .lifecycle_id = LifecycleId(1)});
+
+  EXPECT_CALL(callbacks_, OnLifecycleMessageExpired(LifecycleId(1),
+                                                    /*maybe_delivered=*/false));
+  EXPECT_CALL(callbacks_, OnLifecycleEnd(LifecycleId(1)));
+  EXPECT_FALSE(buf_.Produce(kNow + DurationMs(1001), kOneFragmentPacketSize)
+                   .has_value());
+}
+
+TEST_F(RRSendQueueTest, WillSendLifecycleExpireWhenDiscardingDuringPause) {
+  std::vector<uint8_t> payload(120);
+
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload),
+           SendOptions{.lifecycle_id = LifecycleId(1)});
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload),
+           SendOptions{.lifecycle_id = LifecycleId(2)});
+
+  absl::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 50);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
+  EXPECT_EQ(buf_.total_buffered_amount(), 2 * payload.size() - 50);
+
+  EXPECT_CALL(callbacks_, OnLifecycleMessageExpired(LifecycleId(2),
+                                                    /*maybe_delivered=*/false));
+  EXPECT_CALL(callbacks_, OnLifecycleEnd(LifecycleId(2)));
+  buf_.PrepareResetStream(StreamID(1));
+  EXPECT_EQ(buf_.total_buffered_amount(), payload.size() - 50);
+}
+
+TEST_F(RRSendQueueTest, WillSendLifecycleExpireWhenDiscardingExplicitly) {
+  std::vector<uint8_t> payload(kOneFragmentPacketSize + 20);
+
+  buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload),
+           SendOptions{.lifecycle_id = LifecycleId(1)});
+
+  absl::optional<SendQueue::DataToSend> chunk_one =
+      buf_.Produce(kNow, kOneFragmentPacketSize);
+  ASSERT_TRUE(chunk_one.has_value());
+  EXPECT_FALSE(chunk_one->data.is_end);
+  EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
+  EXPECT_CALL(callbacks_, OnLifecycleMessageExpired(LifecycleId(1),
+                                                    /*maybe_delivered=*/false));
+  EXPECT_CALL(callbacks_, OnLifecycleEnd(LifecycleId(1)));
+  buf_.Discard(IsUnordered(false), chunk_one->data.stream_id,
+               chunk_one->data.message_id);
+}
+}  // namespace
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/send_queue.h b/third_party/libwebrtc/net/dcsctp/tx/send_queue.h
new file mode 100644
index 0000000000..0b96e9041a
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/send_queue.h
@@ -0,0 +1,142 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TX_SEND_QUEUE_H_
+#define NET_DCSCTP_TX_SEND_QUEUE_H_
+
+#include <cstdint>
+#include <limits>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+class SendQueue {
+ public:
+  // Container for a data chunk that is produced by the SendQueue
+  struct DataToSend {
+    explicit DataToSend(Data data) : data(std::move(data)) {}
+    // The data to send, including all parameters.
+    Data data;
+
+    // Partial reliability - RFC3758
+    MaxRetransmits max_retransmissions = MaxRetransmits::NoLimit();
+    TimeMs expires_at = TimeMs::InfiniteFuture();
+
+    // Lifecycle - set for the last fragment, and `LifecycleId::NotSet()` for
+    // all other fragments.
+    LifecycleId lifecycle_id = LifecycleId::NotSet();
+  };
+
+  virtual ~SendQueue() = default;
+
+  // TODO(boivie): This interface is obviously missing an "Add" function, but
+  // that is postponed a bit until the story around how to model message
+  // prioritization, which is important for any advanced stream scheduler, is
+  // further clarified.
+
+  // Produce a chunk to be sent.
+  //
+  // `max_size` refers to how many payload bytes that may be produced, not
+  // including any headers.
+  virtual absl::optional<DataToSend> Produce(TimeMs now, size_t max_size) = 0;
+
+  // Discards a partially sent message identified by the parameters `unordered`,
+  // `stream_id` and `message_id`. The `message_id` comes from the returned
+  // information when having called `Produce`. A partially sent message means
+  // that it has had at least one fragment of it returned when `Produce` was
+  // called prior to calling this method).
+  //
+  // This is used when a message has been found to be expired (by the partial
+  // reliability extension), and the retransmission queue will signal the
+  // receiver that any partially received message fragments should be skipped.
+  // This means that any remaining fragments in the Send Queue must be removed
+  // as well so that they are not sent.
+  //
+  // This function returns true if this message had unsent fragments still in
+  // the queue that were discarded, and false if there were no such fragments.
+  virtual bool Discard(IsUnordered unordered,
+                       StreamID stream_id,
+                       MID message_id) = 0;
+
+  // Prepares the stream to be reset. This is used to close a WebRTC data
+  // channel and will be signaled to the other side.
+  //
+  // Concretely, it discards all whole (not partly sent) messages in the given
+  // stream and pauses that stream so that future added messages aren't
+  // produced until `ResumeStreams` is called.
+  //
+  // TODO(boivie): Investigate if it really should discard any message at all.
+  // RFC8831 only mentions that "[RFC6525] also guarantees that all the messages
+  // are delivered (or abandoned) before the stream is reset."
+  //
+  // This method can be called multiple times to add more streams to be
+  // reset, and paused while they are resetting. This is the first part of the
+  // two-phase commit protocol to reset streams, where the caller completes the
+  // procedure by either calling `CommitResetStreams` or `RollbackResetStreams`.
+  virtual void PrepareResetStream(StreamID stream_id) = 0;
+
+  // Indicates if there are any streams that are ready to be reset.
+  virtual bool HasStreamsReadyToBeReset() const = 0;
+
+  // Returns a list of streams that are ready to be included in an outgoing
+  // stream reset request. Any streams that are returned here must be included
+  // in an outgoing stream reset request, and there must not be concurrent
+  // requests. Before calling this method again, you must have called
+  virtual std::vector<StreamID> GetStreamsReadyToBeReset() = 0;
+
+  // Called to commit to reset the streams returned by
+  // `GetStreamsReadyToBeReset`. It will reset the stream sequence numbers
+  // (SSNs) and message identifiers (MIDs) and resume the paused streams.
+  virtual void CommitResetStreams() = 0;
+
+  // Called to abort the resetting of streams returned by
+  // `GetStreamsReadyToBeReset`. Will resume the paused streams without
+  // resetting the stream sequence numbers (SSNs) or message identifiers (MIDs).
+  // Note that the non-partial messages that were discarded when calling
+  // `PrepareResetStreams` will not be recovered, to better match the intention
+  // from the sender to "close the channel".
+  virtual void RollbackResetStreams() = 0;
+
+  // Resets all message identifier counters (MID, SSN) and makes all partially
+  // messages be ready to be re-sent in full. This is used when the peer has
+  // been detected to have restarted and is used to try to minimize the amount
+  // of data loss. However, data loss cannot be completely guaranteed when a
+  // peer restarts.
+  virtual void Reset() = 0;
+
+  // Returns the amount of buffered data. This doesn't include packets that are
+  // e.g. inflight.
+  virtual size_t buffered_amount(StreamID stream_id) const = 0;
+
+  // Returns the total amount of buffer data, for all streams.
+  virtual size_t total_buffered_amount() const = 0;
+
+  // Returns the limit for the `OnBufferedAmountLow` event. Default value is 0.
+  virtual size_t buffered_amount_low_threshold(StreamID stream_id) const = 0;
+
+  // Sets a limit for the `OnBufferedAmountLow` event.
+  virtual void SetBufferedAmountLowThreshold(StreamID stream_id,
+                                             size_t bytes) = 0;
+
+  // Configures the send queue to support interleaved message sending as
+  // described in RFC8260. Every send queue starts with this value set as
+  // disabled, but can later change it when the capabilities of the connection
+  // have been negotiated. This affects the behavior of the `Produce` method.
+  virtual void EnableMessageInterleaving(bool enabled) = 0;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TX_SEND_QUEUE_H_
diff --git a/third_party/libwebrtc/net/dcsctp/tx/stream_scheduler.cc b/third_party/libwebrtc/net/dcsctp/tx/stream_scheduler.cc
new file mode 100644
index 0000000000..d1560a75e4
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/stream_scheduler.cc
@@ -0,0 +1,199 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/stream_scheduler.h"
+
+#include <algorithm>
+
+#include "absl/algorithm/container.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/str_join.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/tx/send_queue.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+
+void StreamScheduler::Stream::SetPriority(StreamPriority priority) {
+  priority_ = priority;
+  inverse_weight_ = InverseWeight(priority);
+}
+
+absl::optional<SendQueue::DataToSend> StreamScheduler::Produce(
+    TimeMs now,
+    size_t max_size) {
+  // For non-interleaved streams, avoid rescheduling while still sending a
+  // message as it needs to be sent in full. For interleaved messaging,
+  // reschedule for every I-DATA chunk sent.
+  bool rescheduling =
+      enable_message_interleaving_ || !currently_sending_a_message_;
+
+  RTC_LOG(LS_VERBOSE) << "Producing data, rescheduling=" << rescheduling
+                      << ", active="
+                      << StrJoin(active_streams_, ", ",
+                                 [&](rtc::StringBuilder& sb, const auto& p) {
+                                   sb << *p->stream_id() << "@"
+                                      << *p->next_finish_time();
+                                 });
+
+  RTC_DCHECK(rescheduling || current_stream_ != nullptr);
+
+  absl::optional<SendQueue::DataToSend> data;
+  while (!data.has_value() && !active_streams_.empty()) {
+    if (rescheduling) {
+      auto it = active_streams_.begin();
+      current_stream_ = *it;
+      RTC_DLOG(LS_VERBOSE) << "Rescheduling to stream "
+                           << *current_stream_->stream_id();
+
+      active_streams_.erase(it);
+      current_stream_->ForceMarkInactive();
+    } else {
+      RTC_DLOG(LS_VERBOSE) << "Producing from previous stream: "
+                           << *current_stream_->stream_id();
+      RTC_DCHECK(absl::c_any_of(active_streams_, [this](const auto* p) {
+        return p == current_stream_;
+      }));
+    }
+
+    data = current_stream_->Produce(now, max_size);
+  }
+
+  if (!data.has_value()) {
+    RTC_DLOG(LS_VERBOSE)
+        << "There is no stream with data; Can't produce any data.";
+    RTC_DCHECK(IsConsistent());
+
+    return absl::nullopt;
+  }
+
+  RTC_DCHECK(data->data.stream_id == current_stream_->stream_id());
+
+  RTC_DLOG(LS_VERBOSE) << "Producing DATA, type="
+                       << (data->data.is_unordered ? "unordered" : "ordered")
+                       << "::"
+                       << (*data->data.is_beginning && *data->data.is_end
+                               ? "complete"
+                           : *data->data.is_beginning ? "first"
+                           : *data->data.is_end       ? "last"
+                                                      : "middle")
+                       << ", stream_id=" << *current_stream_->stream_id()
+                       << ", ppid=" << *data->data.ppid
+                       << ", length=" << data->data.payload.size();
+
+  currently_sending_a_message_ = !*data->data.is_end;
+  virtual_time_ = current_stream_->current_time();
+
+  // One side-effect of rescheduling is that the new stream will not be present
+  // in `active_streams`.
+  size_t bytes_to_send_next = current_stream_->bytes_to_send_in_next_message();
+  if (rescheduling && bytes_to_send_next > 0) {
+    current_stream_->MakeActive(bytes_to_send_next);
+  } else if (!rescheduling && bytes_to_send_next == 0) {
+    current_stream_->MakeInactive();
+  }
+
+  RTC_DCHECK(IsConsistent());
+  return data;
+}
+
+StreamScheduler::VirtualTime StreamScheduler::Stream::CalculateFinishTime(
+    size_t bytes_to_send_next) const {
+  if (parent_.enable_message_interleaving_) {
+    // Perform weighted fair queuing scheduling.
+    return VirtualTime(*current_virtual_time_ +
+                       bytes_to_send_next * *inverse_weight_);
+  }
+
+  // Perform round-robin scheduling by letting the stream have its next virtual
+  // finish time in the future. It doesn't matter how far into the future, just
+  // any positive number so that any other stream that has the same virtual
+  // finish time as this stream gets to produce their data before revisiting
+  // this stream.
+  return VirtualTime(*current_virtual_time_ + 1);
+}
+
+absl::optional<SendQueue::DataToSend> StreamScheduler::Stream::Produce(
+    TimeMs now,
+    size_t max_size) {
+  absl::optional<SendQueue::DataToSend> data = producer_.Produce(now, max_size);
+
+  if (data.has_value()) {
+    VirtualTime new_current = CalculateFinishTime(data->data.payload.size());
+    RTC_DLOG(LS_VERBOSE) << "Virtual time changed: " << *current_virtual_time_
+                         << " -> " << *new_current;
+    current_virtual_time_ = new_current;
+  }
+
+  return data;
+}
+
+bool StreamScheduler::IsConsistent() const {
+  for (Stream* stream : active_streams_) {
+    if (stream->next_finish_time_ == VirtualTime::Zero()) {
+      RTC_DLOG(LS_VERBOSE) << "Stream " << *stream->stream_id()
+                           << " is active, but has no next-finish-time";
+      return false;
+    }
+  }
+  return true;
+}
+
+void StreamScheduler::Stream::MaybeMakeActive() {
+  RTC_DLOG(LS_VERBOSE) << "MaybeMakeActive(" << *stream_id() << ")";
+  RTC_DCHECK(next_finish_time_ == VirtualTime::Zero());
+  size_t bytes_to_send_next = bytes_to_send_in_next_message();
+  if (bytes_to_send_next == 0) {
+    return;
+  }
+
+  MakeActive(bytes_to_send_next);
+}
+
+void StreamScheduler::Stream::MakeActive(size_t bytes_to_send_next) {
+  current_virtual_time_ = parent_.virtual_time_;
+  RTC_DCHECK_GT(bytes_to_send_next, 0);
+  VirtualTime next_finish_time = CalculateFinishTime(
+      std::min(bytes_to_send_next, parent_.max_payload_bytes_));
+  RTC_DCHECK_GT(*next_finish_time, 0);
+  RTC_DLOG(LS_VERBOSE) << "Making stream " << *stream_id()
+                       << " active, expiring at " << *next_finish_time;
+  RTC_DCHECK(next_finish_time_ == VirtualTime::Zero());
+  next_finish_time_ = next_finish_time;
+  RTC_DCHECK(!absl::c_any_of(parent_.active_streams_,
+                             [this](const auto* p) { return p == this; }));
+  parent_.active_streams_.emplace(this);
+}
+
+void StreamScheduler::Stream::ForceMarkInactive() {
+  RTC_DLOG(LS_VERBOSE) << "Making stream " << *stream_id() << " inactive";
+  RTC_DCHECK(next_finish_time_ != VirtualTime::Zero());
+  next_finish_time_ = VirtualTime::Zero();
+}
+
+void StreamScheduler::Stream::MakeInactive() {
+  ForceMarkInactive();
+  webrtc::EraseIf(parent_.active_streams_,
+                  [&](const auto* s) { return s == this; });
+}
+
+std::set<StreamID> StreamScheduler::ActiveStreamsForTesting() const {
+  std::set<StreamID> stream_ids;
+  for (const auto& stream : active_streams_) {
+    stream_ids.insert(stream->stream_id());
+  }
+  return stream_ids;
+}
+
+}  // namespace dcsctp
diff --git a/third_party/libwebrtc/net/dcsctp/tx/stream_scheduler.h b/third_party/libwebrtc/net/dcsctp/tx/stream_scheduler.h
new file mode 100644
index 0000000000..9c523edbfc
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/stream_scheduler.h
@@ -0,0 +1,222 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_TX_STREAM_SCHEDULER_H_
+#define NET_DCSCTP_TX_STREAM_SCHEDULER_H_
+
+#include <algorithm>
+#include <cstdint>
+#include <deque>
+#include <map>
+#include <memory>
+#include <queue>
+#include <set>
+#include <string>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/tx/send_queue.h"
+#include "rtc_base/containers/flat_set.h"
+#include "rtc_base/strong_alias.h"
+
+namespace dcsctp {
+
+// A parameterized stream scheduler. Currently, it implements the round robin
+// scheduling algorithm using virtual finish time. It is to be used as a part of
+// a send queue and will track all active streams (streams that have any data
+// that can be sent).
+//
+// The stream scheduler works with the concept of associating active streams
+// with a "virtual finish time", which is the time when a stream is allowed to
+// produce data. Streams are ordered by their virtual finish time, and the
+// "current virtual time" will advance to the next following virtual finish time
+// whenever a chunk is to be produced.
+//
+// When message interleaving is enabled, the WFQ - Weighted Fair Queueing -
+// scheduling algorithm will be used. And when it's not, round-robin scheduling
+// will be used instead.
+//
+// In the round robin scheduling algorithm, a stream's virtual finish time will
+// just increment by one (1) after having produced a chunk, which results in a
+// round-robin scheduling.
+//
+// In WFQ scheduling algorithm, a stream's virtual finish time will be defined
+// as the number of bytes in the next fragment to be sent, multiplied by the
+// inverse of the stream's priority, meaning that a high priority - or a smaller
+// fragment - results in a closer virtual finish time, compared to a stream with
+// either a lower priority or a larger fragment to be sent.
+class StreamScheduler {
+ private:
+  class VirtualTime : public webrtc::StrongAlias<class VirtualTimeTag, double> {
+   public:
+    constexpr explicit VirtualTime(const UnderlyingType& v)
+        : webrtc::StrongAlias<class VirtualTimeTag, double>(v) {}
+
+    static constexpr VirtualTime Zero() { return VirtualTime(0); }
+  };
+  class InverseWeight
+      : public webrtc::StrongAlias<class InverseWeightTag, double> {
+   public:
+    constexpr explicit InverseWeight(StreamPriority priority)
+        : webrtc::StrongAlias<class InverseWeightTag, double>(
+              1.0 / std::max(static_cast<double>(*priority), 0.000001)) {}
+  };
+
+ public:
+  class StreamProducer {
+   public:
+    virtual ~StreamProducer() = default;
+
+    // Produces a fragment of data to send. The current wall time is specified
+    // as `now` and should be used to skip chunks with expired limited lifetime.
+    // The parameter `max_size` specifies the maximum amount of actual payload
+    // that may be returned. If these constraints prevents the stream from
+    // sending some data, `absl::nullopt` should be returned.
+    virtual absl::optional<SendQueue::DataToSend> Produce(TimeMs now,
+                                                          size_t max_size) = 0;
+
+    // Returns the number of payload bytes that is scheduled to be sent in the
+    // next enqueued message, or zero if there are no enqueued messages or if
+    // the stream has been actively paused.
+    virtual size_t bytes_to_send_in_next_message() const = 0;
+  };
+
+  class Stream {
+   public:
+    StreamID stream_id() const { return stream_id_; }
+
+    StreamPriority priority() const { return priority_; }
+    void SetPriority(StreamPriority priority);
+
+    // Will activate the stream _if_ it has any data to send. That is, if the
+    // callback to `bytes_to_send_in_next_message` returns non-zero. If the
+    // callback returns zero, the stream will not be made active.
+    void MaybeMakeActive();
+
+    // Will remove the stream from the list of active streams, and will not try
+    // to produce data from it. To make it active again, call `MaybeMakeActive`.
+    void MakeInactive();
+
+    // Make the scheduler move to another message, or another stream. This is
+    // used to abort the scheduler from continuing producing fragments for the
+    // current message in case it's deleted.
+    void ForceReschedule() { parent_.ForceReschedule(); }
+
+   private:
+    friend class StreamScheduler;
+
+    Stream(StreamScheduler* parent,
+           StreamProducer* producer,
+           StreamID stream_id,
+           StreamPriority priority)
+        : parent_(*parent),
+          producer_(*producer),
+          stream_id_(stream_id),
+          priority_(priority),
+          inverse_weight_(priority) {}
+
+    // Produces a message from this stream. This will only be called on streams
+    // that have data.
+    absl::optional<SendQueue::DataToSend> Produce(TimeMs now, size_t max_size);
+
+    void MakeActive(size_t bytes_to_send_next);
+    void ForceMarkInactive();
+
+    VirtualTime current_time() const { return current_virtual_time_; }
+    VirtualTime next_finish_time() const { return next_finish_time_; }
+    size_t bytes_to_send_in_next_message() const {
+      return producer_.bytes_to_send_in_next_message();
+    }
+
+    VirtualTime CalculateFinishTime(size_t bytes_to_send_next) const;
+
+    StreamScheduler& parent_;
+    StreamProducer& producer_;
+    const StreamID stream_id_;
+    StreamPriority priority_;
+    InverseWeight inverse_weight_;
+    // This outgoing stream's "current" virtual_time.
+    VirtualTime current_virtual_time_ = VirtualTime::Zero();
+    VirtualTime next_finish_time_ = VirtualTime::Zero();
+  };
+
+  // The `mtu` parameter represents the maximum SCTP packet size, which should
+  // be the same as `DcSctpOptions::mtu`.
+  explicit StreamScheduler(size_t mtu)
+      : max_payload_bytes_(mtu - SctpPacket::kHeaderSize -
+                           IDataChunk::kHeaderSize) {}
+
+  std::unique_ptr<Stream> CreateStream(StreamProducer* producer,
+                                       StreamID stream_id,
+                                       StreamPriority priority) {
+    return absl::WrapUnique(new Stream(this, producer, stream_id, priority));
+  }
+
+  void EnableMessageInterleaving(bool enabled) {
+    enable_message_interleaving_ = enabled;
+  }
+
+  // Makes the scheduler stop producing message from the current stream and
+  // re-evaluates which stream to produce from.
+  void ForceReschedule() { currently_sending_a_message_ = false; }
+
+  // Produces a fragment of data to send. The current wall time is specified as
+  // `now` and will be used to skip chunks with expired limited lifetime. The
+  // parameter `max_size` specifies the maximum amount of actual payload that
+  // may be returned. If no data can be produced, `absl::nullopt` is returned.
+  absl::optional<SendQueue::DataToSend> Produce(TimeMs now, size_t max_size);
+
+  std::set<StreamID> ActiveStreamsForTesting() const;
+
+ private:
+  struct ActiveStreamComparator {
+    // Ordered by virtual finish time (primary), stream-id (secondary).
+    bool operator()(Stream* a, Stream* b) const {
+      VirtualTime a_vft = a->next_finish_time();
+      VirtualTime b_vft = b->next_finish_time();
+      if (a_vft == b_vft) {
+        return a->stream_id() < b->stream_id();
+      }
+      return a_vft < b_vft;
+    }
+  };
+
+  bool IsConsistent() const;
+
+  const size_t max_payload_bytes_;
+
+  // The current virtual time, as defined in the WFQ algorithm.
+  VirtualTime virtual_time_ = VirtualTime::Zero();
+
+  // The current stream to send chunks from.
+  Stream* current_stream_ = nullptr;
+
+  bool enable_message_interleaving_ = false;
+
+  // Indicates if the streams is currently sending a message, and should then
+  // - if message interleaving is not enabled - continue sending from this
+  // stream until that message has been sent in full.
+  bool currently_sending_a_message_ = false;
+
+  // The currently active streams, ordered by virtual finish time.
+  webrtc::flat_set<Stream*, ActiveStreamComparator> active_streams_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TX_STREAM_SCHEDULER_H_
diff --git a/third_party/libwebrtc/net/dcsctp/tx/stream_scheduler_test.cc b/third_party/libwebrtc/net/dcsctp/tx/stream_scheduler_test.cc
new file mode 100644
index 0000000000..58f0bc4690
--- /dev/null
+++ b/third_party/libwebrtc/net/dcsctp/tx/stream_scheduler_test.cc
@@ -0,0 +1,740 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/tx/stream_scheduler.h"
+
+#include <vector>
+
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/types.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+namespace {
+using ::testing::Return;
+using ::testing::StrictMock;
+
+constexpr size_t kMtu = 1000;
+constexpr size_t kPayloadSize = 4;
+
+MATCHER_P(HasDataWithMid, mid, "") {
+  if (!arg.has_value()) {
+    *result_listener << "There was no produced data";
+    return false;
+  }
+
+  if (arg->data.message_id != mid) {
+    *result_listener << "the produced data had mid " << *arg->data.message_id
+                     << " and not the expected " << *mid;
+    return false;
+  }
+
+  return true;
+}
+
+std::function<absl::optional<SendQueue::DataToSend>(TimeMs, size_t)>
+CreateChunk(StreamID sid, MID mid, size_t payload_size = kPayloadSize) {
+  return [sid, mid, payload_size](TimeMs now, size_t max_size) {
+    return SendQueue::DataToSend(Data(
+        sid, SSN(0), mid, FSN(0), PPID(42), std::vector<uint8_t>(payload_size),
+        Data::IsBeginning(true), Data::IsEnd(true), IsUnordered(true)));
+  };
+}
+
+std::map<StreamID, size_t> GetPacketCounts(StreamScheduler& scheduler,
+                                           size_t packets_to_generate) {
+  std::map<StreamID, size_t> packet_counts;
+  for (size_t i = 0; i < packets_to_generate; ++i) {
+    absl::optional<SendQueue::DataToSend> data =
+        scheduler.Produce(TimeMs(0), kMtu);
+    if (data.has_value()) {
+      ++packet_counts[data->data.stream_id];
+    }
+  }
+  return packet_counts;
+}
+
+class MockStreamProducer : public StreamScheduler::StreamProducer {
+ public:
+  MOCK_METHOD(absl::optional<SendQueue::DataToSend>,
+              Produce,
+              (TimeMs, size_t),
+              (override));
+  MOCK_METHOD(size_t, bytes_to_send_in_next_message, (), (const, override));
+};
+
+class TestStream {
+ public:
+  TestStream(StreamScheduler& scheduler,
+             StreamID stream_id,
+             StreamPriority priority,
+             size_t packet_size = kPayloadSize) {
+    EXPECT_CALL(producer_, Produce)
+        .WillRepeatedly(CreateChunk(stream_id, MID(0), packet_size));
+    EXPECT_CALL(producer_, bytes_to_send_in_next_message)
+        .WillRepeatedly(Return(packet_size));
+    stream_ = scheduler.CreateStream(&producer_, stream_id, priority);
+    stream_->MaybeMakeActive();
+  }
+
+  StreamScheduler::Stream& stream() { return *stream_; }
+
+ private:
+  StrictMock<MockStreamProducer> producer_;
+  std::unique_ptr<StreamScheduler::Stream> stream_;
+};
+
+// A scheduler without active streams doesn't produce data.
+TEST(StreamSchedulerTest, HasNoActiveStreams) {
+  StreamScheduler scheduler(kMtu);
+
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+// Stream properties can be set and retrieved
+TEST(StreamSchedulerTest, CanSetAndGetStreamProperties) {
+  StreamScheduler scheduler(kMtu);
+
+  StrictMock<MockStreamProducer> producer;
+  auto stream =
+      scheduler.CreateStream(&producer, StreamID(1), StreamPriority(2));
+
+  EXPECT_EQ(stream->stream_id(), StreamID(1));
+  EXPECT_EQ(stream->priority(), StreamPriority(2));
+
+  stream->SetPriority(StreamPriority(0));
+  EXPECT_EQ(stream->priority(), StreamPriority(0));
+}
+
+// A scheduler with a single stream produced packets from it.
+TEST(StreamSchedulerTest, CanProduceFromSingleStream) {
+  StreamScheduler scheduler(kMtu);
+
+  StrictMock<MockStreamProducer> producer;
+  EXPECT_CALL(producer, Produce).WillOnce(CreateChunk(StreamID(1), MID(0)));
+  EXPECT_CALL(producer, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(0));
+  auto stream =
+      scheduler.CreateStream(&producer, StreamID(1), StreamPriority(2));
+  stream->MaybeMakeActive();
+
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(0)));
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+// Switches between two streams after every packet.
+TEST(StreamSchedulerTest, WillRoundRobinBetweenStreams) {
+  StreamScheduler scheduler(kMtu);
+
+  StrictMock<MockStreamProducer> producer1;
+  EXPECT_CALL(producer1, Produce)
+      .WillOnce(CreateChunk(StreamID(1), MID(100)))
+      .WillOnce(CreateChunk(StreamID(1), MID(101)))
+      .WillOnce(CreateChunk(StreamID(1), MID(102)));
+  EXPECT_CALL(producer1, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(0));
+  auto stream1 =
+      scheduler.CreateStream(&producer1, StreamID(1), StreamPriority(2));
+  stream1->MaybeMakeActive();
+
+  StrictMock<MockStreamProducer> producer2;
+  EXPECT_CALL(producer2, Produce)
+      .WillOnce(CreateChunk(StreamID(2), MID(200)))
+      .WillOnce(CreateChunk(StreamID(2), MID(201)))
+      .WillOnce(CreateChunk(StreamID(2), MID(202)));
+  EXPECT_CALL(producer2, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(0));
+  auto stream2 =
+      scheduler.CreateStream(&producer2, StreamID(2), StreamPriority(2));
+  stream2->MaybeMakeActive();
+
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(100)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(200)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(101)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(201)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(102)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(202)));
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+// Switches between two streams after every packet, but keeps producing from the
+// same stream when a packet contains of multiple fragments.
+TEST(StreamSchedulerTest, WillRoundRobinOnlyWhenFinishedProducingChunk) {
+  StreamScheduler scheduler(kMtu);
+
+  StrictMock<MockStreamProducer> producer1;
+  EXPECT_CALL(producer1, Produce)
+      .WillOnce(CreateChunk(StreamID(1), MID(100)))
+      .WillOnce([](...) {
+        return SendQueue::DataToSend(
+            Data(StreamID(1), SSN(0), MID(101), FSN(0), PPID(42),
+                 std::vector<uint8_t>(4), Data::IsBeginning(true),
+                 Data::IsEnd(false), IsUnordered(true)));
+      })
+      .WillOnce([](...) {
+        return SendQueue::DataToSend(
+            Data(StreamID(1), SSN(0), MID(101), FSN(0), PPID(42),
+                 std::vector<uint8_t>(4), Data::IsBeginning(false),
+                 Data::IsEnd(false), IsUnordered(true)));
+      })
+      .WillOnce([](...) {
+        return SendQueue::DataToSend(
+            Data(StreamID(1), SSN(0), MID(101), FSN(0), PPID(42),
+                 std::vector<uint8_t>(4), Data::IsBeginning(false),
+                 Data::IsEnd(true), IsUnordered(true)));
+      })
+      .WillOnce(CreateChunk(StreamID(1), MID(102)));
+  EXPECT_CALL(producer1, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(0));
+  auto stream1 =
+      scheduler.CreateStream(&producer1, StreamID(1), StreamPriority(2));
+  stream1->MaybeMakeActive();
+
+  StrictMock<MockStreamProducer> producer2;
+  EXPECT_CALL(producer2, Produce)
+      .WillOnce(CreateChunk(StreamID(2), MID(200)))
+      .WillOnce(CreateChunk(StreamID(2), MID(201)))
+      .WillOnce(CreateChunk(StreamID(2), MID(202)));
+  EXPECT_CALL(producer2, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(0));
+  auto stream2 =
+      scheduler.CreateStream(&producer2, StreamID(2), StreamPriority(2));
+  stream2->MaybeMakeActive();
+
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(100)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(200)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(101)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(101)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(101)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(201)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(102)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(202)));
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+// Deactivates a stream before it has finished producing all packets.
+TEST(StreamSchedulerTest, StreamsCanBeMadeInactive) {
+  StreamScheduler scheduler(kMtu);
+
+  StrictMock<MockStreamProducer> producer1;
+  EXPECT_CALL(producer1, Produce)
+      .WillOnce(CreateChunk(StreamID(1), MID(100)))
+      .WillOnce(CreateChunk(StreamID(1), MID(101)));
+  EXPECT_CALL(producer1, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize));  // hints that there is a MID(2) coming.
+  auto stream1 =
+      scheduler.CreateStream(&producer1, StreamID(1), StreamPriority(2));
+  stream1->MaybeMakeActive();
+
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(100)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(101)));
+
+  // ... but the stream is made inactive before it can be produced.
+  stream1->MakeInactive();
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+// Resumes a paused stream - makes a stream active after inactivating it.
+TEST(StreamSchedulerTest, SingleStreamCanBeResumed) {
+  StreamScheduler scheduler(kMtu);
+
+  StrictMock<MockStreamProducer> producer1;
+  // Callbacks are setup so that they hint that there is a MID(2) coming...
+  EXPECT_CALL(producer1, Produce)
+      .WillOnce(CreateChunk(StreamID(1), MID(100)))
+      .WillOnce(CreateChunk(StreamID(1), MID(101)))
+      .WillOnce(CreateChunk(StreamID(1), MID(102)));
+  EXPECT_CALL(producer1, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))  // When making active again
+      .WillOnce(Return(0));
+  auto stream1 =
+      scheduler.CreateStream(&producer1, StreamID(1), StreamPriority(2));
+  stream1->MaybeMakeActive();
+
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(100)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(101)));
+
+  stream1->MakeInactive();
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+  stream1->MaybeMakeActive();
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(102)));
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+// Iterates between streams, where one is suddenly paused and later resumed.
+TEST(StreamSchedulerTest, WillRoundRobinWithPausedStream) {
+  StreamScheduler scheduler(kMtu);
+
+  StrictMock<MockStreamProducer> producer1;
+  EXPECT_CALL(producer1, Produce)
+      .WillOnce(CreateChunk(StreamID(1), MID(100)))
+      .WillOnce(CreateChunk(StreamID(1), MID(101)))
+      .WillOnce(CreateChunk(StreamID(1), MID(102)));
+  EXPECT_CALL(producer1, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(0));
+  auto stream1 =
+      scheduler.CreateStream(&producer1, StreamID(1), StreamPriority(2));
+  stream1->MaybeMakeActive();
+
+  StrictMock<MockStreamProducer> producer2;
+  EXPECT_CALL(producer2, Produce)
+      .WillOnce(CreateChunk(StreamID(2), MID(200)))
+      .WillOnce(CreateChunk(StreamID(2), MID(201)))
+      .WillOnce(CreateChunk(StreamID(2), MID(202)));
+  EXPECT_CALL(producer2, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(0));
+  auto stream2 =
+      scheduler.CreateStream(&producer2, StreamID(2), StreamPriority(2));
+  stream2->MaybeMakeActive();
+
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(100)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(200)));
+  stream1->MakeInactive();
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(201)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(202)));
+  stream1->MaybeMakeActive();
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(101)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(102)));
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+// Verifies that packet counts are evenly distributed in round robin scheduling.
+TEST(StreamSchedulerTest, WillDistributeRoundRobinPacketsEvenlyTwoStreams) {
+  StreamScheduler scheduler(kMtu);
+  TestStream stream1(scheduler, StreamID(1), StreamPriority(1));
+  TestStream stream2(scheduler, StreamID(2), StreamPriority(1));
+
+  std::map<StreamID, size_t> packet_counts = GetPacketCounts(scheduler, 10);
+  EXPECT_EQ(packet_counts[StreamID(1)], 5U);
+  EXPECT_EQ(packet_counts[StreamID(2)], 5U);
+}
+
+// Verifies that packet counts are evenly distributed among active streams,
+// where a stream is suddenly made inactive, two are added, and then the paused
+// stream is resumed.
+TEST(StreamSchedulerTest, WillDistributeEvenlyWithPausedAndAddedStreams) {
+  StreamScheduler scheduler(kMtu);
+  TestStream stream1(scheduler, StreamID(1), StreamPriority(1));
+  TestStream stream2(scheduler, StreamID(2), StreamPriority(1));
+
+  std::map<StreamID, size_t> packet_counts = GetPacketCounts(scheduler, 10);
+  EXPECT_EQ(packet_counts[StreamID(1)], 5U);
+  EXPECT_EQ(packet_counts[StreamID(2)], 5U);
+
+  stream2.stream().MakeInactive();
+
+  TestStream stream3(scheduler, StreamID(3), StreamPriority(1));
+  TestStream stream4(scheduler, StreamID(4), StreamPriority(1));
+
+  std::map<StreamID, size_t> counts2 = GetPacketCounts(scheduler, 15);
+  EXPECT_EQ(counts2[StreamID(1)], 5U);
+  EXPECT_EQ(counts2[StreamID(2)], 0U);
+  EXPECT_EQ(counts2[StreamID(3)], 5U);
+  EXPECT_EQ(counts2[StreamID(4)], 5U);
+
+  stream2.stream().MaybeMakeActive();
+
+  std::map<StreamID, size_t> counts3 = GetPacketCounts(scheduler, 20);
+  EXPECT_EQ(counts3[StreamID(1)], 5U);
+  EXPECT_EQ(counts3[StreamID(2)], 5U);
+  EXPECT_EQ(counts3[StreamID(3)], 5U);
+  EXPECT_EQ(counts3[StreamID(4)], 5U);
+}
+
+// Degrades to fair queuing with streams having identical priority.
+TEST(StreamSchedulerTest, WillDoFairQueuingWithSamePriority) {
+  StreamScheduler scheduler(kMtu);
+  scheduler.EnableMessageInterleaving(true);
+
+  constexpr size_t kSmallPacket = 30;
+  constexpr size_t kLargePacket = 70;
+
+  StrictMock<MockStreamProducer> callback1;
+  EXPECT_CALL(callback1, Produce)
+      .WillOnce(CreateChunk(StreamID(1), MID(100), kSmallPacket))
+      .WillOnce(CreateChunk(StreamID(1), MID(101), kSmallPacket))
+      .WillOnce(CreateChunk(StreamID(1), MID(102), kSmallPacket));
+  EXPECT_CALL(callback1, bytes_to_send_in_next_message)
+      .WillOnce(Return(kSmallPacket))  // When making active
+      .WillOnce(Return(kSmallPacket))
+      .WillOnce(Return(kSmallPacket))
+      .WillOnce(Return(0));
+  auto stream1 =
+      scheduler.CreateStream(&callback1, StreamID(1), StreamPriority(2));
+  stream1->MaybeMakeActive();
+
+  StrictMock<MockStreamProducer> callback2;
+  EXPECT_CALL(callback2, Produce)
+      .WillOnce(CreateChunk(StreamID(2), MID(200), kLargePacket))
+      .WillOnce(CreateChunk(StreamID(2), MID(201), kLargePacket))
+      .WillOnce(CreateChunk(StreamID(2), MID(202), kLargePacket));
+  EXPECT_CALL(callback2, bytes_to_send_in_next_message)
+      .WillOnce(Return(kLargePacket))  // When making active
+      .WillOnce(Return(kLargePacket))
+      .WillOnce(Return(kLargePacket))
+      .WillOnce(Return(0));
+  auto stream2 =
+      scheduler.CreateStream(&callback2, StreamID(2), StreamPriority(2));
+  stream2->MaybeMakeActive();
+
+  // t = 30
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(100)));
+  // t = 60
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(101)));
+  // t = 70
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(200)));
+  // t = 90
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(102)));
+  // t = 140
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(201)));
+  // t = 210
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(202)));
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+// Will do weighted fair queuing with three streams having different priority.
+TEST(StreamSchedulerTest, WillDoWeightedFairQueuingSameSizeDifferentPriority) {
+  StreamScheduler scheduler(kMtu);
+  scheduler.EnableMessageInterleaving(true);
+
+  StrictMock<MockStreamProducer> callback1;
+  EXPECT_CALL(callback1, Produce)
+      .WillOnce(CreateChunk(StreamID(1), MID(100)))
+      .WillOnce(CreateChunk(StreamID(1), MID(101)))
+      .WillOnce(CreateChunk(StreamID(1), MID(102)));
+  EXPECT_CALL(callback1, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(0));
+  // Priority 125 -> allowed to produce every 1000/125 ~= 80 time units.
+  auto stream1 =
+      scheduler.CreateStream(&callback1, StreamID(1), StreamPriority(125));
+  stream1->MaybeMakeActive();
+
+  StrictMock<MockStreamProducer> callback2;
+  EXPECT_CALL(callback2, Produce)
+      .WillOnce(CreateChunk(StreamID(2), MID(200)))
+      .WillOnce(CreateChunk(StreamID(2), MID(201)))
+      .WillOnce(CreateChunk(StreamID(2), MID(202)));
+  EXPECT_CALL(callback2, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(0));
+  // Priority 200 -> allowed to produce every 1000/200 ~= 50 time units.
+  auto stream2 =
+      scheduler.CreateStream(&callback2, StreamID(2), StreamPriority(200));
+  stream2->MaybeMakeActive();
+
+  StrictMock<MockStreamProducer> callback3;
+  EXPECT_CALL(callback3, Produce)
+      .WillOnce(CreateChunk(StreamID(3), MID(300)))
+      .WillOnce(CreateChunk(StreamID(3), MID(301)))
+      .WillOnce(CreateChunk(StreamID(3), MID(302)));
+  EXPECT_CALL(callback3, bytes_to_send_in_next_message)
+      .WillOnce(Return(kPayloadSize))  // When making active
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(kPayloadSize))
+      .WillOnce(Return(0));
+  // Priority 500 -> allowed to produce every 1000/500 ~= 20 time units.
+  auto stream3 =
+      scheduler.CreateStream(&callback3, StreamID(3), StreamPriority(500));
+  stream3->MaybeMakeActive();
+
+  // t ~= 20
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(300)));
+  // t ~= 40
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(301)));
+  // t ~= 50
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(200)));
+  // t ~= 60
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(302)));
+  // t ~= 80
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(100)));
+  // t ~= 100
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(201)));
+  // t ~= 150
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(202)));
+  // t ~= 160
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(101)));
+  // t ~= 240
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(102)));
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+// Will do weighted fair queuing with three streams having different priority
+// and sending different payload sizes.
+TEST(StreamSchedulerTest, WillDoWeightedFairQueuingDifferentSizeAndPriority) {
+  StreamScheduler scheduler(kMtu);
+  scheduler.EnableMessageInterleaving(true);
+
+  constexpr size_t kSmallPacket = 20;
+  constexpr size_t kMediumPacket = 50;
+  constexpr size_t kLargePacket = 70;
+
+  // Stream with priority = 125 -> inverse weight ~=80
+  StrictMock<MockStreamProducer> callback1;
+  EXPECT_CALL(callback1, Produce)
+      // virtual finish time ~ 0 + 50 * 80 = 4000
+      .WillOnce(CreateChunk(StreamID(1), MID(100), kMediumPacket))
+      // virtual finish time ~ 4000 + 20 * 80 = 5600
+      .WillOnce(CreateChunk(StreamID(1), MID(101), kSmallPacket))
+      // virtual finish time ~ 5600 + 70 * 80 = 11200
+      .WillOnce(CreateChunk(StreamID(1), MID(102), kLargePacket));
+  EXPECT_CALL(callback1, bytes_to_send_in_next_message)
+      .WillOnce(Return(kMediumPacket))  // When making active
+      .WillOnce(Return(kSmallPacket))
+      .WillOnce(Return(kLargePacket))
+      .WillOnce(Return(0));
+  auto stream1 =
+      scheduler.CreateStream(&callback1, StreamID(1), StreamPriority(125));
+  stream1->MaybeMakeActive();
+
+  // Stream with priority = 200 -> inverse weight ~=50
+  StrictMock<MockStreamProducer> callback2;
+  EXPECT_CALL(callback2, Produce)
+      // virtual finish time ~ 0 + 50 * 50 = 2500
+      .WillOnce(CreateChunk(StreamID(2), MID(200), kMediumPacket))
+      // virtual finish time ~ 2500 + 70 * 50 = 6000
+      .WillOnce(CreateChunk(StreamID(2), MID(201), kLargePacket))
+      // virtual finish time ~ 6000 + 20 * 50 = 7000
+      .WillOnce(CreateChunk(StreamID(2), MID(202), kSmallPacket));
+  EXPECT_CALL(callback2, bytes_to_send_in_next_message)
+      .WillOnce(Return(kMediumPacket))  // When making active
+      .WillOnce(Return(kLargePacket))
+      .WillOnce(Return(kSmallPacket))
+      .WillOnce(Return(0));
+  auto stream2 =
+      scheduler.CreateStream(&callback2, StreamID(2), StreamPriority(200));
+  stream2->MaybeMakeActive();
+
+  // Stream with priority = 500 -> inverse weight ~=20
+  StrictMock<MockStreamProducer> callback3;
+  EXPECT_CALL(callback3, Produce)
+      // virtual finish time ~ 0 + 20 * 20 = 400
+      .WillOnce(CreateChunk(StreamID(3), MID(300), kSmallPacket))
+      // virtual finish time ~ 400 + 50 * 20 = 1400
+      .WillOnce(CreateChunk(StreamID(3), MID(301), kMediumPacket))
+      // virtual finish time ~ 1400 + 70 * 20 = 2800
+      .WillOnce(CreateChunk(StreamID(3), MID(302), kLargePacket));
+  EXPECT_CALL(callback3, bytes_to_send_in_next_message)
+      .WillOnce(Return(kSmallPacket))  // When making active
+      .WillOnce(Return(kMediumPacket))
+      .WillOnce(Return(kLargePacket))
+      .WillOnce(Return(0));
+  auto stream3 =
+      scheduler.CreateStream(&callback3, StreamID(3), StreamPriority(500));
+  stream3->MaybeMakeActive();
+
+  // t ~= 400
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(300)));
+  // t ~= 1400
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(301)));
+  // t ~= 2500
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(200)));
+  // t ~= 2800
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(302)));
+  // t ~= 4000
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(100)));
+  // t ~= 5600
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(101)));
+  // t ~= 6000
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(201)));
+  // t ~= 7000
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(202)));
+  // t ~= 11200
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(102)));
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+TEST(StreamSchedulerTest, WillDistributeWFQPacketsInTwoStreamsByPriority) {
+  // A simple test with two streams of different priority, but sending packets
+  // of identical size. Verifies that the ratio of sent packets represent their
+  // priority.
+  StreamScheduler scheduler(kMtu);
+  scheduler.EnableMessageInterleaving(true);
+
+  TestStream stream1(scheduler, StreamID(1), StreamPriority(100), kPayloadSize);
+  TestStream stream2(scheduler, StreamID(2), StreamPriority(200), kPayloadSize);
+
+  std::map<StreamID, size_t> packet_counts = GetPacketCounts(scheduler, 15);
+  EXPECT_EQ(packet_counts[StreamID(1)], 5U);
+  EXPECT_EQ(packet_counts[StreamID(2)], 10U);
+}
+
+TEST(StreamSchedulerTest, WillDistributeWFQPacketsInFourStreamsByPriority) {
+  // Same as `WillDistributeWFQPacketsInTwoStreamsByPriority` but with more
+  // streams.
+  StreamScheduler scheduler(kMtu);
+  scheduler.EnableMessageInterleaving(true);
+
+  TestStream stream1(scheduler, StreamID(1), StreamPriority(100), kPayloadSize);
+  TestStream stream2(scheduler, StreamID(2), StreamPriority(200), kPayloadSize);
+  TestStream stream3(scheduler, StreamID(3), StreamPriority(300), kPayloadSize);
+  TestStream stream4(scheduler, StreamID(4), StreamPriority(400), kPayloadSize);
+
+  std::map<StreamID, size_t> packet_counts = GetPacketCounts(scheduler, 50);
+  EXPECT_EQ(packet_counts[StreamID(1)], 5U);
+  EXPECT_EQ(packet_counts[StreamID(2)], 10U);
+  EXPECT_EQ(packet_counts[StreamID(3)], 15U);
+  EXPECT_EQ(packet_counts[StreamID(4)], 20U);
+}
+
+TEST(StreamSchedulerTest, WillDistributeFromTwoStreamsFairly) {
+  // A simple test with two streams of different priority, but sending packets
+  // of different size. Verifies that the ratio of total packet payload
+  // represent their priority.
+  // In this example,
+  // * stream1 has priority 100 and sends packets of size 8
+  // * stream2 has priority 400 and sends packets of size 4
+  // With round robin, stream1 would get twice as many payload bytes on the wire
+  // as stream2, but with WFQ and a 4x priority increase, stream2 should 4x as
+  // many payload bytes on the wire. That translates to stream2 getting 8x as
+  // many packets on the wire as they are half as large.
+  StreamScheduler scheduler(kMtu);
+  // Enable WFQ scheduler.
+  scheduler.EnableMessageInterleaving(true);
+
+  TestStream stream1(scheduler, StreamID(1), StreamPriority(100),
+                     /*packet_size=*/8);
+  TestStream stream2(scheduler, StreamID(2), StreamPriority(400),
+                     /*packet_size=*/4);
+
+  std::map<StreamID, size_t> packet_counts = GetPacketCounts(scheduler, 90);
+  EXPECT_EQ(packet_counts[StreamID(1)], 10U);
+  EXPECT_EQ(packet_counts[StreamID(2)], 80U);
+}
+
+TEST(StreamSchedulerTest, WillDistributeFromFourStreamsFairly) {
+  // Same as `WillDistributeWeightedFairFromTwoStreamsFairly` but more
+  // complicated.
+  StreamScheduler scheduler(kMtu);
+  // Enable WFQ scheduler.
+  scheduler.EnableMessageInterleaving(true);
+
+  TestStream stream1(scheduler, StreamID(1), StreamPriority(100),
+                     /*packet_size=*/10);
+  TestStream stream2(scheduler, StreamID(2), StreamPriority(200),
+                     /*packet_size=*/10);
+  TestStream stream3(scheduler, StreamID(3), StreamPriority(200),
+                     /*packet_size=*/20);
+  TestStream stream4(scheduler, StreamID(4), StreamPriority(400),
+                     /*packet_size=*/30);
+
+  std::map<StreamID, size_t> packet_counts = GetPacketCounts(scheduler, 80);
+  // 15 packets * 10 bytes = 150 bytes at priority 100.
+  EXPECT_EQ(packet_counts[StreamID(1)], 15U);
+  // 30 packets * 10 bytes = 300 bytes at priority 200.
+  EXPECT_EQ(packet_counts[StreamID(2)], 30U);
+  // 15 packets * 20 bytes = 300 bytes at priority 200.
+  EXPECT_EQ(packet_counts[StreamID(3)], 15U);
+  // 20 packets * 30 bytes = 600 bytes at priority 400.
+  EXPECT_EQ(packet_counts[StreamID(4)], 20U);
+}
+
+// Sending large messages with small MTU will fragment the messages and produce
+// a first fragment not larger than the MTU, and will then not first send from
+// the stream with the smallest message, as their first fragment will be equally
+// small for both streams. See `LargeMessageWithLargeMtu` for the same test, but
+// with a larger MTU.
+TEST(StreamSchedulerTest, SendLargeMessageWithSmallMtu) {
+  StreamScheduler scheduler(100 + SctpPacket::kHeaderSize +
+                            IDataChunk::kHeaderSize);
+  scheduler.EnableMessageInterleaving(true);
+
+  StrictMock<MockStreamProducer> producer1;
+  EXPECT_CALL(producer1, Produce)
+      .WillOnce(CreateChunk(StreamID(1), MID(0), 100))
+      .WillOnce(CreateChunk(StreamID(1), MID(0), 100));
+  EXPECT_CALL(producer1, bytes_to_send_in_next_message)
+      .WillOnce(Return(200))  // When making active
+      .WillOnce(Return(100))
+      .WillOnce(Return(0));
+  auto stream1 =
+      scheduler.CreateStream(&producer1, StreamID(1), StreamPriority(1));
+  stream1->MaybeMakeActive();
+
+  StrictMock<MockStreamProducer> producer2;
+  EXPECT_CALL(producer2, Produce)
+      .WillOnce(CreateChunk(StreamID(2), MID(1), 100))
+      .WillOnce(CreateChunk(StreamID(2), MID(1), 50));
+  EXPECT_CALL(producer2, bytes_to_send_in_next_message)
+      .WillOnce(Return(150))  // When making active
+      .WillOnce(Return(50))
+      .WillOnce(Return(0));
+  auto stream2 =
+      scheduler.CreateStream(&producer2, StreamID(2), StreamPriority(1));
+  stream2->MaybeMakeActive();
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(0)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(1)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(1)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(0)));
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+// Sending large messages with large MTU will not fragment messages and will
+// send the message first from the stream that has the smallest message.
+TEST(StreamSchedulerTest, SendLargeMessageWithLargeMtu) {
+  StreamScheduler scheduler(200 + SctpPacket::kHeaderSize +
+                            IDataChunk::kHeaderSize);
+  scheduler.EnableMessageInterleaving(true);
+
+  StrictMock<MockStreamProducer> producer1;
+  EXPECT_CALL(producer1, Produce)
+      .WillOnce(CreateChunk(StreamID(1), MID(0), 200));
+  EXPECT_CALL(producer1, bytes_to_send_in_next_message)
+      .WillOnce(Return(200))  // When making active
+      .WillOnce(Return(0));
+  auto stream1 =
+      scheduler.CreateStream(&producer1, StreamID(1), StreamPriority(1));
+  stream1->MaybeMakeActive();
+
+  StrictMock<MockStreamProducer> producer2;
+  EXPECT_CALL(producer2, Produce)
+      .WillOnce(CreateChunk(StreamID(2), MID(1), 150));
+  EXPECT_CALL(producer2, bytes_to_send_in_next_message)
+      .WillOnce(Return(150))  // When making active
+      .WillOnce(Return(0));
+  auto stream2 =
+      scheduler.CreateStream(&producer2, StreamID(2), StreamPriority(1));
+  stream2->MaybeMakeActive();
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(1)));
+  EXPECT_THAT(scheduler.Produce(TimeMs(0), kMtu), HasDataWithMid(MID(0)));
+  EXPECT_EQ(scheduler.Produce(TimeMs(0), kMtu), absl::nullopt);
+}
+
+}  // namespace
+}  // namespace dcsctp