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diff --git a/third_party/libwebrtc/api/packet_socket_factory.h b/third_party/libwebrtc/api/packet_socket_factory.h
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+/*
+ *  Copyright 2019 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 API_PACKET_SOCKET_FACTORY_H_
+#define API_PACKET_SOCKET_FACTORY_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/async_dns_resolver.h"
+#include "api/wrapping_async_dns_resolver.h"
+#include "rtc_base/async_packet_socket.h"
+#include "rtc_base/proxy_info.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace rtc {
+
+class SSLCertificateVerifier;
+class AsyncResolverInterface;
+
+struct PacketSocketTcpOptions {
+  PacketSocketTcpOptions() = default;
+  ~PacketSocketTcpOptions() = default;
+
+  int opts = 0;
+  std::vector<std::string> tls_alpn_protocols;
+  std::vector<std::string> tls_elliptic_curves;
+  // An optional custom SSL certificate verifier that an API user can provide to
+  // inject their own certificate verification logic (not available to users
+  // outside of the WebRTC repo).
+  SSLCertificateVerifier* tls_cert_verifier = nullptr;
+};
+
+class RTC_EXPORT PacketSocketFactory {
+ public:
+  enum Options {
+    OPT_STUN = 0x04,
+
+    // The TLS options below are mutually exclusive.
+    OPT_TLS = 0x02,           // Real and secure TLS.
+    OPT_TLS_FAKE = 0x01,      // Fake TLS with a dummy SSL handshake.
+    OPT_TLS_INSECURE = 0x08,  // Insecure TLS without certificate validation.
+
+    // Deprecated, use OPT_TLS_FAKE.
+    OPT_SSLTCP = OPT_TLS_FAKE,
+  };
+
+  PacketSocketFactory() = default;
+  virtual ~PacketSocketFactory() = default;
+
+  virtual AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address,
+                                             uint16_t min_port,
+                                             uint16_t max_port) = 0;
+  virtual AsyncListenSocket* CreateServerTcpSocket(
+      const SocketAddress& local_address,
+      uint16_t min_port,
+      uint16_t max_port,
+      int opts) = 0;
+
+  virtual AsyncPacketSocket* CreateClientTcpSocket(
+      const SocketAddress& local_address,
+      const SocketAddress& remote_address,
+      const ProxyInfo& proxy_info,
+      const std::string& user_agent,
+      const PacketSocketTcpOptions& tcp_options) = 0;
+
+  // The AsyncResolverInterface is deprecated; users are encouraged
+  // to switch to the AsyncDnsResolverInterface.
+  // TODO(bugs.webrtc.org/12598): Remove once all downstream users
+  // are converted.
+  virtual AsyncResolverInterface* CreateAsyncResolver() {
+    // Default implementation, so that downstream users can remove this
+    // immediately after changing to CreateAsyncDnsResolver
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface>
+  CreateAsyncDnsResolver() {
+    // Default implementation, to aid in transition to AsyncDnsResolverInterface
+    return std::make_unique<webrtc::WrappingAsyncDnsResolver>(
+        CreateAsyncResolver());
+  }
+
+ private:
+  PacketSocketFactory(const PacketSocketFactory&) = delete;
+  PacketSocketFactory& operator=(const PacketSocketFactory&) = delete;
+};
+
+}  // namespace rtc
+
+#endif  // API_PACKET_SOCKET_FACTORY_H_