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diff --git a/third_party/libwebrtc/pc/peer_connection_sdp_methods.h b/third_party/libwebrtc/pc/peer_connection_sdp_methods.h
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+++ b/third_party/libwebrtc/pc/peer_connection_sdp_methods.h
@@ -0,0 +1,131 @@
+/*
+ *  Copyright 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 PC_PEER_CONNECTION_SDP_METHODS_H_
+#define PC_PEER_CONNECTION_SDP_METHODS_H_
+
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "api/peer_connection_interface.h"
+#include "pc/jsep_transport_controller.h"
+#include "pc/peer_connection_message_handler.h"
+#include "pc/sctp_data_channel.h"
+#include "pc/usage_pattern.h"
+
+namespace webrtc {
+
+class DataChannelController;
+class RtpTransmissionManager;
+class StatsCollector;
+
+// This interface defines the functions that are needed for
+// SdpOfferAnswerHandler to access PeerConnection internal state.
+class PeerConnectionSdpMethods {
+ public:
+  virtual ~PeerConnectionSdpMethods() = default;
+
+  // The SDP session ID as defined by RFC 3264.
+  virtual std::string session_id() const = 0;
+
+  // Returns true if the ICE restart flag above was set, and no ICE restart has
+  // occurred yet for this transport (by applying a local description with
+  // changed ufrag/password). If the transport has been deleted as a result of
+  // bundling, returns false.
+  virtual bool NeedsIceRestart(const std::string& content_name) const = 0;
+
+  virtual absl::optional<std::string> sctp_mid() const = 0;
+
+  // Functions below this comment are known to only be accessed
+  // from SdpOfferAnswerHandler.
+  // Return a pointer to the active configuration.
+  virtual const PeerConnectionInterface::RTCConfiguration* configuration()
+      const = 0;
+
+  // Report the UMA metric SdpFormatReceived for the given remote description.
+  virtual void ReportSdpFormatReceived(
+      const SessionDescriptionInterface& remote_description) = 0;
+
+  // Report the UMA metric BundleUsage for the given remote description.
+  virtual void ReportSdpBundleUsage(
+      const SessionDescriptionInterface& remote_description) = 0;
+
+  virtual PeerConnectionMessageHandler* message_handler() = 0;
+  virtual RtpTransmissionManager* rtp_manager() = 0;
+  virtual const RtpTransmissionManager* rtp_manager() const = 0;
+  virtual bool dtls_enabled() const = 0;
+  virtual const PeerConnectionFactoryInterface::Options* options() const = 0;
+
+  // Returns the CryptoOptions for this PeerConnection. This will always
+  // return the RTCConfiguration.crypto_options if set and will only default
+  // back to the PeerConnectionFactory settings if nothing was set.
+  virtual CryptoOptions GetCryptoOptions() = 0;
+  virtual JsepTransportController* transport_controller_s() = 0;
+  virtual JsepTransportController* transport_controller_n() = 0;
+  virtual DataChannelController* data_channel_controller() = 0;
+  virtual cricket::PortAllocator* port_allocator() = 0;
+  virtual StatsCollector* stats() = 0;
+  // Returns the observer. Will crash on CHECK if the observer is removed.
+  virtual PeerConnectionObserver* Observer() const = 0;
+  virtual bool GetSctpSslRole(rtc::SSLRole* role) = 0;
+  virtual PeerConnectionInterface::IceConnectionState
+  ice_connection_state_internal() = 0;
+  virtual void SetIceConnectionState(
+      PeerConnectionInterface::IceConnectionState new_state) = 0;
+  virtual void NoteUsageEvent(UsageEvent event) = 0;
+  virtual bool IsClosed() const = 0;
+  // Returns true if the PeerConnection is configured to use Unified Plan
+  // semantics for creating offers/answers and setting local/remote
+  // descriptions. If this is true the RtpTransceiver API will also be available
+  // to the user. If this is false, Plan B semantics are assumed.
+  // TODO(bugs.webrtc.org/8530): Flip the default to be Unified Plan once
+  // sufficient time has passed.
+  virtual bool IsUnifiedPlan() const = 0;
+  virtual bool ValidateBundleSettings(
+      const cricket::SessionDescription* desc,
+      const std::map<std::string, const cricket::ContentGroup*>&
+          bundle_groups_by_mid) = 0;
+
+  virtual absl::optional<std::string> GetDataMid() const = 0;
+  // Internal implementation for AddTransceiver family of methods. If
+  // `fire_callback` is set, fires OnRenegotiationNeeded callback if successful.
+  virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
+  AddTransceiver(cricket::MediaType media_type,
+                 rtc::scoped_refptr<MediaStreamTrackInterface> track,
+                 const RtpTransceiverInit& init,
+                 bool fire_callback = true) = 0;
+  // Asynchronously calls SctpTransport::Start() on the network thread for
+  // `sctp_mid()` if set. Called as part of setting the local description.
+  virtual void StartSctpTransport(int local_port,
+                                  int remote_port,
+                                  int max_message_size) = 0;
+
+  // Asynchronously adds a remote candidate on the network thread.
+  virtual void AddRemoteCandidate(const std::string& mid,
+                                  const cricket::Candidate& candidate) = 0;
+
+  virtual Call* call_ptr() = 0;
+  // Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
+  // this session.
+  virtual bool SrtpRequired() const = 0;
+  virtual bool SetupDataChannelTransport_n(const std::string& mid) = 0;
+  virtual void TeardownDataChannelTransport_n() = 0;
+  virtual void SetSctpDataMid(const std::string& mid) = 0;
+  virtual void ResetSctpDataMid() = 0;
+
+  virtual const FieldTrialsView& trials() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // PC_PEER_CONNECTION_SDP_METHODS_H_