diff --git a/pc/channel.cc b/pc/channel.cc
index f66083f..5cff129 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -293,36 +293,17 @@
 bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
                                   SdpType type,
                                   std::string* error_desc) {
-  RTC_DCHECK_RUN_ON(signaling_thread());
+  RTC_DCHECK_RUN_ON(worker_thread());
   TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
-
-  SetContent_s(content, type);
-
-  return InvokeOnWorker<bool>(RTC_FROM_HERE, [this, content, type, error_desc] {
-    RTC_DCHECK_RUN_ON(worker_thread());
-    return SetLocalContent_w(content, type, error_desc);
-  });
+  return SetLocalContent_w(content, type, error_desc);
 }
 
 bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
                                    SdpType type,
                                    std::string* error_desc) {
-  RTC_DCHECK_RUN_ON(signaling_thread());
+  RTC_DCHECK_RUN_ON(worker_thread());
   TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent");
-
-  SetContent_s(content, type);
-
-  return InvokeOnWorker<bool>(RTC_FROM_HERE, [this, content, type, error_desc] {
-    RTC_DCHECK_RUN_ON(worker_thread());
-    return SetRemoteContent_w(content, type, error_desc);
-  });
-}
-
-void BaseChannel::SetContent_s(const MediaContentDescription* content,
-                               SdpType type) {
-  RTC_DCHECK(content);
-  if (type == SdpType::kAnswer)
-    negotiated_header_extensions_ = content->rtp_header_extensions();
+  return SetRemoteContent_w(content, type, error_desc);
 }
 
 bool BaseChannel::SetPayloadTypeDemuxingEnabled(bool enabled) {
@@ -863,11 +844,6 @@
   media_channel()->OnPacketSent(sent_packet);
 }
 
-RtpHeaderExtensions BaseChannel::GetNegotiatedRtpHeaderExtensions() const {
-  RTC_DCHECK_RUN_ON(signaling_thread());
-  return negotiated_header_extensions_;
-}
-
 VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
                            rtc::Thread* network_thread,
                            rtc::Thread* signaling_thread,
diff --git a/pc/channel.h b/pc/channel.h
index 5c2235e..e9401e4 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -288,13 +288,6 @@
   // From MessageHandler
   void OnMessage(rtc::Message* pmsg) override;
 
-  // Helper function template for invoking methods on the worker thread.
-  template <class T>
-  T InvokeOnWorker(const rtc::Location& posted_from,
-                   rtc::FunctionView<T()> functor) {
-    return worker_thread_->Invoke<T>(posted_from, functor);
-  }
-
   // Add |payload_type| to |demuxer_criteria_| if payload type demuxing is
   // enabled.
   void MaybeAddHandledPayloadType(int payload_type) RTC_RUN_ON(worker_thread());
@@ -309,15 +302,10 @@
   // Return description of media channel to facilitate logging
   std::string ToString() const;
 
-  // ChannelInterface overrides
-  RtpHeaderExtensions GetNegotiatedRtpHeaderExtensions() const override;
-
  private:
   bool ConnectToRtpTransport() RTC_RUN_ON(network_thread());
   void DisconnectFromRtpTransport() RTC_RUN_ON(network_thread());
   void SignalSentPacket_n(const rtc::SentPacket& sent_packet);
-  void SetContent_s(const MediaContentDescription* content,
-                    webrtc::SdpType type) RTC_RUN_ON(signaling_thread());
 
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;
@@ -348,6 +336,24 @@
   bool was_ever_writable_n_ RTC_GUARDED_BY(network_thread()) = false;
   bool was_ever_writable_ RTC_GUARDED_BY(worker_thread()) = false;
   const bool srtp_required_ = true;
+
+  // TODO(tommi): This field shouldn't be necessary. It's a copy of
+  // PeerConnection::GetCryptoOptions(), which is const state. It's also only
+  // used to filter header extensions when calling
+  // `rtp_transport_->UpdateRtpHeaderExtensionMap()` when the local/remote
+  // content description is updated. Since the transport is actually owned
+  // by the transport controller that also gets updated whenever the content
+  // description changes, it seems we have two paths into the transports, along
+  // with several thread hops via various classes (such as the Channel classes)
+  // that only serve as additional layers and store duplicate state. The Jsep*
+  // family of classes already apply session description updates on the network
+  // thread every time it changes.
+  // For the Channel classes, we should be able to get rid of:
+  // * crypto_options (and fewer construction parameters)_
+  // * UpdateRtpHeaderExtensionMap
+  // * GetFilteredRtpHeaderExtensions
+  // * Blocking thread hop to the network thread for every call to set
+  //   local/remote content is updated.
   const webrtc::CryptoOptions crypto_options_;
 
   // MediaChannel related members that should be accessed from the worker
@@ -382,12 +388,6 @@
   // like in Simulcast.
   // This object is not owned by the channel so it must outlive it.
   rtc::UniqueRandomIdGenerator* const ssrc_generator_;
-
-  // |negotiated_header_extensions_| is read and written to on the signaling
-  // thread from the SdpOfferAnswerHandler class (e.g.
-  // PushdownMediaDescription().
-  RtpHeaderExtensions negotiated_header_extensions_
-      RTC_GUARDED_BY(signaling_thread());
 };
 
 // VoiceChannel is a specialization that adds support for early media, DTMF,
diff --git a/pc/channel_interface.h b/pc/channel_interface.h
index fced8cc..3b71f0f 100644
--- a/pc/channel_interface.h
+++ b/pc/channel_interface.h
@@ -64,9 +64,6 @@
   //   * A DtlsSrtpTransport for DTLS-SRTP.
   virtual bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) = 0;
 
-  // Returns the last negotiated header extensions.
-  virtual RtpHeaderExtensions GetNegotiatedRtpHeaderExtensions() const = 0;
-
  protected:
   virtual ~ChannelInterface() = default;
 };
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 15c4c4c..0b7de31 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -509,10 +509,9 @@
 
 std::vector<RtpHeaderExtensionCapability>
 RtpTransceiver::HeaderExtensionsNegotiated() const {
-  if (!channel_)
-    return {};
+  RTC_DCHECK_RUN_ON(thread_);
   std::vector<RtpHeaderExtensionCapability> result;
-  for (const auto& ext : channel_->GetNegotiatedRtpHeaderExtensions()) {
+  for (const auto& ext : negotiated_header_extensions_) {
     result.emplace_back(ext.uri, ext.id, RtpTransceiverDirection::kSendRecv);
   }
   return result;
@@ -562,6 +561,15 @@
   return RTCError::OK();
 }
 
+void RtpTransceiver::OnNegotiationUpdate(
+    SdpType sdp_type,
+    const cricket::MediaContentDescription* content) {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_DCHECK(content);
+  if (sdp_type == SdpType::kAnswer)
+    negotiated_header_extensions_ = content->rtp_header_extensions();
+}
+
 void RtpTransceiver::SetPeerConnectionClosed() {
   is_pc_closed_ = true;
 }
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index 32da9af..35dea25 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -233,6 +233,16 @@
       rtc::ArrayView<const RtpHeaderExtensionCapability>
           header_extensions_to_offer) override;
 
+  // Called on the signaling thread when the local or remote content description
+  // is updated. Used to update the negotiated header extensions.
+  // TODO(tommi): The implementation of this method is currently very simple and
+  // only used for updating the negotiated headers. However, we're planning to
+  // move all the updates done on the channel from the transceiver into this
+  // method. This will happen with the ownership of the channel object being
+  // moved into the transceiver.
+  void OnNegotiationUpdate(SdpType sdp_type,
+                           const cricket::MediaContentDescription* content);
+
  private:
   void OnFirstPacketReceived();
   void StopSendingAndReceiving();
@@ -264,6 +274,13 @@
   cricket::ChannelManager* channel_manager_ = nullptr;
   std::vector<RtpCodecCapability> codec_preferences_;
   std::vector<RtpHeaderExtensionCapability> header_extensions_to_offer_;
+
+  // |negotiated_header_extensions_| is read and written to on the signaling
+  // thread from the SdpOfferAnswerHandler class (e.g.
+  // PushdownMediaDescription().
+  cricket::RtpHeaderExtensions negotiated_header_extensions_
+      RTC_GUARDED_BY(thread_);
+
   const std::function<void()> on_negotiation_needed_;
 };
 
diff --git a/pc/rtp_transceiver_unittest.cc b/pc/rtp_transceiver_unittest.cc
index 523f307..f9f4f20 100644
--- a/pc/rtp_transceiver_unittest.cc
+++ b/pc/rtp_transceiver_unittest.cc
@@ -287,9 +287,6 @@
   EXPECT_CALL(mock_channel, media_type())
       .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
   EXPECT_CALL(mock_channel, media_channel()).WillRepeatedly(Return(nullptr));
-  cricket::RtpHeaderExtensions extensions;
-  EXPECT_CALL(mock_channel, GetNegotiatedRtpHeaderExtensions)
-      .WillOnce(Return(extensions));
   transceiver_.SetChannel(&mock_channel);
   EXPECT_THAT(transceiver_.HeaderExtensionsNegotiated(), ElementsAre());
 }
@@ -306,10 +303,13 @@
   EXPECT_CALL(mock_channel, media_type())
       .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
   EXPECT_CALL(mock_channel, media_channel()).WillRepeatedly(Return(nullptr));
+
   cricket::RtpHeaderExtensions extensions = {webrtc::RtpExtension("uri1", 1),
                                              webrtc::RtpExtension("uri2", 2)};
-  EXPECT_CALL(mock_channel, GetNegotiatedRtpHeaderExtensions)
-      .WillOnce(Return(extensions));
+  cricket::AudioContentDescription description;
+  description.set_rtp_header_extensions(extensions);
+  transceiver_.OnNegotiationUpdate(SdpType::kAnswer, &description);
+
   transceiver_.SetChannel(&mock_channel);
   EXPECT_THAT(transceiver_.HeaderExtensionsNegotiated(),
               ElementsAre(RtpHeaderExtensionCapability(
@@ -320,34 +320,27 @@
 
 TEST_F(RtpTransceiverTestForHeaderExtensions,
        ReturnsNegotiatedHdrExtsSecondTime) {
-  EXPECT_CALL(*receiver_.get(), SetMediaChannel(_));
   EXPECT_CALL(*receiver_.get(), StopAndEndTrack());
-  EXPECT_CALL(*sender_.get(), SetMediaChannel(_));
   EXPECT_CALL(*sender_.get(), SetTransceiverAsStopped());
   EXPECT_CALL(*sender_.get(), Stop());
 
-  cricket::MockChannelInterface mock_channel;
-  EXPECT_CALL(mock_channel, SetFirstPacketReceivedCallback(_));
-  EXPECT_CALL(mock_channel, media_type())
-      .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
-  EXPECT_CALL(mock_channel, media_channel()).WillRepeatedly(Return(nullptr));
-
   cricket::RtpHeaderExtensions extensions = {webrtc::RtpExtension("uri1", 1),
                                              webrtc::RtpExtension("uri2", 2)};
+  cricket::AudioContentDescription description;
+  description.set_rtp_header_extensions(extensions);
+  transceiver_.OnNegotiationUpdate(SdpType::kAnswer, &description);
 
-  EXPECT_CALL(mock_channel, GetNegotiatedRtpHeaderExtensions)
-      .WillOnce(Return(extensions));
-  transceiver_.SetChannel(&mock_channel);
-  transceiver_.HeaderExtensionsNegotiated();
-  testing::Mock::VerifyAndClearExpectations(&mock_channel);
-  EXPECT_CALL(mock_channel, media_type())
-      .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
-  EXPECT_CALL(mock_channel, media_channel()).WillRepeatedly(Return(nullptr));
+  EXPECT_THAT(transceiver_.HeaderExtensionsNegotiated(),
+              ElementsAre(RtpHeaderExtensionCapability(
+                              "uri1", 1, RtpTransceiverDirection::kSendRecv),
+                          RtpHeaderExtensionCapability(
+                              "uri2", 2, RtpTransceiverDirection::kSendRecv)));
 
   extensions = {webrtc::RtpExtension("uri3", 4),
                 webrtc::RtpExtension("uri5", 6)};
-  EXPECT_CALL(mock_channel, GetNegotiatedRtpHeaderExtensions)
-      .WillOnce(Return(extensions));
+  description.set_rtp_header_extensions(extensions);
+  transceiver_.OnNegotiationUpdate(SdpType::kAnswer, &description);
+
   EXPECT_THAT(transceiver_.HeaderExtensionsNegotiated(),
               ElementsAre(RtpHeaderExtensionCapability(
                               "uri3", 4, RtpTransceiverDirection::kSendRecv),
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index e3fe002..f3f22fc 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -2499,12 +2499,7 @@
 
   // Update internal objects according to the session description's media
   // descriptions.
-  RTCError error = PushdownMediaDescription(type, source, bundle_groups_by_mid);
-  if (!error.ok()) {
-    return error;
-  }
-
-  return RTCError::OK();
+  return PushdownMediaDescription(type, source, bundle_groups_by_mid);
 }
 
 bool SdpOfferAnswerHandler::ShouldFireNegotiationNeededEvent(
@@ -4191,7 +4186,11 @@
   }
 
   // Push down the new SDP media section for each audio/video transceiver.
-  for (const auto& transceiver : transceivers()->ListInternal()) {
+  auto rtp_transceivers = transceivers()->ListInternal();
+  std::vector<
+      std::pair<cricket::ChannelInterface*, const MediaContentDescription*>>
+      channels;
+  for (const auto& transceiver : rtp_transceivers) {
     const ContentInfo* content_info =
         FindMediaSectionForTransceiver(transceiver, sdesc);
     cricket::ChannelInterface* channel = transceiver->channel();
@@ -4203,12 +4202,28 @@
     if (!content_desc) {
       continue;
     }
-    std::string error;
-    bool success = (source == cricket::CS_LOCAL)
-                       ? channel->SetLocalContent(content_desc, type, &error)
-                       : channel->SetRemoteContent(content_desc, type, &error);
-    if (!success) {
-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
+
+    transceiver->OnNegotiationUpdate(type, content_desc);
+    channels.push_back(std::make_pair(channel, content_desc));
+  }
+
+  if (!channels.empty()) {
+    RTCError error =
+        pc_->worker_thread()->Invoke<RTCError>(RTC_FROM_HERE, [&]() {
+          std::string error;
+          for (const auto& entry : channels) {
+            bool success =
+                (source == cricket::CS_LOCAL)
+                    ? entry.first->SetLocalContent(entry.second, type, &error)
+                    : entry.first->SetRemoteContent(entry.second, type, &error);
+            if (!success) {
+              LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
+            }
+          }
+          return RTCError::OK();
+        });
+    if (!error.ok()) {
+      return error;
     }
   }
 
diff --git a/pc/test/mock_channel_interface.h b/pc/test/mock_channel_interface.h
index d376e42..6faba5c 100644
--- a/pc/test/mock_channel_interface.h
+++ b/pc/test/mock_channel_interface.h
@@ -58,10 +58,6 @@
               SetRtpTransport,
               (webrtc::RtpTransportInternal*),
               (override));
-  MOCK_METHOD(RtpHeaderExtensions,
-              GetNegotiatedRtpHeaderExtensions,
-              (),
-              (const));
 };
 
 }  // namespace cricket