diff --git a/call/BUILD.gn b/call/BUILD.gn
index 0c4255a593..5b21ddaaf9 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -801,6 +801,17 @@ if (rtc_include_tests) {
     ]
   }
 
+  rtc_source_set("fake_payload_type_suggester") {
+    testonly = true
+    sources = [ "fake_payload_type_suggester.h" ]
+    deps = [
+      ":payload_type",
+      ":payload_type_picker",
+      "../api:rtc_error",
+      "../media:codec",
+    ]
+  }
+
   rtc_library("fake_network_pipe_unittests") {
     testonly = true
 
diff --git a/call/DEPS b/call/DEPS
index e74b22f677..98a8a4b68d 100644
--- a/call/DEPS
+++ b/call/DEPS
@@ -70,5 +70,8 @@ specific_include_rules = {
   ],
   "call\.cc": [
     "+media/base/codec.h",
+  ],
+  "fake_payload_type_suggester": [
+    "+media/base/codec.h",
   ]
 }
diff --git a/call/fake_payload_type_suggester.h b/call/fake_payload_type_suggester.h
new file mode 100644
index 0000000000..c2aa6ebd01
--- /dev/null
+++ b/call/fake_payload_type_suggester.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2015 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 CALL_FAKE_PAYLOAD_TYPE_SUGGESTER_H_
+#define CALL_FAKE_PAYLOAD_TYPE_SUGGESTER_H_
+
+#include <string>
+
+#include "api/rtc_error.h"
+#include "call/payload_type.h"
+#include "call/payload_type_picker.h"
+#include "media/base/codec.h"
+
+namespace webrtc {
+// Fake payload type suggester, for use in tests.
+// It uses a real PayloadTypePicker in order to do consistent PT
+// assignment.
+class FakePayloadTypeSuggester : public webrtc::PayloadTypeSuggester {
+ public:
+  webrtc::RTCErrorOr<webrtc::PayloadType> SuggestPayloadType(
+      const std::string& mid,
+      cricket::Codec codec) override {
+    // Ignores mid argument.
+    return pt_picker_.SuggestMapping(codec, nullptr);
+  }
+  webrtc::RTCError AddLocalMapping(const std::string& mid,
+                                   webrtc::PayloadType payload_type,
+                                   const cricket::Codec& codec) override {
+    return webrtc::RTCError::OK();
+  }
+
+ private:
+  webrtc::PayloadTypePicker pt_picker_;
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_FAKE_PAYLOAD_TYPE_SUGGESTER_H_
diff --git a/call/payload_type.h b/call/payload_type.h
index c0520e8f4c..635533327c 100644
--- a/call/payload_type.h
+++ b/call/payload_type.h
@@ -26,11 +26,24 @@ class PayloadType : public StrongAlias<class PayloadTypeTag, uint8_t> {
   // removed once calling code is upgraded.
   PayloadType(uint8_t pt) { value_ = pt; }                // NOLINT: explicit
   constexpr operator uint8_t() const& { return value_; }  // NOLINT: Explicit
+  static bool IsValid(PayloadType id, bool rtcp_mux) {
+    if (rtcp_mux && (id > 63 && id < 96)) {
+      return false;
+    }
+    return id >= 0 && id <= 127;
+  }
 };
 
+// Does not compile either within or after class
+// static const PayloadType kFirstDynamicPayloadTypeUpperRange{96};
+//  static const PayloadType kLastDynamicPayloadTypeUpperRange{127};
+//  static const PayloadType kFirstDynamicPayloadTypeLowerRange{35};
+//  static const PayloadType kLastDynamicPayloadTypeLowerRange{63};
+
 class PayloadTypeSuggester {
  public:
   virtual ~PayloadTypeSuggester() = default;
+
   // Suggest a payload type for a given codec on a given media section.
   // Media section is indicated by MID.
   // The function will either return a PT already in use on the connection
diff --git a/call/payload_type_picker.cc b/call/payload_type_picker.cc
index 12854cfec3..915a3b8b22 100644
--- a/call/payload_type_picker.cc
+++ b/call/payload_type_picker.cc
@@ -140,10 +140,11 @@ RTCErrorOr<PayloadType> PayloadTypePicker::SuggestMapping(
   // The first matching entry is returned, unless excluder
   // maps it to something different.
   for (auto entry : entries_) {
-    if (MatchesWithCodecRules(entry.codec(), codec)) {
+    if (MatchesWithReferenceAttributes(entry.codec(), codec)) {
       if (excluder) {
         auto result = excluder->LookupCodec(entry.payload_type());
-        if (result.ok() && !MatchesWithCodecRules(result.value(), codec)) {
+        if (result.ok() &&
+            !MatchesWithReferenceAttributes(result.value(), codec)) {
           continue;
         }
       }
@@ -164,7 +165,7 @@ RTCError PayloadTypePicker::AddMapping(PayloadType payload_type,
   // Multiple mappings for the same codec and the same PT are legal;
   for (auto entry : entries_) {
     if (payload_type == entry.payload_type() &&
-        MatchesWithCodecRules(codec, entry.codec())) {
+        MatchesWithReferenceAttributes(codec, entry.codec())) {
       return RTCError::OK();
     }
   }
@@ -177,7 +178,7 @@ RTCError PayloadTypeRecorder::AddMapping(PayloadType payload_type,
                                          cricket::Codec codec) {
   auto existing_codec_it = payload_type_to_codec_.find(payload_type);
   if (existing_codec_it != payload_type_to_codec_.end() &&
-      !MatchesWithCodecRules(codec, existing_codec_it->second)) {
+      !MatchesWithReferenceAttributes(codec, existing_codec_it->second)) {
     if (absl::EqualsIgnoreCase(codec.name, existing_codec_it->second.name)) {
       // The difference is in clock rate, channels or FMTP parameters.
       RTC_LOG(LS_INFO) << "Warning: Attempt to change a codec's parameters";
@@ -185,8 +186,8 @@ RTCError PayloadTypeRecorder::AddMapping(PayloadType payload_type,
       // This is done in production today, so we can't return an error.
     } else {
       RTC_LOG(LS_WARNING) << "Warning: You attempted to redefine a codec from "
-                          << existing_codec_it->second.ToString() << " to "
-                          << " new codec " << codec.ToString();
+                          << existing_codec_it->second << " to "
+                          << " new codec " << codec;
       // This is a spec violation.
       // TODO: https://issues.webrtc.org/41480892 - return an error.
     }
@@ -211,7 +212,7 @@ RTCErrorOr<PayloadType> PayloadTypeRecorder::LookupPayloadType(
   auto result =
       std::find_if(payload_type_to_codec_.begin(), payload_type_to_codec_.end(),
                    [codec](const auto& iter) {
-                     return MatchesWithCodecRules(iter.second, codec);
+                     return MatchesWithReferenceAttributes(iter.second, codec);
                    });
   if (result == payload_type_to_codec_.end()) {
     return RTCError(RTCErrorType::INVALID_PARAMETER,
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 5126519d81..994dddb026 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -818,6 +818,7 @@ if (rtc_include_tests) {
       "../api/video:video_rtp_headers",
       "../api/video_codecs:video_codecs_api",
       "../call:call_interfaces",
+      "../call:fake_payload_type_suggester",
       "../call:mock_rtp_interfaces",
       "../call:payload_type",
       "../call:payload_type_picker",
diff --git a/media/DEPS b/media/DEPS
index 414e08f206..85b9c7ef3e 100644
--- a/media/DEPS
+++ b/media/DEPS
@@ -25,6 +25,9 @@ specific_include_rules = {
   ".*webrtc_video_engine\.h": [
     "+video/config",
   ],
+  ".*webrtc_video_engine\.cc": [
+    "+video/config",
+  ],
   ".*media_channel\.h": [
     "+video/config",
   ],
diff --git a/media/base/codec_comparators.cc b/media/base/codec_comparators.cc
index 207f093a76..52dd23221e 100644
--- a/media/base/codec_comparators.cc
+++ b/media/base/codec_comparators.cc
@@ -278,6 +278,11 @@ bool MatchesWithCodecRules(const Codec& left_codec, const Codec& right_codec) {
   return matches_id && matches_type_specific();
 }
 
+bool MatchesWithReferenceAttributes(const Codec& codec1, const Codec& codec2) {
+  return MatchesWithReferenceAttributesAndComparator(
+      codec1, codec2, [](int a, int b) { return a == b; });
+}
+
 // Finds a codec in `codecs2` that matches `codec_to_match`, which is
 // a member of `codecs1`. If `codec_to_match` is an RED or RTX codec, both
 // the codecs themselves and their associated codecs must match.
diff --git a/media/base/codec_comparators.h b/media/base/codec_comparators.h
index 2c96d6a3d6..a8b425566b 100644
--- a/media/base/codec_comparators.h
+++ b/media/base/codec_comparators.h
@@ -18,14 +18,15 @@
 
 namespace webrtc {
 
-// Comparison used in the PayloadTypePicker
-bool MatchesForSdp(const cricket::Codec& codec_1,
-                   const cricket::Codec& codec_2);
-
 // Comparison used for the Codec::Matches function
 bool MatchesWithCodecRules(const cricket::Codec& left_codec,
                            const cricket::Codec& codec);
 
+// Comparison that also checks on codecs referenced by PT in the
+// fmtp line, as used with RED and RTX "codecs".
+bool MatchesWithReferenceAttributes(const cricket::Codec& left_codec,
+                                    const cricket::Codec& right_codec);
+
 // Finds a codec in `codecs2` that matches `codec_to_match`, which is
 // a member of `codecs1`. If `codec_to_match` is an RED or RTX codec, both
 // the codecs themselves and their associated codecs must match.
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index c4e5f615a2..8f0b773902 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -37,7 +37,6 @@
 #include "api/environment/environment.h"
 #include "api/frame_transformer_interface.h"
 #include "api/media_types.h"
-#include "api/rtc_error.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_sender_interface.h"
@@ -54,15 +53,14 @@
 #include "call/audio_receive_stream.h"
 #include "call/audio_send_stream.h"
 #include "call/call.h"
+#include "call/fake_payload_type_suggester.h"
 #include "call/flexfec_receive_stream.h"
 #include "call/packet_receiver.h"
 #include "call/payload_type.h"
-#include "call/payload_type_picker.h"
 #include "call/rtp_transport_controller_send_interface.h"
 #include "call/test/mock_rtp_transport_controller_send.h"
 #include "call/video_receive_stream.h"
 #include "call/video_send_stream.h"
-#include "media/base/codec.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "rtc_base/buffer.h"
@@ -395,26 +393,6 @@ class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
   webrtc::FlexfecReceiveStream::Config config_;
 };
 
-// Fake payload type suggester.
-// This is injected into FakeCall at initialization.
-class FakePayloadTypeSuggester : public webrtc::PayloadTypeSuggester {
- public:
-  webrtc::RTCErrorOr<webrtc::PayloadType> SuggestPayloadType(
-      const std::string& mid,
-      cricket::Codec codec) override {
-    // Ignores mid argument.
-    return pt_picker_.SuggestMapping(codec, nullptr);
-  }
-  webrtc::RTCError AddLocalMapping(const std::string& mid,
-                                   webrtc::PayloadType payload_type,
-                                   const cricket::Codec& codec) override {
-    return webrtc::RTCError::OK();
-  }
-
- private:
-  webrtc::PayloadTypePicker pt_picker_;
-};
-
 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
  public:
   explicit FakeCall(const webrtc::Environment& env);
@@ -558,7 +536,7 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
   int num_created_send_streams_;
   int num_created_receive_streams_;
 
-  FakePayloadTypeSuggester pt_suggester_;
+  webrtc::FakePayloadTypeSuggester pt_suggester_;
 };
 
 }  // namespace cricket
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index bbcaf1b52d..4fee59c6df 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -14,28 +14,46 @@
 
 #include <algorithm>
 #include <cstdint>
+#include <functional>
 #include <initializer_list>
+#include <map>
+#include <memory>
 #include <optional>
 #include <set>
 #include <string>
-#include <type_traits>
 #include <utility>
+#include <vector>
 
 #include "absl/algorithm/container.h"
-#include "absl/container/inlined_vector.h"
 #include "absl/functional/bind_front.h"
 #include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/field_trials_view.h"
+#include "api/frame_transformer_interface.h"
 #include "api/make_ref_counted.h"
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
 #include "api/priority.h"
 #include "api/rtc_error.h"
+#include "api/rtp_headers.h"
 #include "api/rtp_parameters.h"
+#include "api/rtp_sender_interface.h"
 #include "api/rtp_transceiver_direction.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/transport/rtp/rtp_source.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
-#include "api/video/resolution.h"
+#include "api/video/recordable_encoded_frame.h"
+#include "api/video/video_bitrate_allocator_factory.h"
 #include "api/video/video_codec_type.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_source_interface.h"
 #include "api/video_codecs/scalability_mode.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_codec.h"
@@ -43,23 +61,29 @@
 #include "api/video_codecs/video_encoder.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "call/call.h"
+#include "call/flexfec_receive_stream.h"
 #include "call/packet_receiver.h"
 #include "call/receive_stream.h"
 #include "call/rtp_config.h"
 #include "call/rtp_transport_controller_send_interface.h"
+#include "call/video_receive_stream.h"
 #include "call/video_send_stream.h"
 #include "common_video/frame_counts.h"
-#include "common_video/include/quality_limitation_reason.h"
 #include "media/base/codec.h"
 #include "media/base/media_channel.h"
+#include "media/base/media_channel_impl.h"
+#include "media/base/media_config.h"
 #include "media/base/media_constants.h"
+#include "media/base/media_engine.h"
 #include "media/base/rid_description.h"
 #include "media/base/rtp_utils.h"
+#include "media/base/stream_params.h"
 #include "media/engine/webrtc_media_engine.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
-#include "modules/rtp_rtcp/include/report_block_data.h"
 #include "modules/rtp_rtcp/include/rtcp_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "modules/video_coding/svc/scalability_mode_util.h"
 #include "rtc_base/checks.h"
@@ -68,8 +92,10 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/socket.h"
 #include "rtc_base/strings/string_builder.h"
+#include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/time_utils.h"
 #include "rtc_base/trace_event.h"
+#include "video/config/video_encoder_config.h"
 
 namespace cricket {
 
@@ -583,8 +609,7 @@ std::vector<VideoCodecSettings> MapCodecs(const std::vector<Codec>& codecs) {
     const int payload_type = in_codec.id;
 
     if (payload_codec_type.find(payload_type) != payload_codec_type.end()) {
-      RTC_LOG(LS_ERROR) << "Payload type already registered: "
-                        << in_codec.ToString();
+      RTC_LOG(LS_ERROR) << "Payload type already registered: " << in_codec;
       return {};
     }
     payload_codec_type[payload_type] = in_codec.GetResiliencyType();
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index d1435115d5..894f8b7992 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -2025,6 +2025,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/video:builtin_video_bitrate_allocator_factory",
       "../api/video:recordable_encoded_frame",
       "../api/video/test:mock_recordable_encoded_frame",
+      "../call:fake_payload_type_suggester",
       "../call:payload_type_picker",
       "../call:rtp_interfaces",
       "../call:rtp_receiver",
@@ -2565,6 +2566,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:ice_transport_interface",
       "../api:libjingle_logging_api",
       "../api:libjingle_peerconnection_api",
+      "../api:make_ref_counted",
       "../api:media_stream_interface",
       "../api:mock_async_dns_resolver",
       "../api:mock_rtp",
@@ -2575,6 +2577,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:rtp_sender_interface",
       "../api:rtp_transceiver_direction",
       "../api:scoped_refptr",
+      "../api:sequence_checker",
       "../api/audio:audio_device",
       "../api/audio:audio_mixer_api",
       "../api/audio:audio_processing",
@@ -2582,6 +2585,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/crypto:frame_decryptor_interface",
       "../api/crypto:frame_encryptor_interface",
       "../api/crypto:options",
+      "../api/metronome",
       "../api/rtc_event_log",
       "../api/rtc_event_log:rtc_event_log_factory",
       "../api/task_queue",
@@ -2628,6 +2632,8 @@ if (rtc_include_tests && !build_with_chromium) {
       "../rtc_base:rtc_json",
       "../rtc_base:safe_conversions",
       "../rtc_base:socket_address",
+      "../rtc_base:socket_factory",
+      "../rtc_base:socket_server",
       "../rtc_base:ssl",
       "../rtc_base:ssl_adapter",
       "../rtc_base:task_queue_for_test",
@@ -2645,6 +2651,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../test/pc/sctp:fake_sctp_transport",
       "../test/time_controller",
       "//third_party/abseil-cpp/absl/algorithm:container",
+      "//third_party/abseil-cpp/absl/functional:any_invocable",
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/abseil-cpp/absl/strings:string_view",
     ]
diff --git a/pc/media_session.cc b/pc/media_session.cc
index 3d6eccda06..1c1ac3286a 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -495,7 +495,6 @@ webrtc::RTCError AssignCodecIdsAndLinkRed(
           char buffer[100];
           rtc::SimpleStringBuilder param(buffer);
           param << opus_codec << "/" << opus_codec;
-          RTC_LOG(LS_ERROR) << "DEBUG: Setting RED param to " << param.str();
           codec.SetParam(kCodecParamNotInNameValueFormat, param.str());
         }
       }
@@ -641,9 +640,10 @@ const Codec* GetAssociatedCodecForRed(const std::vector<Codec>& codec_list,
 // Adds all codecs from `reference_codecs` to `offered_codecs` that don't
 // already exist in `offered_codecs` and ensure the payload types don't
 // collide.
-void MergeCodecs(const std::vector<Codec>& reference_codecs,
-                 std::vector<Codec>* offered_codecs,
-                 UsedPayloadTypes* used_pltypes) {
+webrtc::RTCError MergeCodecs(const std::vector<Codec>& reference_codecs,
+                             std::vector<Codec>* offered_codecs,
+                             std::string mid,
+                             webrtc::PayloadTypeSuggester& pt_suggester) {
   // Add all new codecs that are not RTX/RED codecs.
   // The two-pass splitting of the loops means preferring payload types
   // of actual codecs with respect to collisions.
@@ -653,7 +653,11 @@ void MergeCodecs(const std::vector<Codec>& reference_codecs,
         !webrtc::FindMatchingCodec(reference_codecs, *offered_codecs,
                                    reference_codec)) {
       Codec codec = reference_codec;
-      used_pltypes->FindAndSetIdUsed(&codec);
+      auto id_or_error = pt_suggester.SuggestPayloadType(mid, codec);
+      if (!id_or_error.ok()) {
+        return id_or_error.MoveError();
+      }
+      codec.id = id_or_error.value();
       offered_codecs->push_back(codec);
     }
   }
@@ -681,15 +685,35 @@ void MergeCodecs(const std::vector<Codec>& reference_codecs,
 
       rtx_codec.params[kCodecParamAssociatedPayloadType] =
           rtc::ToString(matching_codec->id);
-      used_pltypes->FindAndSetIdUsed(&rtx_codec);
+      auto id_or_error = pt_suggester.SuggestPayloadType(mid, rtx_codec);
+      if (!id_or_error.ok()) {
+        return id_or_error.MoveError();
+      }
+      rtx_codec.id = id_or_error.value();
       offered_codecs->push_back(rtx_codec);
     } else if (reference_codec.GetResiliencyType() ==
                    Codec::ResiliencyType::kRed &&
                !webrtc::FindMatchingCodec(reference_codecs, *offered_codecs,
                                           reference_codec)) {
       Codec red_codec = reference_codec;
-      const Codec* associated_codec =
-          GetAssociatedCodecForRed(reference_codecs, red_codec);
+      const Codec* associated_codec = nullptr;
+      // Special case: For voice RED, if parameter is not set, look for
+      // OPUS as the matching codec.
+      // This is because we add the RED codec in audio engine init, but
+      // can't set the parameter until PTs are assigned.
+      if (red_codec.type == Codec::Type::kAudio &&
+          red_codec.params.find(kCodecParamNotInNameValueFormat) ==
+              red_codec.params.end()) {
+        for (const Codec& codec : reference_codecs) {
+          if (absl::EqualsIgnoreCase(codec.name, kOpusCodecName)) {
+            associated_codec = &codec;
+            break;
+          }
+        }
+      } else {
+        associated_codec =
+            GetAssociatedCodecForRed(reference_codecs, red_codec);
+      }
       if (associated_codec) {
         std::optional<Codec> matching_codec = webrtc::FindMatchingCodec(
             reference_codecs, *offered_codecs, *associated_codec);
@@ -703,10 +727,15 @@ void MergeCodecs(const std::vector<Codec>& reference_codecs,
             rtc::ToString(matching_codec->id) + "/" +
             rtc::ToString(matching_codec->id);
       }
-      used_pltypes->FindAndSetIdUsed(&red_codec);
+      auto id_or_error = pt_suggester.SuggestPayloadType(mid, red_codec);
+      if (!id_or_error.ok()) {
+        return id_or_error.MoveError();
+      }
+      red_codec.id = id_or_error.value();
       offered_codecs->push_back(red_codec);
     }
   }
+  return webrtc::RTCError::OK();
 }
 
 // `codecs` is a full list of codecs with correct payload type mappings, which
@@ -795,20 +824,28 @@ std::vector<Codec> MatchCodecPreference(
 }
 
 // Compute the union of `codecs1` and `codecs2`.
-std::vector<Codec> ComputeCodecsUnion(const std::vector<Codec>& codecs1,
-                                      const std::vector<Codec>& codecs2) {
+webrtc::RTCErrorOr<std::vector<Codec>> ComputeCodecsUnion(
+    const std::vector<Codec>& codecs1,
+    const std::vector<Codec>& codecs2,
+    std::string mid,
+    webrtc::PayloadTypeSuggester& pt_suggester) {
   std::vector<Codec> all_codecs;
-  UsedPayloadTypes used_payload_types;
   for (const Codec& codec : codecs1) {
     Codec codec_mutable = codec;
-    used_payload_types.FindAndSetIdUsed(&codec_mutable);
+    auto id_or_error = pt_suggester.SuggestPayloadType(mid, codec);
+    if (!id_or_error.ok()) {
+      return id_or_error.MoveError();
+    }
+    codec_mutable.id = id_or_error.value();
     all_codecs.push_back(codec_mutable);
   }
 
   // Use MergeCodecs to merge the second half of our list as it already checks
   // and fixes problems with duplicate payload types.
-  MergeCodecs(codecs2, &all_codecs, &used_payload_types);
-
+  webrtc::RTCError error = MergeCodecs(codecs2, &all_codecs, mid, pt_suggester);
+  if (!error.ok()) {
+    return error;
+  }
   return all_codecs;
 }
 
@@ -1147,7 +1184,8 @@ webrtc::RTCErrorOr<Codecs> GetNegotiatedCodecsForAnswer(
     const MediaSessionOptions& session_options,
     const ContentInfo* current_content,
     const std::vector<Codec>& codecs,
-    const std::vector<Codec>& supported_codecs) {
+    const std::vector<Codec>& supported_codecs,
+    webrtc::PayloadTypeSuggester& pt_suggester) {
   std::vector<Codec> filtered_codecs;
 
   if (!media_description_options.codec_preferences.empty()) {
@@ -1187,7 +1225,19 @@ webrtc::RTCErrorOr<Codecs> GetNegotiatedCodecsForAnswer(
 
     // Use ComputeCodecsUnion to avoid having duplicate payload IDs.
     // This is a no-op for audio until RTX is added.
-    filtered_codecs = ComputeCodecsUnion(filtered_codecs, other_codecs);
+    // TODO(hta): figure out why current_content is not always there.
+    std::string mid;
+    if (current_content) {
+      mid = current_content->name;
+    } else {
+      mid = "";
+    }
+    auto codecs_or_error =
+        ComputeCodecsUnion(filtered_codecs, other_codecs, mid, pt_suggester);
+    if (!codecs_or_error.ok()) {
+      return codecs_or_error.MoveError();
+    }
+    filtered_codecs = codecs_or_error.MoveValue();
   }
 
   if (media_description_options.type == MEDIA_TYPE_AUDIO &&
@@ -1264,6 +1314,7 @@ MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
           transport_desc_factory_->trials().IsEnabled(
               "WebRTC-PayloadTypesInTransport")) {
   RTC_CHECK(transport_desc_factory_);
+  RTC_CHECK(pt_suggester_);
   if (media_engine) {
     audio_send_codecs_ = media_engine->voice().send_codecs();
     audio_recv_codecs_ = media_engine->voice().recv_codecs();
@@ -1726,20 +1777,29 @@ const Codecs& MediaSessionDescriptionFactory::GetVideoCodecsForAnswer(
   RTC_CHECK_NOTREACHED();
 }
 
-void MergeCodecsFromDescription(
+webrtc::RTCError MergeCodecsFromDescription(
     const std::vector<const ContentInfo*>& current_active_contents,
     Codecs* audio_codecs,
     Codecs* video_codecs,
-    UsedPayloadTypes* used_pltypes) {
+    webrtc::PayloadTypeSuggester& pt_suggester) {
   for (const ContentInfo* content : current_active_contents) {
     if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) {
-      MergeCodecs(content->media_description()->codecs(), audio_codecs,
-                  used_pltypes);
+      webrtc::RTCError error =
+          MergeCodecs(content->media_description()->codecs(), audio_codecs,
+                      content->name, pt_suggester);
+      if (!error.ok()) {
+        return error;
+      }
     } else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
-      MergeCodecs(content->media_description()->codecs(), video_codecs,
-                  used_pltypes);
+      webrtc::RTCError error =
+          MergeCodecs(content->media_description()->codecs(), video_codecs,
+                      content->name, pt_suggester);
+      if (!error.ok()) {
+        return error;
+      }
     }
   }
+  return webrtc::RTCError::OK();
 }
 
 // Getting codecs for an offer involves these steps:
@@ -1755,13 +1815,15 @@ void MediaSessionDescriptionFactory::GetCodecsForOffer(
   // First - get all codecs from the current description if the media type
   // is used. Add them to `used_pltypes` so the payload type is not reused if a
   // new media type is added.
-  UsedPayloadTypes used_pltypes;
-  MergeCodecsFromDescription(current_active_contents, audio_codecs,
-                             video_codecs, &used_pltypes);
+  webrtc::RTCError error = MergeCodecsFromDescription(
+      current_active_contents, audio_codecs, video_codecs, *pt_suggester_);
+  RTC_CHECK(error.ok());
 
   // Add our codecs that are not in the current description.
-  MergeCodecs(all_audio_codecs_, audio_codecs, &used_pltypes);
-  MergeCodecs(all_video_codecs_, video_codecs, &used_pltypes);
+  error = MergeCodecs(all_audio_codecs_, audio_codecs, "", *pt_suggester_);
+  RTC_CHECK(error.ok());
+  error = MergeCodecs(all_video_codecs_, video_codecs, "", *pt_suggester_);
+  RTC_CHECK(error.ok());
 }
 
 // Getting codecs for an answer involves these steps:
@@ -1779,9 +1841,9 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
   // First - get all codecs from the current description if the media type
   // is used. Add them to `used_pltypes` so the payload type is not reused if a
   // new media type is added.
-  UsedPayloadTypes used_pltypes;
-  MergeCodecsFromDescription(current_active_contents, audio_codecs,
-                             video_codecs, &used_pltypes);
+  webrtc::RTCError error = MergeCodecsFromDescription(
+      current_active_contents, audio_codecs, video_codecs, *pt_suggester_);
+  RTC_CHECK(error.ok());
 
   // Second - filter out codecs that we don't support at all and should ignore.
   Codecs filtered_offered_audio_codecs;
@@ -1814,8 +1876,12 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
 
   // Add codecs that are not in the current description but were in
   // `remote_offer`.
-  MergeCodecs(filtered_offered_audio_codecs, audio_codecs, &used_pltypes);
-  MergeCodecs(filtered_offered_video_codecs, video_codecs, &used_pltypes);
+  error = MergeCodecs(filtered_offered_audio_codecs, audio_codecs, "",
+                      *pt_suggester_);
+  RTC_CHECK(error.ok());
+  error = MergeCodecs(filtered_offered_video_codecs, video_codecs, "",
+                      *pt_suggester_);
+  RTC_CHECK(error.ok());
 }
 
 MediaSessionDescriptionFactory::AudioVideoRtpHeaderExtensions
@@ -2123,7 +2189,8 @@ RTCError MediaSessionDescriptionFactory::AddRtpContentForAnswer(
             : GetVideoCodecsForAnswer(offer_rtd, answer_rtd);
     webrtc::RTCErrorOr<std::vector<Codec>> error_or_filtered_codecs =
         GetNegotiatedCodecsForAnswer(media_description_options, session_options,
-                                     current_content, codecs, supported_codecs);
+                                     current_content, codecs, supported_codecs,
+                                     *pt_suggester_);
     if (!error_or_filtered_codecs.ok()) {
       return error_or_filtered_codecs.MoveError();
     }
@@ -2345,9 +2412,10 @@ void MediaSessionDescriptionFactory::ComputeVideoCodecsIntersectionAndUnion() {
   video_sendrecv_codecs_.clear();
 
   // Use ComputeCodecsUnion to avoid having duplicate payload IDs
-  all_video_codecs_ =
-      ComputeCodecsUnion(video_recv_codecs_, video_send_codecs_);
-
+  auto error_or_codecs = ComputeCodecsUnion(
+      video_recv_codecs_, video_send_codecs_, "", *pt_suggester_);
+  RTC_CHECK(error_or_codecs.ok());
+  all_video_codecs_ = error_or_codecs.MoveValue();
   // Use NegotiateCodecs to merge our codec lists, since the operation is
   // essentially the same. Put send_codecs as the offered_codecs, which is the
   // order we'd like to follow. The reasoning is that encoding is usually more
diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc
index c17ab75570..71172ccd54 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -23,11 +23,13 @@
 #include "absl/algorithm/container.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
+#include "api/array_view.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/candidate.h"
 #include "api/media_types.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
+#include "call/fake_payload_type_suggester.h"
 #include "media/base/codec.h"
 #include "media/base/media_constants.h"
 #include "media/base/rid_description.h"
@@ -117,6 +119,26 @@ const Codec kVideoCodecs2[] = {CreateVideoCodec(126, "H264"),
 
 const Codec kVideoCodecsAnswer[] = {CreateVideoCodec(97, "H264")};
 
+// Match two codec lists for content, but ignore the ID.
+bool CodecListsMatch(rtc::ArrayView<const Codec> list1,
+                     rtc::ArrayView<const Codec> list2) {
+  if (list1.size() != list2.size()) {
+    return false;
+  }
+  for (size_t i = 0; i < list1.size(); ++i) {
+    Codec codec1 = list1[i];
+    Codec codec2 = list2[i];
+    codec1.id = Codec::kIdNotSet;
+    codec2.id = Codec::kIdNotSet;
+    if (codec1 != codec2) {
+      RTC_LOG(LS_ERROR) << "Mismatch at position " << i << " between " << codec1
+                        << " and " << codec2;
+      return false;
+    }
+  }
+  return true;
+}
+
 const RtpExtension kAudioRtpExtension1[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
     RtpExtension("http://google.com/testing/audio_something", 10),
@@ -438,8 +460,8 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
   MediaSessionDescriptionFactoryTest()
       : tdf1_(field_trials),
         tdf2_(field_trials),
-        f1_(nullptr, false, &ssrc_generator1, &tdf1_, nullptr),
-        f2_(nullptr, false, &ssrc_generator2, &tdf2_, nullptr) {
+        f1_(nullptr, false, &ssrc_generator1, &tdf1_, &pt_suggester_1_),
+        f2_(nullptr, false, &ssrc_generator2, &tdf2_, &pt_suggester_2_) {
     f1_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs1),
                          MAKE_VECTOR(kAudioCodecs1));
     f1_.set_video_codecs(MAKE_VECTOR(kVideoCodecs1),
@@ -688,6 +710,8 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
   UniqueRandomIdGenerator ssrc_generator2;
   TransportDescriptionFactory tdf1_;
   TransportDescriptionFactory tdf2_;
+  webrtc::FakePayloadTypeSuggester pt_suggester_1_;
+  webrtc::FakePayloadTypeSuggester pt_suggester_2_;
   MediaSessionDescriptionFactory f1_;
   MediaSessionDescriptionFactory f2_;
 };
@@ -2947,11 +2971,11 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 
   const AudioContentDescription* updated_acd =
       GetFirstAudioContentDescription(updated_offer.get());
-  EXPECT_THAT(updated_acd->codecs(), ElementsAreArray(kUpdatedAudioCodecOffer));
+  EXPECT_TRUE(CodecListsMatch(updated_acd->codecs(), kUpdatedAudioCodecOffer));
 
   const VideoContentDescription* updated_vcd =
       GetFirstVideoContentDescription(updated_offer.get());
-  EXPECT_THAT(updated_vcd->codecs(), ElementsAreArray(kUpdatedVideoCodecOffer));
+  EXPECT_TRUE(CodecListsMatch(updated_vcd->codecs(), kUpdatedVideoCodecOffer));
 }
 
 // Test that a reoffer does not reuse audio codecs from a previous media section
@@ -2979,7 +3003,13 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   // section was not recycled the payload types would match the initial offerer.
   const AudioContentDescription* acd =
       GetFirstAudioContentDescription(reoffer.get());
-  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecs2));
+  // EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecs2)),
+  // except that we don't want to check the PT numbers.
+  EXPECT_EQ(acd->codecs().size(),
+            sizeof(kAudioCodecs2) / sizeof(kAudioCodecs2[0]));
+  for (size_t i = 0; i < acd->codecs().size(); ++i) {
+    EXPECT_EQ(acd->codecs()[i].name, kAudioCodecs2[i].name);
+  }
 }
 
 // Test that a reoffer does not reuse video codecs from a previous media section
@@ -3006,7 +3036,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   // section was not recycled the payload types would match the initial offerer.
   const VideoContentDescription* vcd =
       GetFirstVideoContentDescription(reoffer.get());
-  EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecs2));
+  EXPECT_TRUE(CodecListsMatch(vcd->codecs(), kVideoCodecs2));
 }
 
 // Test that a reanswer does not reuse audio codecs from a previous media
@@ -3104,7 +3134,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   std::vector<Codec> expected_codecs = MAKE_VECTOR(kVideoCodecsAnswer);
   AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &expected_codecs);
 
-  EXPECT_EQ(expected_codecs, vcd->codecs());
+  EXPECT_TRUE(CodecListsMatch(expected_codecs, vcd->codecs()));
 
   // Now, make sure we get same result (except for the order) if `f2_` creates
   // an updated offer even though the default payload types between `f1_` and
@@ -3119,7 +3149,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   const VideoContentDescription* updated_vcd =
       GetFirstVideoContentDescription(updated_answer.get());
 
-  EXPECT_EQ(expected_codecs, updated_vcd->codecs());
+  EXPECT_TRUE(CodecListsMatch(expected_codecs, updated_vcd->codecs()));
 }
 
 // Regression test for:
@@ -3274,7 +3304,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   // New offer should attempt to add H263, and RTX for H264.
   expected_codecs.push_back(kVideoCodecs2[1]);
   AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs1[1].id), &expected_codecs);
-  EXPECT_EQ(expected_codecs, updated_vcd->codecs());
+  EXPECT_TRUE(CodecListsMatch(expected_codecs, updated_vcd->codecs()));
 }
 
 // Test that RTX is ignored when there is no associated payload type parameter.
@@ -3383,7 +3413,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   std::vector<Codec> expected_codecs = MAKE_VECTOR(kVideoCodecsAnswer);
   AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &expected_codecs);
 
-  EXPECT_EQ(expected_codecs, vcd->codecs());
+  EXPECT_TRUE(CodecListsMatch(expected_codecs, vcd->codecs()));
 }
 
 // Test that after one RTX codec has been negotiated, a new offer can attempt
@@ -3406,7 +3436,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, AddSecondRtxInNewOffer) {
 
   std::vector<Codec> expected_codecs = MAKE_VECTOR(kVideoCodecs1);
   AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &expected_codecs);
-  EXPECT_EQ(expected_codecs, vcd->codecs());
+  EXPECT_TRUE(CodecListsMatch(expected_codecs, vcd->codecs()));
 
   // Now, attempt to add RTX for H264-SVC.
   AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs1[0].id), &f1_codecs);
@@ -3418,7 +3448,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, AddSecondRtxInNewOffer) {
   vcd = GetFirstVideoContentDescription(updated_offer.get());
 
   AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs1[0].id), &expected_codecs);
-  EXPECT_EQ(expected_codecs, vcd->codecs());
+  EXPECT_TRUE(CodecListsMatch(expected_codecs, vcd->codecs()));
 }
 
 // Test that when RTX is used in conjunction with simulcast, an RTX ssrc is
@@ -4507,8 +4537,8 @@ class MediaProtocolTest : public testing::TestWithParam<const char*> {
   MediaProtocolTest()
       : tdf1_(field_trials_),
         tdf2_(field_trials_),
-        f1_(nullptr, false, &ssrc_generator1, &tdf1_, nullptr),
-        f2_(nullptr, false, &ssrc_generator2, &tdf2_, nullptr) {
+        f1_(nullptr, false, &ssrc_generator1, &tdf1_, &pt_suggester_1_),
+        f2_(nullptr, false, &ssrc_generator2, &tdf2_, &pt_suggester_2_) {
     f1_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs1),
                          MAKE_VECTOR(kAudioCodecs1));
     f1_.set_video_codecs(MAKE_VECTOR(kVideoCodecs1),
@@ -4527,6 +4557,8 @@ class MediaProtocolTest : public testing::TestWithParam<const char*> {
   webrtc::test::ScopedKeyValueConfig field_trials_;
   TransportDescriptionFactory tdf1_;
   TransportDescriptionFactory tdf2_;
+  webrtc::FakePayloadTypeSuggester pt_suggester_1_;
+  webrtc::FakePayloadTypeSuggester pt_suggester_2_;
   MediaSessionDescriptionFactory f1_;
   MediaSessionDescriptionFactory f2_;
   UniqueRandomIdGenerator ssrc_generator1;
@@ -4571,8 +4603,9 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestSetAudioCodecs) {
       std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id"))));
 
   UniqueRandomIdGenerator ssrc_generator;
+  webrtc::FakePayloadTypeSuggester pt_suggester;
   MediaSessionDescriptionFactory sf(nullptr, false, &ssrc_generator, &tdf,
-                                    nullptr);
+                                    &pt_suggester);
   std::vector<Codec> send_codecs = MAKE_VECTOR(kAudioCodecs1);
   std::vector<Codec> recv_codecs = MAKE_VECTOR(kAudioCodecs2);
 
@@ -4643,8 +4676,9 @@ void TestAudioCodecsOffer(RtpTransceiverDirection direction) {
       std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id"))));
 
   UniqueRandomIdGenerator ssrc_generator;
+  webrtc::FakePayloadTypeSuggester pt_suggester;
   MediaSessionDescriptionFactory sf(nullptr, false, &ssrc_generator, &tdf,
-                                    nullptr);
+                                    &pt_suggester);
   const std::vector<Codec> send_codecs = MAKE_VECTOR(kAudioCodecs1);
   const std::vector<Codec> recv_codecs = MAKE_VECTOR(kAudioCodecs2);
   const std::vector<Codec> sendrecv_codecs = MAKE_VECTOR(kAudioCodecsAnswer);
@@ -4685,13 +4719,15 @@ void TestAudioCodecsOffer(RtpTransceiverDirection direction) {
   }
 }
 
-const Codec kOfferAnswerCodecs[] = {CreateAudioCodec(0, "codec0", 16000, 1),
-                                    CreateAudioCodec(1, "codec1", 8000, 1),
-                                    CreateAudioCodec(2, "codec2", 8000, 1),
-                                    CreateAudioCodec(3, "codec3", 8000, 1),
-                                    CreateAudioCodec(4, "codec4", 8000, 2),
-                                    CreateAudioCodec(5, "codec5", 32000, 1),
-                                    CreateAudioCodec(6, "codec6", 48000, 1)};
+// Since the PT suggester reserves the static range for specific codecs,
+// PT numbers from the 36-63 range are used.
+const Codec kOfferAnswerCodecs[] = {CreateAudioCodec(40, "codec0", 16000, 1),
+                                    CreateAudioCodec(41, "codec1", 8000, 1),
+                                    CreateAudioCodec(42, "codec2", 8000, 1),
+                                    CreateAudioCodec(43, "codec3", 8000, 1),
+                                    CreateAudioCodec(44, "codec4", 8000, 2),
+                                    CreateAudioCodec(45, "codec5", 32000, 1),
+                                    CreateAudioCodec(46, "codec6", 48000, 1)};
 
 /* The codecs groups below are chosen as per the matrix below. The objective
  * is to have different sets of codecs in the inputs, to get unique sets of
@@ -4747,10 +4783,12 @@ void TestAudioCodecsAnswer(RtpTransceiverDirection offer_direction,
       rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
           new rtc::FakeSSLIdentity("answer_id"))));
   UniqueRandomIdGenerator ssrc_generator1, ssrc_generator2;
+  webrtc::FakePayloadTypeSuggester offer_pt_suggester;
   MediaSessionDescriptionFactory offer_factory(nullptr, false, &ssrc_generator1,
-                                               &offer_tdf, nullptr);
+                                               &offer_tdf, &offer_pt_suggester);
+  webrtc::FakePayloadTypeSuggester answer_pt_suggester;
   MediaSessionDescriptionFactory answer_factory(
-      nullptr, false, &ssrc_generator2, &answer_tdf, nullptr);
+      nullptr, false, &ssrc_generator2, &answer_tdf, &answer_pt_suggester);
 
   offer_factory.set_audio_codecs(
       VectorFromIndices(kOfferAnswerCodecs, kOfferSendCodecs),
@@ -4833,7 +4871,7 @@ void TestAudioCodecsAnswer(RtpTransceiverDirection offer_direction,
       bool first = true;
       os << "{";
       for (const auto& c : codecs) {
-        os << (first ? " " : ", ") << c.id;
+        os << (first ? " " : ", ") << c.id << ":" << c.name;
         first = false;
       }
       os << " }";
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 0542910941..b4e774d53c 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -84,7 +84,6 @@
 #include "pc/stream_collection.h"
 #include "pc/transceiver_list.h"
 #include "pc/usage_pattern.h"
-#include "pc/used_ids.h"
 #include "pc/webrtc_session_description_factory.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/crypto_random.h"
@@ -592,8 +591,7 @@ RTCError ValidatePayloadTypes(const cricket::SessionDescription& description) {
     if (type == cricket::MEDIA_TYPE_AUDIO ||
         type == cricket::MEDIA_TYPE_VIDEO) {
       for (const auto& codec : media_description->codecs()) {
-        if (!cricket::UsedPayloadTypes::IsIdValid(
-                codec, media_description->rtcp_mux())) {
+        if (!PayloadType::IsValid(codec.id, media_description->rtcp_mux())) {
           LOG_AND_RETURN_ERROR(
               RTCErrorType::INVALID_PARAMETER,
               "The media section with MID='" + content.mid() +
diff --git a/pc/test/integration_test_helpers.cc b/pc/test/integration_test_helpers.cc
index 525a370a40..bd9298c3b1 100644
--- a/pc/test/integration_test_helpers.cc
+++ b/pc/test/integration_test_helpers.cc
@@ -10,7 +10,37 @@
 
 #include "pc/test/integration_test_helpers.h"
 
+#include <cstddef>
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
 #include "api/audio/builtin_audio_processing_factory.h"
+#include "api/enable_media_with_defaults.h"
+#include "api/jsep.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/sequence_checker.h"
+#include "api/stats/rtcstats_objects.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/units/time_delta.h"
+#include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
+#include "p2p/base/basic_packet_socket_factory.h"
+#include "p2p/base/port_allocator.h"
+#include "p2p/client/basic_port_allocator.h"
+#include "pc/peer_connection_factory.h"
+#include "pc/test/fake_audio_capture_module.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/fake_network.h"
+#include "rtc_base/socket_server.h"
+#include "rtc_base/thread.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 
diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h
index e228386860..5d5293bc23 100644
--- a/pc/test/integration_test_helpers.h
+++ b/pc/test/integration_test_helpers.h
@@ -18,7 +18,6 @@
 #include <algorithm>
 #include <functional>
 #include <limits>
-#include <list>
 #include <map>
 #include <memory>
 #include <optional>
@@ -27,66 +26,44 @@
 #include <utility>
 #include <vector>
 
-#include "absl/algorithm/container.h"
+#include "absl/functional/any_invocable.h"
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
-#include "api/audio/audio_device.h"
-#include "api/audio/audio_processing.h"
 #include "api/audio_options.h"
 #include "api/candidate.h"
 #include "api/crypto/crypto_options.h"
 #include "api/data_channel_interface.h"
-#include "api/enable_media_with_defaults.h"
 #include "api/field_trials_view.h"
 #include "api/ice_transport_interface.h"
 #include "api/jsep.h"
+#include "api/make_ref_counted.h"
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
+#include "api/metronome/metronome.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtc_error.h"
-#include "api/rtc_event_log/rtc_event_log_factory.h"
-#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
 #include "api/rtc_event_log_output.h"
 #include "api/rtp_receiver_interface.h"
 #include "api/rtp_sender_interface.h"
 #include "api/rtp_transceiver_interface.h"
 #include "api/scoped_refptr.h"
-#include "api/stats/rtc_stats.h"
+#include "api/sequence_checker.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/stats/rtcstats_objects.h"
-#include "api/task_queue/default_task_queue_factory.h"
 #include "api/task_queue/pending_task_safety_flag.h"
-#include "api/task_queue/task_queue_factory.h"
 #include "api/test/mock_async_dns_resolver.h"
-#include "api/transport/field_trial_based_config.h"
-#include "api/uma_metrics.h"
 #include "api/units/time_delta.h"
 #include "api/video/video_rotation.h"
-#include "api/video_codecs/sdp_video_format.h"
-#include "api/video_codecs/video_decoder_factory.h"
-#include "api/video_codecs/video_encoder_factory.h"
-#include "call/call.h"
 #include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
-#include "media/base/media_engine.h"
 #include "media/base/stream_params.h"
-#include "media/engine/fake_webrtc_video_engine.h"
 #include "p2p/base/fake_ice_transport.h"
 #include "p2p/base/ice_transport_internal.h"
-#include "p2p/base/p2p_constants.h"
 #include "p2p/base/port.h"
 #include "p2p/base/port_allocator.h"
 #include "p2p/base/port_interface.h"
-#include "p2p/base/test_stun_server.h"
 #include "p2p/base/test_turn_customizer.h"
 #include "p2p/base/test_turn_server.h"
-#include "p2p/client/basic_port_allocator.h"
-#include "pc/dtmf_sender.h"
-#include "pc/local_audio_source.h"
-#include "pc/media_session.h"
-#include "pc/peer_connection.h"
 #include "pc/peer_connection_factory.h"
-#include "pc/peer_connection_proxy.h"
-#include "pc/rtp_media_utils.h"
 #include "pc/session_description.h"
 #include "pc/test/fake_audio_capture_module.h"
 #include "pc/test/fake_periodic_video_source.h"
@@ -97,28 +74,23 @@
 #include "pc/video_track_source.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/crypto_random.h"
-#include "rtc_base/event.h"
-#include "rtc_base/fake_clock.h"
 #include "rtc_base/fake_mdns_responder.h"
 #include "rtc_base/fake_network.h"
 #include "rtc_base/firewall_socket_server.h"
 #include "rtc_base/gunit.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/mdns_responder_interface.h"
-#include "rtc_base/numerics/safe_conversions.h"
-#include "rtc_base/rtc_certificate_generator.h"
 #include "rtc_base/socket_address.h"
+#include "rtc_base/socket_factory.h"
+#include "rtc_base/socket_server.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/task_queue_for_test.h"
-#include "rtc_base/task_utils/repeating_task.h"
-#include "rtc_base/test_certificate_verifier.h"
 #include "rtc_base/thread.h"
-#include "rtc_base/thread_annotations.h"
 #include "rtc_base/time_utils.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "system_wrappers/include/metrics.h"
 #include "test/gmock.h"
+#include "test/gtest.h"
 #include "test/scoped_key_value_config.h"
 
 namespace webrtc {
@@ -666,9 +638,7 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
 
   bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc) {
     auto observer = rtc::make_ref_counted<FakeSetRemoteDescriptionObserver>();
-    std::string str;
-    desc->ToString(&str);
-    RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription SDP:\n" << str;
+    RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription SDP:" << desc;
     pc()->SetRemoteDescription(std::move(desc), observer);  // desc.release());
     RemoveUnusedVideoRenderers();
     EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
diff --git a/pc/used_ids.h b/pc/used_ids.h
index 42ef00a7c0..d55eead98e 100644
--- a/pc/used_ids.h
+++ b/pc/used_ids.h
@@ -14,9 +14,7 @@
 #include <vector>
 
 #include "api/rtp_parameters.h"
-#include "media/base/codec.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
 
 namespace cricket {
 template <typename IdStruct>
@@ -88,41 +86,6 @@ class UsedIds {
   std::set<int> id_set_;
 };
 
-// Helper class used for finding duplicate RTP payload types among audio, video
-// and data codecs. When bundle is used the payload types may not collide.
-class UsedPayloadTypes : public UsedIds<Codec> {
- public:
-  UsedPayloadTypes()
-      : UsedIds<Codec>(kFirstDynamicPayloadTypeLowerRange,
-                       kLastDynamicPayloadTypeUpperRange) {}
-
-  // Check if a payload type is valid. The range [64-95] is forbidden
-  // when rtcp-mux is used.
-  static bool IsIdValid(Codec codec, bool rtcp_mux) {
-    if (rtcp_mux && (codec.id > kLastDynamicPayloadTypeLowerRange &&
-                     codec.id < kFirstDynamicPayloadTypeUpperRange)) {
-      return false;
-    }
-    return codec.id >= 0 && codec.id <= kLastDynamicPayloadTypeUpperRange;
-  }
-
- protected:
-  bool IsIdUsed(int new_id) override {
-    // Range marked for RTCP avoidance is "used".
-    if (new_id > kLastDynamicPayloadTypeLowerRange &&
-        new_id < kFirstDynamicPayloadTypeUpperRange)
-      return true;
-    return UsedIds<Codec>::IsIdUsed(new_id);
-  }
-
- private:
-  static const int kFirstDynamicPayloadTypeLowerRange = 35;
-  static const int kLastDynamicPayloadTypeLowerRange = 63;
-
-  static const int kFirstDynamicPayloadTypeUpperRange = 96;
-  static const int kLastDynamicPayloadTypeUpperRange = 127;
-};
-
 // Helper class used for finding duplicate RTP Header extension ids among
 // audio and video extensions.
 class UsedRtpHeaderExtensionIds : public UsedIds<webrtc::RtpExtension> {