commit 416fd907ecfb59402834c57ddc831014c373859b
parent 2eb400c53144947874748fee1b1dc4588678440a
Author: Dan Baker <dbaker@mozilla.com>
Date: Thu, 23 Oct 2025 15:57:24 -0600
Bug 1995393 - Vendor libwebrtc from ac2597e9ce
We already cherry-picked this when we vendored f4453c3711.
Upstream commit: https://webrtc.googlesource.com/src/+/ac2597e9ce2b05751ac7ebddb268e874462c5e9f
Revert "Use field clock instead of global clock where available"
This reverts commit f4453c3711eff9e8851364687cc55a31de5c059f.
Reason for revert: Broken downstream
Bug: webrtc:42223992
Original change's description:
> Use field clock instead of global clock where available
>
> Bug: webrtc:42223992
> Change-Id: Ia0eb6a5f9ebfe4031fc72026f31767b2ce7dcc2d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404280
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#45333}
Bug: webrtc:42223992
Change-Id: I5c496d744b977edcda80a1877f9bae1f5864af2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404581
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#45352}
Diffstat:
5 files changed, 139 insertions(+), 382 deletions(-)
diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor
@@ -1,4 +1,4 @@
# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
-libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-23T21:54:26.795938+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-23T21:57:12.784990+00:00.
# base of lastest vendoring
-ffd474e5fa
+ac2597e9ce
diff --git a/third_party/libwebrtc/moz-patch-stack/ac2597e9ce.no-op-cherry-pick-msg b/third_party/libwebrtc/moz-patch-stack/ac2597e9ce.no-op-cherry-pick-msg
@@ -1 +0,0 @@
-We already cherry-picked this when we vendored f4453c3711.
diff --git a/third_party/libwebrtc/moz-patch-stack/p0001.patch b/third_party/libwebrtc/moz-patch-stack/p0001.patch
@@ -1,242 +1,44 @@
-From: Daniel Baker <dbaker@mozilla.com>
-Date: Thu, 23 Oct 2025 15:03:13 -0600
-Subject: (tmp-cherry-pick) Revert "Use field clock instead of global clock
- where available" (ac2597e9ce)
+From: Philipp Hancke <phancke@meta.com>
+Date: Sat, 16 Aug 2025 12:15:50 -0700
+Subject: (cherry-pick-branch-heads/7339) [M140] sdp munging: do not misdetect
+ rtp attributes on datachannel
-This reverts commit f4453c3711eff9e8851364687cc55a31de5c059f.
+rtcp-mux and rtcp-rsize are unconditionally set for datachannel
+but since they are not serialized in the SDP they are not parsed.
+This leads the munging detector to misclassify this.
-Reason for revert: Broken downstream
+a unit test is added as a separate CL for easier backmerging.
-Bug: webrtc:42223992
-Original change's description:
-> Use field clock instead of global clock where available
->
-> Bug: webrtc:42223992
-> Change-Id: Ia0eb6a5f9ebfe4031fc72026f31767b2ce7dcc2d
-> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404280
-> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
-> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
-> Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
-> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
-> Cr-Commit-Position: refs/heads/main@{#45333}
+(cherry picked from commit 602b06b1125ea4d107fbfbda7d314b4157c4c74b)
-Bug: webrtc:42223992
-Change-Id: I5c496d744b977edcda80a1877f9bae1f5864af2a
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404581
-Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
-Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
-Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
-Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
-Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
-Cr-Commit-Position: refs/heads/main@{#45352}
+Bug: webrtc:414284082,chromium:40567530
+Fixed: chromium:439807581
+Change-Id: I4ebb2ad5062ea42fc47a314ed929de42f91bbea7
+Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404740
+Commit-Queue: Philipp Hancke <phancke@meta.com>
+Reviewed-by: Harald Alvestrand <hta@webrtc.org>
+Cr-Original-Commit-Position: refs/heads/main@{#45357}
+Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/405281
+Cr-Commit-Position: refs/branch-heads/7339@{#1}
+Cr-Branched-From: 9bd64751d9b3b35a820cb72c9029993e218146a1-refs/heads/main@{#45270}
---
- api/units/timestamp.h | 4 ----
- api/units/timestamp_unittest.cc | 1 -
- audio/channel_receive.cc | 5 +++--
- audio/voip/voip_core.cc | 3 ++-
- modules/audio_device/test_audio_device_impl.cc | 3 ++-
- p2p/base/port.cc | 6 +++---
- pc/rtc_stats_collector.cc | 2 +-
- video/adaptation/overuse_frame_detector.cc | 2 +-
- video/encoder_bitrate_adjuster.cc | 3 ++-
- video/video_receive_stream2.cc | 3 ++-
- 10 files changed, 16 insertions(+), 16 deletions(-)
+ pc/sdp_munging_detector.cc | 6 ++++++
+ 1 file changed, 6 insertions(+)
-diff --git a/api/units/timestamp.h b/api/units/timestamp.h
-index 0317e9faef..7831bba2d0 100644
---- a/api/units/timestamp.h
-+++ b/api/units/timestamp.h
-@@ -60,10 +60,6 @@ class Timestamp final : public rtc_units_impl::UnitBase<Timestamp> {
- constexpr T us() const {
- return ToValue<T>();
- }
-- template <typename T = int64_t>
-- constexpr T ns() const {
-- return ToMultiple<1000, T>();
-- }
-
- constexpr int64_t seconds_or(int64_t fallback_value) const {
- return ToFractionOr<1000000>(fallback_value);
-diff --git a/api/units/timestamp_unittest.cc b/api/units/timestamp_unittest.cc
-index c9254c25f0..1b88cb7f42 100644
---- a/api/units/timestamp_unittest.cc
-+++ b/api/units/timestamp_unittest.cc
-@@ -37,7 +37,6 @@ TEST(TimestampTest, ConstExpr) {
- EXPECT_EQ(kTimestampSeconds.seconds(), kValue);
- EXPECT_EQ(kTimestampMs.ms(), kValue);
- EXPECT_EQ(kTimestampUs.us(), kValue);
-- EXPECT_EQ(kTimestampUs.ns(), kValue * 1000);
- }
-
- TEST(TimestampTest, GetBackSameValues) {
-diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
-index 90d3c70aa0..eda5824569 100644
---- a/audio/channel_receive.cc
-+++ b/audio/channel_receive.cc
-@@ -78,6 +78,7 @@
- #include "rtc_base/synchronization/mutex.h"
- #include "rtc_base/system/no_unique_address.h"
- #include "rtc_base/thread_annotations.h"
-+#include "rtc_base/time_utils.h"
- #include "rtc_base/trace_event.h"
- #include "system_wrappers/include/metrics.h"
- #include "system_wrappers/include/ntp_time.h"
-@@ -657,7 +658,7 @@ void ChannelReceive::SetReceiveCodecs(
-
- void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
- RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-- Timestamp now = env_.clock().CurrentTime();
-+ Timestamp now = Timestamp::Millis(TimeMillis());
-
- last_received_rtp_timestamp_ = packet.Timestamp();
- last_received_rtp_system_time_ = now;
-@@ -754,7 +755,7 @@ void ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
- RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-
- // Store playout timestamp for the received RTCP packet
-- UpdatePlayoutTimestamp(true, env_.clock().CurrentTime());
-+ UpdatePlayoutTimestamp(true, Timestamp::Millis(TimeMillis()));
-
- // Deliver RTCP packet to RTP/RTCP module for parsing
- rtp_rtcp_->IncomingRtcpPacket(MakeArrayView(data, length));
-diff --git a/audio/voip/voip_core.cc b/audio/voip/voip_core.cc
-index d3ce340ac9..7676811698 100644
---- a/audio/voip/voip_core.cc
-+++ b/audio/voip/voip_core.cc
-@@ -40,6 +40,7 @@
- #include "rtc_base/logging.h"
- #include "rtc_base/random.h"
- #include "rtc_base/synchronization/mutex.h"
-+#include "rtc_base/time_utils.h"
-
- namespace webrtc {
-
-@@ -151,7 +152,7 @@ ChannelId VoipCore::CreateChannel(Transport* transport,
-
- // Set local ssrc to random if not set by caller.
- if (!local_ssrc) {
-- Random random(env_.clock().TimeInMicroseconds());
-+ Random random(TimeMicros());
- local_ssrc = random.Rand<uint32_t>();
- }
-
-diff --git a/modules/audio_device/test_audio_device_impl.cc b/modules/audio_device/test_audio_device_impl.cc
-index 80402d3ea3..fe3d10dcf3 100644
---- a/modules/audio_device/test_audio_device_impl.cc
-+++ b/modules/audio_device/test_audio_device_impl.cc
-@@ -25,6 +25,7 @@
- #include "rtc_base/checks.h"
- #include "rtc_base/synchronization/mutex.h"
- #include "rtc_base/task_utils/repeating_task.h"
-+#include "rtc_base/time_utils.h"
-
- namespace webrtc {
- namespace {
-@@ -174,7 +175,7 @@ void TestAudioDevice::ProcessAudio() {
- audio_buffer_->SetRecordedBuffer(
- recording_buffer_.data(),
- recording_buffer_.size() / capturer_->NumChannels(),
-- std::make_optional(env_.clock().CurrentTime().ns()));
-+ std::make_optional(TimeNanos()));
- audio_buffer_->DeliverRecordedData();
+diff --git a/pc/sdp_munging_detector.cc b/pc/sdp_munging_detector.cc
+index 04f0d53fd0..88f38da30d 100644
+--- a/pc/sdp_munging_detector.cc
++++ b/pc/sdp_munging_detector.cc
+@@ -427,6 +427,12 @@ SdpMungingType DetermineSdpMungingType(
}
- if (!keep_capturing) {
-diff --git a/p2p/base/port.cc b/p2p/base/port.cc
-index 033e3f6dd2..5a43cc7c70 100644
---- a/p2p/base/port.cc
-+++ b/p2p/base/port.cc
-@@ -53,6 +53,7 @@
- #include "rtc_base/string_encode.h"
- #include "rtc_base/string_utils.h"
- #include "rtc_base/strings/string_builder.h"
-+#include "rtc_base/time_utils.h"
- #include "rtc_base/trace_event.h"
- #include "rtc_base/weak_ptr.h"
-
-@@ -829,8 +830,7 @@ void Port::DestroyIfDead() {
- bool dead =
- (state_ == State::INIT || state_ == State::PRUNED) &&
- connections_.empty() &&
-- env_.clock().TimeInMilliseconds() - last_time_all_connections_removed_ >=
-- timeout_delay_;
-+ TimeMillis() - last_time_all_connections_removed_ >= timeout_delay_;
- if (dead) {
- Destroy();
- }
-@@ -899,7 +899,7 @@ bool Port::OnConnectionDestroyed(Connection* conn) {
- // fails and is removed before kPortTimeoutDelay, then this message will
- // not cause the Port to be destroyed.
- if (connections_.empty()) {
-- last_time_all_connections_removed_ = env_.clock().TimeInMilliseconds();
-+ last_time_all_connections_removed_ = TimeMillis();
- PostDestroyIfDead(/*delayed=*/true);
- }
-
-diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
-index ce0954a7b3..be90539f93 100644
---- a/pc/rtc_stats_collector.cc
-+++ b/pc/rtc_stats_collector.cc
-@@ -1207,7 +1207,7 @@ void RTCStatsCollector::GetStatsReportInternal(
- requests_.push_back(std::move(request));
-
- // "Now" using a monotonically increasing timer.
-- int64_t cache_now_us = env_.clock().TimeInMicroseconds();
-+ int64_t cache_now_us = TimeMicros();
- if (cached_report_ &&
- cache_now_us - cache_timestamp_us_ <= cache_lifetime_us_) {
- // We have a fresh cached report to deliver. Deliver asynchronously, since
-diff --git a/video/adaptation/overuse_frame_detector.cc b/video/adaptation/overuse_frame_detector.cc
-index 827b0a5dd9..959dbb0418 100644
---- a/video/adaptation/overuse_frame_detector.cc
-+++ b/video/adaptation/overuse_frame_detector.cc
-@@ -600,7 +600,7 @@ void OveruseFrameDetector::CheckForOveruse(
- !encode_usage_percent_)
- return;
-
-- int64_t now_ms = env_.clock().TimeInMilliseconds();
-+ int64_t now_ms = TimeMillis();
- const char* action = "NoAction";
-
- if (IsOverusing(*encode_usage_percent_)) {
-diff --git a/video/encoder_bitrate_adjuster.cc b/video/encoder_bitrate_adjuster.cc
-index 1d17e10b87..d9d64e4f0c 100644
---- a/video/encoder_bitrate_adjuster.cc
-+++ b/video/encoder_bitrate_adjuster.cc
-@@ -31,6 +31,7 @@
- #include "rtc_base/checks.h"
- #include "rtc_base/experiments/rate_control_settings.h"
- #include "rtc_base/logging.h"
-+#include "rtc_base/time_utils.h"
- #include "system_wrappers/include/clock.h"
- #include "video/encoder_overshoot_detector.h"
- #include "video/rate_utilization_tracker.h"
-@@ -385,7 +386,7 @@ void EncoderBitrateAdjuster::OnEncodedFrame(DataSize size,
- // Detectors may not exist, for instance if ScreenshareLayers is used.
- auto& detector = overshoot_detectors_[stream_index][temporal_index];
- if (detector) {
-- detector->OnEncodedFrame(size.bytes(), clock_.TimeInMilliseconds());
-+ detector->OnEncodedFrame(size.bytes(), TimeMillis());
- }
- if (media_rate_trackers_[stream_index]) {
- media_rate_trackers_[stream_index]->OnDataProduced(size,
-diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
-index eccec79bb3..4dff35605c 100644
---- a/video/video_receive_stream2.cc
-+++ b/video/video_receive_stream2.cc
-@@ -81,6 +81,7 @@
- #include "rtc_base/strings/string_builder.h"
- #include "rtc_base/synchronization/mutex.h"
- #include "rtc_base/system/file_wrapper.h"
-+#include "rtc_base/time_utils.h"
- #include "rtc_base/trace_event.h"
- #include "system_wrappers/include/clock.h"
- #include "system_wrappers/include/ntp_time.h"
-@@ -586,7 +587,7 @@ void VideoReceiveStream2::CreateAndRegisterExternalDecoder(
- char filename_buffer[256];
- SimpleStringBuilder ssb(filename_buffer);
- ssb << decoded_output_file << "/webrtc_receive_stream_" << remote_ssrc()
-- << "-" << env_.clock().TimeInMicroseconds() << ".ivf";
-+ << "-" << TimeMicros() << ".ivf";
- video_decoder = CreateFrameDumpingDecoderWrapper(
- std::move(video_decoder), FileWrapper::OpenWriteOnly(ssb.str()));
- }
+ // Validate video and audio contents.
+ MediaType media_type = last_created_media_description->type();
++ bool is_rtp =
++ media_type == MediaType::AUDIO || media_type == MediaType::VIDEO;
++ if (!is_rtp) {
++ // The checks that follow only apply for RTP-based contents.
++ continue;
++ }
+ if (media_type == MediaType::VIDEO) {
+ type = DetermineVideoSdpMungingType(last_created_media_description,
+ media_description_to_set);
diff --git a/third_party/libwebrtc/moz-patch-stack/p0002.patch b/third_party/libwebrtc/moz-patch-stack/p0002.patch
@@ -1,44 +1,109 @@
-From: Philipp Hancke <phancke@meta.com>
-Date: Sat, 16 Aug 2025 12:15:50 -0700
-Subject: (cherry-pick-branch-heads/7339) [M140] sdp munging: do not misdetect
- rtp attributes on datachannel
+From: Jonas Oreland <jonaso@webrtc.org>
+Date: Tue, 26 Aug 2025 15:57:29 +0200
+Subject: (cherry-pick-branch-heads/7339) [M140] DTLS1.3: Fix missing
+ retransmission due to failure to set timer.
-rtcp-mux and rtcp-rsize are unconditionally set for datachannel
-but since they are not serialized in the SDP they are not parsed.
-This leads the munging detector to misclassify this.
+DTLS 1.3 considers itself connected earlier than DTLS 1.2 did - when
+second flight reaches the client. This CL fixes a bug that when
+client is connected (state_ != SSL_CONNECTING), it would not
+continue retransmitting. Continuous retransmission is needed
+when the third flight is lost multiple times. Or really anytime that DTLS request it :)
-a unit test is added as a separate CL for easier backmerging.
+This fixes the TODO in dtls_ice_integrationtest.cc in which dtls1.3
+spuriously failed with certain (packet loss intensive) configurations.
-(cherry picked from commit 602b06b1125ea4d107fbfbda7d314b4157c4c74b)
+CREDITS: sergeysu@ that found and fixed the problem!
-Bug: webrtc:414284082,chromium:40567530
-Fixed: chromium:439807581
-Change-Id: I4ebb2ad5062ea42fc47a314ed929de42f91bbea7
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404740
-Commit-Queue: Philipp Hancke <phancke@meta.com>
+(cherry picked from commit 4221e1bc2fc360448c1663f0bbae058ff9f65a48)
+
+No-Try: true
+Bug: chromium:441245658, chromium:441486101
+Change-Id: I3302f6f384d7e4cda090184094a6fadaf7e4f129
+Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406320
+Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
-Cr-Original-Commit-Position: refs/heads/main@{#45357}
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/405281
-Cr-Commit-Position: refs/branch-heads/7339@{#1}
+Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
+Cr-Original-Commit-Position: refs/heads/main@{#45439}
+Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407780
+Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
+Cr-Commit-Position: refs/branch-heads/7339@{#2}
Cr-Branched-From: 9bd64751d9b3b35a820cb72c9029993e218146a1-refs/heads/main@{#45270}
---
- pc/sdp_munging_detector.cc | 6 ++++++
- 1 file changed, 6 insertions(+)
+ p2p/dtls/dtls_ice_integrationtest.cc | 6 +-----
+ rtc_base/openssl_stream_adapter.cc | 15 ++++++++++++---
+ rtc_base/openssl_stream_adapter.h | 2 ++
+ 3 files changed, 15 insertions(+), 8 deletions(-)
-diff --git a/pc/sdp_munging_detector.cc b/pc/sdp_munging_detector.cc
-index 04f0d53fd0..88f38da30d 100644
---- a/pc/sdp_munging_detector.cc
-+++ b/pc/sdp_munging_detector.cc
-@@ -427,6 +427,12 @@ SdpMungingType DetermineSdpMungingType(
+diff --git a/p2p/dtls/dtls_ice_integrationtest.cc b/p2p/dtls/dtls_ice_integrationtest.cc
+index 339238d664..1610f0002a 100644
+--- a/p2p/dtls/dtls_ice_integrationtest.cc
++++ b/p2p/dtls/dtls_ice_integrationtest.cc
+@@ -127,11 +127,7 @@ class DtlsIceIntegrationTest : public ::testing::TestWithParam<std::tuple<
+
+ BuiltInNetworkBehaviorConfig networkBehavior;
+ networkBehavior.link_capacity = DataRate::KilobitsPerSec(220);
+- // TODO (webrtc:383141571) : Investigate why this testcase fails for
+- // DTLS 1.3 delay if networkBehavior.queue_delay_ms = 100ms.
+- // - unless both peers support dtls in stun, in which case it passes.
+- // - note: only for dtls1.3, it works for dtls1.2!
+- networkBehavior.queue_delay_ms = 50;
++ networkBehavior.queue_delay_ms = 100;
+ networkBehavior.queue_length_packets = 30;
+ networkBehavior.loss_percent = 50;
+
+diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc
+index 50be6b5c26..8c952cd3a2 100644
+--- a/rtc_base/openssl_stream_adapter.cc
++++ b/rtc_base/openssl_stream_adapter.cc
+@@ -852,10 +852,15 @@ void OpenSSLStreamAdapter::SetTimeout(int delay_ms) {
+ // We check the timer even after SSL_CONNECTED,
+ // but ContinueSSL() is only needed when SSL_CONNECTING
+ if (state_ == SSL_CONNECTING) {
++ // Note: timeout is set inside ContinueSSL()
+ ContinueSSL();
++ } else if (state_ == SSL_CONNECTED) {
++ MaybeSetTimeout();
++ } else {
++ RTC_DCHECK_NOTREACHED() << "state_: " << state_;
+ }
+ } else {
+- RTC_DCHECK_NOTREACHED();
++ RTC_DCHECK_NOTREACHED() << "flag->alive() == false";
+ }
+ // This callback will never run again (stopped above).
+ return TimeDelta::PlusInfinity();
+@@ -970,6 +975,12 @@ int OpenSSLStreamAdapter::ContinueSSL() {
+ }
+ }
+
++ MaybeSetTimeout();
++
++ return 0;
++}
++
++void OpenSSLStreamAdapter::MaybeSetTimeout() {
+ if (ssl_ != nullptr) {
+ struct timeval timeout;
+ if (DTLSv1_get_timeout(ssl_, &timeout)) {
+@@ -977,8 +988,6 @@ int OpenSSLStreamAdapter::ContinueSSL() {
+ SetTimeout(delay);
}
- // Validate video and audio contents.
- MediaType media_type = last_created_media_description->type();
-+ bool is_rtp =
-+ media_type == MediaType::AUDIO || media_type == MediaType::VIDEO;
-+ if (!is_rtp) {
-+ // The checks that follow only apply for RTP-based contents.
-+ continue;
-+ }
- if (media_type == MediaType::VIDEO) {
- type = DetermineVideoSdpMungingType(last_created_media_description,
- media_description_to_set);
+ }
+-
+- return 0;
+ }
+
+ void OpenSSLStreamAdapter::Error(absl::string_view context,
+diff --git a/rtc_base/openssl_stream_adapter.h b/rtc_base/openssl_stream_adapter.h
+index 856d37c156..c10e93e3d4 100644
+--- a/rtc_base/openssl_stream_adapter.h
++++ b/rtc_base/openssl_stream_adapter.h
+@@ -210,6 +210,8 @@ class OpenSSLStreamAdapter final : public SSLStreamAdapter {
+ !peer_certificate_digest_value_.empty();
+ }
+
++ void MaybeSetTimeout();
++
+ const std::unique_ptr<StreamInterface> stream_;
+ absl::AnyInvocable<void(SSLHandshakeError)> handshake_error_;
+
diff --git a/third_party/libwebrtc/moz-patch-stack/p0003.patch b/third_party/libwebrtc/moz-patch-stack/p0003.patch
@@ -1,109 +0,0 @@
-From: Jonas Oreland <jonaso@webrtc.org>
-Date: Tue, 26 Aug 2025 15:57:29 +0200
-Subject: (cherry-pick-branch-heads/7339) [M140] DTLS1.3: Fix missing
- retransmission due to failure to set timer.
-
-DTLS 1.3 considers itself connected earlier than DTLS 1.2 did - when
-second flight reaches the client. This CL fixes a bug that when
-client is connected (state_ != SSL_CONNECTING), it would not
-continue retransmitting. Continuous retransmission is needed
-when the third flight is lost multiple times. Or really anytime that DTLS request it :)
-
-This fixes the TODO in dtls_ice_integrationtest.cc in which dtls1.3
-spuriously failed with certain (packet loss intensive) configurations.
-
-CREDITS: sergeysu@ that found and fixed the problem!
-
-(cherry picked from commit 4221e1bc2fc360448c1663f0bbae058ff9f65a48)
-
-No-Try: true
-Bug: chromium:441245658, chromium:441486101
-Change-Id: I3302f6f384d7e4cda090184094a6fadaf7e4f129
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406320
-Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
-Reviewed-by: Harald Alvestrand <hta@webrtc.org>
-Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
-Cr-Original-Commit-Position: refs/heads/main@{#45439}
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407780
-Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
-Cr-Commit-Position: refs/branch-heads/7339@{#2}
-Cr-Branched-From: 9bd64751d9b3b35a820cb72c9029993e218146a1-refs/heads/main@{#45270}
----
- p2p/dtls/dtls_ice_integrationtest.cc | 6 +-----
- rtc_base/openssl_stream_adapter.cc | 15 ++++++++++++---
- rtc_base/openssl_stream_adapter.h | 2 ++
- 3 files changed, 15 insertions(+), 8 deletions(-)
-
-diff --git a/p2p/dtls/dtls_ice_integrationtest.cc b/p2p/dtls/dtls_ice_integrationtest.cc
-index 339238d664..1610f0002a 100644
---- a/p2p/dtls/dtls_ice_integrationtest.cc
-+++ b/p2p/dtls/dtls_ice_integrationtest.cc
-@@ -127,11 +127,7 @@ class DtlsIceIntegrationTest : public ::testing::TestWithParam<std::tuple<
-
- BuiltInNetworkBehaviorConfig networkBehavior;
- networkBehavior.link_capacity = DataRate::KilobitsPerSec(220);
-- // TODO (webrtc:383141571) : Investigate why this testcase fails for
-- // DTLS 1.3 delay if networkBehavior.queue_delay_ms = 100ms.
-- // - unless both peers support dtls in stun, in which case it passes.
-- // - note: only for dtls1.3, it works for dtls1.2!
-- networkBehavior.queue_delay_ms = 50;
-+ networkBehavior.queue_delay_ms = 100;
- networkBehavior.queue_length_packets = 30;
- networkBehavior.loss_percent = 50;
-
-diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc
-index 50be6b5c26..8c952cd3a2 100644
---- a/rtc_base/openssl_stream_adapter.cc
-+++ b/rtc_base/openssl_stream_adapter.cc
-@@ -852,10 +852,15 @@ void OpenSSLStreamAdapter::SetTimeout(int delay_ms) {
- // We check the timer even after SSL_CONNECTED,
- // but ContinueSSL() is only needed when SSL_CONNECTING
- if (state_ == SSL_CONNECTING) {
-+ // Note: timeout is set inside ContinueSSL()
- ContinueSSL();
-+ } else if (state_ == SSL_CONNECTED) {
-+ MaybeSetTimeout();
-+ } else {
-+ RTC_DCHECK_NOTREACHED() << "state_: " << state_;
- }
- } else {
-- RTC_DCHECK_NOTREACHED();
-+ RTC_DCHECK_NOTREACHED() << "flag->alive() == false";
- }
- // This callback will never run again (stopped above).
- return TimeDelta::PlusInfinity();
-@@ -970,6 +975,12 @@ int OpenSSLStreamAdapter::ContinueSSL() {
- }
- }
-
-+ MaybeSetTimeout();
-+
-+ return 0;
-+}
-+
-+void OpenSSLStreamAdapter::MaybeSetTimeout() {
- if (ssl_ != nullptr) {
- struct timeval timeout;
- if (DTLSv1_get_timeout(ssl_, &timeout)) {
-@@ -977,8 +988,6 @@ int OpenSSLStreamAdapter::ContinueSSL() {
- SetTimeout(delay);
- }
- }
--
-- return 0;
- }
-
- void OpenSSLStreamAdapter::Error(absl::string_view context,
-diff --git a/rtc_base/openssl_stream_adapter.h b/rtc_base/openssl_stream_adapter.h
-index 856d37c156..c10e93e3d4 100644
---- a/rtc_base/openssl_stream_adapter.h
-+++ b/rtc_base/openssl_stream_adapter.h
-@@ -210,6 +210,8 @@ class OpenSSLStreamAdapter final : public SSLStreamAdapter {
- !peer_certificate_digest_value_.empty();
- }
-
-+ void MaybeSetTimeout();
-+
- const std::unique_ptr<StreamInterface> stream_;
- absl::AnyInvocable<void(SSLHandshakeError)> handshake_error_;
-
