[Unified Plan] Support legacy endpoints that do not use a=mid
These legacy endpoints were supported with Plan B since the SDP
parser would fill in default MID values if a=mid was absent that
happened to match the default offered MIDs.
With Unified Plan, these default MIDs changed so the autofilled
MIDs do not match any more.
This CL adds information to the SessionDescription struct to
indicate whether or not a=mid was present and modified
PeerConnection::SetRemoteDescription to copy MIDs from the local
description if the a=mid lines are not present.
Bug: webrtc:9540
Change-Id: Ibf923b4ad59edb0facd06ddbd01cc10c62fc48e6
Reviewed-on: https://webrtc-review.googlesource.com/c/114820
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26054}diff --git a/pc/peerconnection_jsep_unittest.cc b/pc/peerconnection_jsep_unittest.cc
index f0c9111..b860664 100644
--- a/pc/peerconnection_jsep_unittest.cc
+++ b/pc/peerconnection_jsep_unittest.cc
@@ -1604,4 +1604,94 @@
EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.type());
}
+// Removes the RTP header extension associated with the given URI from the media
+// description.
+static void RemoveRtpHeaderExtensionByUri(
+ MediaContentDescription* media_description,
+ absl::string_view uri) {
+ std::vector<RtpExtension> header_extensions =
+ media_description->rtp_header_extensions();
+ header_extensions.erase(std::remove_if(
+ header_extensions.begin(), header_extensions.end(),
+ [uri](const RtpExtension& extension) { return extension.uri == uri; }));
+ media_description->set_rtp_header_extensions(header_extensions);
+}
+
+// Transforms a session description to emulate a legacy endpoint which does not
+// support a=mid, BUNDLE, and the MID header extension.
+static void ClearMids(SessionDescriptionInterface* sdesc) {
+ cricket::SessionDescription* desc = sdesc->description();
+ desc->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
+ cricket::ContentInfo* audio_content = cricket::GetFirstAudioContent(desc);
+ if (audio_content) {
+ desc->GetTransportInfoByName(audio_content->name)->content_name = "";
+ audio_content->name = "";
+ RemoveRtpHeaderExtensionByUri(audio_content->media_description(),
+ RtpExtension::kMidUri);
+ }
+ cricket::ContentInfo* video_content = cricket::GetFirstVideoContent(desc);
+ if (video_content) {
+ desc->GetTransportInfoByName(video_content->name)->content_name = "";
+ video_content->name = "";
+ RemoveRtpHeaderExtensionByUri(video_content->media_description(),
+ RtpExtension::kMidUri);
+ }
+}
+
+// Test that negotiation works with legacy endpoints which do not support a=mid.
+TEST_F(PeerConnectionJsepTest, LegacyNoMidAudioOnlyOffer) {
+ auto caller = CreatePeerConnection();
+ caller->AddAudioTrack("audio");
+ auto callee = CreatePeerConnection();
+ callee->AddAudioTrack("audio");
+
+ auto offer = caller->CreateOffer();
+ ClearMids(offer.get());
+
+ ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
+ EXPECT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
+}
+TEST_F(PeerConnectionJsepTest, LegacyNoMidAudioVideoOffer) {
+ auto caller = CreatePeerConnection();
+ caller->AddAudioTrack("audio");
+ caller->AddVideoTrack("video");
+ auto callee = CreatePeerConnection();
+ callee->AddAudioTrack("audio");
+ callee->AddVideoTrack("video");
+
+ auto offer = caller->CreateOffer();
+ ClearMids(offer.get());
+
+ ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
+ EXPECT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
+}
+TEST_F(PeerConnectionJsepTest, LegacyNoMidAudioOnlyAnswer) {
+ auto caller = CreatePeerConnection();
+ caller->AddAudioTrack("audio");
+ auto callee = CreatePeerConnection();
+ callee->AddAudioTrack("audio");
+
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+
+ auto answer = callee->CreateAnswer();
+ ClearMids(answer.get());
+
+ EXPECT_TRUE(caller->SetRemoteDescription(std::move(answer)));
+}
+TEST_F(PeerConnectionJsepTest, LegacyNoMidAudioVideoAnswer) {
+ auto caller = CreatePeerConnection();
+ caller->AddAudioTrack("audio");
+ caller->AddVideoTrack("video");
+ auto callee = CreatePeerConnection();
+ callee->AddAudioTrack("audio");
+ callee->AddVideoTrack("video");
+
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+
+ auto answer = callee->CreateAnswer();
+ ClearMids(answer.get());
+
+ ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
+}
+
} // namespace webrtc