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https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00
Use unified plan for 1:1 calls
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parent
c8bf0736a0
commit
c632a4acfc
3 changed files with 9 additions and 5 deletions
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@ -414,6 +414,8 @@ Rust_sessionDescriptionFromV4(bool offer,
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// We'll set it around just in case.
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// We'll set it around just in case.
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// But everything seems to work fine without it.
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// But everything seems to work fine without it.
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stream->cname = "CNAMECNAMECNAME!";
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stream->cname = "CNAMECNAMECNAME!";
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stream->set_stream_ids({"s"});
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}
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}
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audio->AddStream(audio_stream);
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audio->AddStream(audio_stream);
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@ -439,8 +441,7 @@ Rust_sessionDescriptionFromV4(bool offer,
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bundle.AddContentName(video_content_name);
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bundle.AddContentName(video_content_name);
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session->AddGroup(bundle);
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session->AddGroup(bundle);
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// This is the default and used for "Plan B" SDP, which is what we use in V1, V2, and V3.
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session->set_msid_signaling(cricket::kMsidSignalingMediaSection);
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session->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute);
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auto typ = offer ? SdpType::kOffer : SdpType::kAnswer;
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auto typ = offer ? SdpType::kOffer : SdpType::kAnswer;
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return new webrtc::JsepSessionDescription(typ, std::move(session), "1", "1");
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return new webrtc::JsepSessionDescription(typ, std::move(session), "1", "1");
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@ -384,7 +384,11 @@ RUSTEXPORT PeerConnectionInterface* Rust_createPeerConnection(
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config.audio_jitter_buffer_max_packets = audio_jitter_buffer_max_packets;
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config.audio_jitter_buffer_max_packets = audio_jitter_buffer_max_packets;
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config.set_audio_jitter_buffer_max_target_delay_ms(audio_jitter_buffer_max_target_delay_ms);
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config.set_audio_jitter_buffer_max_target_delay_ms(audio_jitter_buffer_max_target_delay_ms);
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config.set_audio_rtcp_report_interval_ms(audio_rtcp_report_interval_ms);
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config.set_audio_rtcp_report_interval_ms(audio_rtcp_report_interval_ms);
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config.sdp_semantics = SdpSemantics::kPlanB_DEPRECATED;
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if (kind == RffiPeerConnectionKind::kGroupCall) {
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config.sdp_semantics = SdpSemantics::kPlanB_DEPRECATED;
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} else {
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config.sdp_semantics = SdpSemantics::kUnifiedPlan;
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}
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if (ice_server.urls_size > 0) {
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if (ice_server.urls_size > 0) {
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webrtc::PeerConnectionInterface::IceServer rtc_ice_server;
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webrtc::PeerConnectionInterface::IceServer rtc_ice_server;
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rtc_ice_server.username = std::string(ice_server.username_borrowed);
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rtc_ice_server.username = std::string(ice_server.username_borrowed);
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@ -69,8 +69,7 @@
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- (instancetype)init {
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- (instancetype)init {
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// Copy defaults.
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// Copy defaults.
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webrtc::PeerConnectionInterface::RTCConfiguration config;
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webrtc::PeerConnectionInterface::RTCConfiguration config;
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// RingRTC Change to use "Plan B"
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config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
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config.sdp_semantics = webrtc::SdpSemantics::kPlanB_DEPRECATED;
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return [self initWithNativeConfiguration:config];
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return [self initWithNativeConfiguration:config];
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}
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}
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