From 6f555dcafeadd843e47a70fb65a8fa49db7d294b Mon Sep 17 00:00:00 2001 From: "mallinath@webrtc.org" Date: Mon, 22 Aug 2011 18:33:51 +0000 Subject: [PATCH] Review URL: http://webrtc-codereview.appspot.com/119002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@413 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../source/talk/app/webrtc/webrtc_json.cc | 8 ++++++++ .../source/talk/app/webrtc/webrtcsession.cc | 14 ++++++++++++++ .../source/talk/session/phone/webrtcvideoengine.cc | 3 ++- 3 files changed, 24 insertions(+), 1 deletion(-) diff --git a/third_party_mods/libjingle/source/talk/app/webrtc/webrtc_json.cc b/third_party_mods/libjingle/source/talk/app/webrtc/webrtc_json.cc index 4073ab988..26f6a707c 100644 --- a/third_party_mods/libjingle/source/talk/app/webrtc/webrtc_json.cc +++ b/third_party_mods/libjingle/source/talk/app/webrtc/webrtc_json.cc @@ -270,6 +270,10 @@ bool ParseJSONSignalingMessage(const std::string& signaling_message, cricket::AudioContentDescription* audio_content = new cricket::AudioContentDescription(); ParseAudioCodec(mlines[i], audio_content); + + // enabling RTCP mux by default at both ends, without + // exchanging it through signaling message. + audio_content->set_rtcp_mux(true); audio_content->SortCodecs(); sdp->AddContent(cricket::CN_AUDIO, cricket::NS_JINGLE_RTP, audio_content); ParseICECandidates(mlines[i], candidates); @@ -277,6 +281,10 @@ bool ParseJSONSignalingMessage(const std::string& signaling_message, cricket::VideoContentDescription* video_content = new cricket::VideoContentDescription(); ParseVideoCodec(mlines[i], video_content); + + // enabling RTCP mux by default at both ends, without + // exchanging it through signaling message. + video_content->set_rtcp_mux(true); video_content->SortCodecs(); sdp->AddContent(cricket::CN_VIDEO, cricket::NS_JINGLE_RTP, video_content); ParseICECandidates(mlines[i], candidates); diff --git a/third_party_mods/libjingle/source/talk/app/webrtc/webrtcsession.cc b/third_party_mods/libjingle/source/talk/app/webrtc/webrtcsession.cc index 67d6dd097..37f8d0ff4 100644 --- a/third_party_mods/libjingle/source/talk/app/webrtc/webrtcsession.cc +++ b/third_party_mods/libjingle/source/talk/app/webrtc/webrtcsession.cc @@ -612,6 +612,9 @@ cricket::SessionDescription* WebRtcSession::CreateOffer() { video->AddCodec(*codec); } + // enabling RTCP mux by default at both ends, without + // exchanging it through signaling message. + video->set_rtcp_mux(true); video->SortCodecs(); offer->AddContent(cricket::CN_VIDEO, cricket::NS_JINGLE_RTP, video); } else { @@ -625,6 +628,9 @@ cricket::SessionDescription* WebRtcSession::CreateOffer() { audio->AddCodec(*codec); } + // enabling RTCP mux by default at both ends, without + // exchanging it through signaling message. + audio->set_rtcp_mux(true); audio->SortCodecs(); offer->AddContent(cricket::CN_AUDIO, cricket::NS_JINGLE_RTP, audio); } @@ -658,6 +664,10 @@ cricket::SessionDescription* WebRtcSession::CreateAnswer( } } } + + // enabling RTCP mux by default at both ends, without + // exchanging it through signaling message. + audio_accept->set_rtcp_mux(true); audio_accept->SortCodecs(); answer->AddContent(audio_content->name, audio_content->type, audio_accept); } @@ -684,6 +694,10 @@ cricket::SessionDescription* WebRtcSession::CreateAnswer( } } } + + // enabling RTCP mux by default at both ends, without + // exchanging it through signaling message. + video_accept->set_rtcp_mux(true); video_accept->SortCodecs(); answer->AddContent(video_content->name, video_content->type, video_accept); } diff --git a/third_party_mods/libjingle/source/talk/session/phone/webrtcvideoengine.cc b/third_party_mods/libjingle/source/talk/session/phone/webrtcvideoengine.cc index bb1157b24..cca79b011 100644 --- a/third_party_mods/libjingle/source/talk/session/phone/webrtcvideoengine.cc +++ b/third_party_mods/libjingle/source/talk/session/phone/webrtcvideoengine.cc @@ -566,7 +566,7 @@ bool WebRtcVideoMediaChannel::Init() { vie_channel_, *this) != 0) { ret = false; } else { - // EnableRtcp(); // by default RTCP is disabled. + EnableRtcp(); EnablePLI(); } return ret; @@ -901,6 +901,7 @@ int WebRtcVideoMediaChannel::SendPacket(int channel, const void* data, if (!network_interface_) { return -1; } + talk_base::Buffer packet(data, len, kMaxRtpPacketLen); return network_interface_->SendPacket(&packet) ? len : -1; }