WEBRTC SDP
2016-07-19 23:07
633 查看
本端SDP形成
1.发起CreateOffer请求 或CreateAnswer响应
2.PeerConnection::CreateOffer->PeerConnection::GetOptionsForOffer[获取SDP媒体属性]
3.WebRtcSession::CreateOffer[每个PeerConnection一个WebRtcSession]
4.WebRtcSessionDescriptionFactory::CreateOffer->WebRtcSessionDescriptionFactory::InternalCreateOffer
5.MediaSessionDescriptionFactory::CreateOffer->GetCodecsToOffer[获取音视频数据的所支持的编码]
MediaSessionDescriptionFactory::AddDataContentForOffer->FilterDataCodecs->
set_protocol(kMediaProtocolSctp)->CreateMediaContentOffer->MediaContentDescriptionImpl::AddCodecs
SessionDescription::AddContent->MediaSessionDescriptionFactory::AddTransportOffer
6.JsepSessionDescription::Initialize(SessionDescription* description)[Offer SDP已创建]
7.WebRtcSessionDescriptionFactory::PostCreateSessionDescriptionSucceeded->
Post(this, MSG_CREATE_SESSIONDESCRIPTION_SUCCESS)[生成SDP已成功]->
CreateSessionDescriptionObserver::OnSuccess(SessionDescriptionInterface* desc)[回调给用户]
|OnFailure(const std::string& error)[错误返回]
设置本端SDP
注:用户主动调用PeerConnection::CreateOffer 通过回调OnSuccess拿到了SDP对象,之后需要设置本端SDP
1.PeerConnection::SetLocalDescription->
2.WebRtcSession::SetLocalDescription->WebRtcSession::ValidateSessionDescription[合法性校验]->
VerifyCrypto-ValidateBundleSettings
3.Post(MSG_SET_SESSIONDESCRIPTION_SUCCESS)[设置SDP已成功]->
SetSessionDescriptionObserver::OnSuccess()[回调给用户]
4.WebRtcSession::MaybeStartGathering->TransportController::MaybeStartGathering->
5.P2PTransportChannel::MaybeStartGathering
Offer对象转换成SDP字符串(需要发给对端)
1.JsepSessionDescription::ToString
2.SdpSerialize[webrtcsdp.cc]->
接收对端SDP
1.CreateSessionDescription(const std::string& sdp)->JsepSessionDescription::Initialize->
2.SdpDeserialize->ParseMediaDescription->
3.ParseContentDescription->ParseContent->
4.以数据通道为例AddSctpDataCodec->MediaContentDescriptionImpl::AddCodec[google-sctp-data 90000 ]
5.SessionDescription::AddContent
const char NS_JINGLE_RTP[] = "urn:xmpp:jingle:apps:rtp:1";
sctp 的data 对应得ContentInfos中的type必须为NS_JINGLE_DRAFT_SCTP
const char NS_JINGLE_DRAFT_SCTP[] = "google:jingle:sctp";
SCTP 去掉DTLS 出现如下错误:
(channel.cc:2195): Setting local data description
(channel.cc:867): Failure in SetLocalContent with action 2
(webrtcsession.cc:327): Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote data description recv parameters..
(conductor.cc:104): OnFailure Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote data description recv parameters..
将bool IsDtlsSctp(const std::string& protocol) {
// This intentionally excludes "SCTP" and "SCTP/DTLS".
return protocol.find(cricket::kMediaProtocolDtlsSctp) != std::string::npos;
//return protocol.find(cricket::kMediaProtocolSctp) != std::string::npos;//add by sjz
}
将kMediaProtocolDtlsSctp改为kMediaProtocolSctp
a=sctpmap:5000 webrtc-datachannel 1024
========================sdp================================
local:
v=0
o=- 8900646765147570832 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS stream_label
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 127 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:qiC7
a=ice-pwd:97xpdM9B0N8ox6jbS2PI+0G7
a=fingerprint:sha-256 A8:C7:B8:88:DE:B9:F6:70:A3:6E:9D:C7:FD:41:4C:1E:61:9F:B7:E1:92:5F:0A:4C:48:FD:9E:D6:B7:B2:C9:55
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:127 red/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:3486011156 cname:RS5piJcFSbw2XOoP
a=ssrc:3486011156 msid:stream_label audio_label
a=ssrc:3486011156 mslabel:stream_label
a=ssrc:3486011156 label:audio_label
m=video 9 UDP/TLS/RTP/SAVPF 100 101 116 117 96 97 98
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:qiC7
a=ice-pwd:97xpdM9B0N8ox6jbS2PI+0G7
a=fingerprint:sha-256 A8:C7:B8:88:DE:B9:F6:70:A3:6E:9D:C7:FD:41:4C:1E:61:9F:B7:E1:92:5F:0A:4C:48:FD:9E:D6:B7:B2:C9:55
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtpmap:101 VP9/90000
a=rtcp-fb:101 ccm fir
a=rtcp-fb:101 nack
a=rtcp-fb:101 nack pli
a=rtcp-fb:101 goog-remb
a=rtcp-fb:101 transport-cc
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=101
a=rtpmap:98 rtx/90000
a=fmtp:98 apt=116
a=ssrc-group:FID 3678723866 3268471119
a=ssrc:3678723866 cname:RS5piJcFSbw2XOoP
a=ssrc:3678723866 msid:stream_label video_label
a=ssrc:3678723866 mslabel:stream_label
a=ssrc:3678723866 label:video_label
a=ssrc:3268471119 cname:RS5piJcFSbw2XOoP
a=ssrc:3268471119 msid:stream_label video_label
a=ssrc:3268471119 mslabel:stream_label
a=ssrc:3268471119 label:video_label
{
"candidate" : "candidate:430735571 1 udp 2122260223 192.168.1.102 59637 typ host generation 0 ufrag pa+u network-id 3 network-cost 50",
"sdpMLineIndex" : 0,
"sdpMid" : "audio"
}
remote:
v=0
o=- 5895813448736697051 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 127 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:1MuA
a=ice-pwd:YAf0BbrY4nSGW5iQ3faC1MVC
a=fingerprint:sha-256 C3:A8:1B:82:A4:2C:93:4C:F6:67:9E:E4:F3:00:7B:D1:BB:56:4D:D4:C1:9E:34:E7:82:73:D9:D0:AF:02:D2:D2
a=setup:active
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:127 red/8000
a=rtpmap:126 telephone-event/8000
m=video 9 UDP/TLS/RTP/SAVPF 100 101 116 117 96 97 98
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:1MuA
a=ice-pwd:YAf0BbrY4nSGW5iQ3faC1MVC
a=fingerprint:sha-256 C3:A8:1B:82:A4:2C:93:4C:F6:67:9E:E4:F3:00:7B:D1:BB:56:4D:D4:C1:9E:34:E7:82:73:D9:D0:AF:02:D2:D2
a=setup:active
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=recvonly
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtpmap:101 VP9/90000
a=rtcp-fb:101 ccm fir
a=rtcp-fb:101 nack
a=rtcp-fb:101 nack pli
a=rtcp-fb:101 goog-remb
a=rtcp-fb:101 transport-cc
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=101
a=rtpmap:98 rtx/90000
a=fmtp:98 apt=116
============auido and data=============
v=0
o=- 1390922805343918830 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio data
a=msid-semantic: WMS
m=audio 9 RTP/AVPF 111 8
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:ShBp
a=ice-pwd:a7DnyIUa0qvcB4yEmzOZBC3c
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:8 PCMA/8000
m=application 9 SCTP 5000
c=IN IP4 0.0.0.0
b=AS:30
a=ice-ufrag:ShBp
a=ice-pwd:a7DnyIUa0qvcB4yEmzOZBC3c
a=mid:data
a=sctpmap:5000 webrtc-datachannel 1024
本端SDP形成
1.发起CreateOffer请求 或CreateAnswer响应
2.PeerConnection::CreateOffer->PeerConnection::GetOptionsForOffer[获取SDP媒体属性]
3.WebRtcSession::CreateOffer[每个PeerConnection一个WebRtcSession]
4.WebRtcSessionDescriptionFactory::CreateOffer->WebRtcSessionDescriptionFactory::InternalCreateOffer
5.MediaSessionDescriptionFactory::CreateOffer->GetCodecsToOffer[获取音视频数据的所支持的编码]
MediaSessionDescriptionFactory::AddDataContentForOffer->FilterDataCodecs->
set_protocol(kMediaProtocolSctp)->CreateMediaContentOffer->MediaContentDescriptionImpl::AddCodecs
SessionDescription::AddContent->MediaSessionDescriptionFactory::AddTransportOffer
6.JsepSessionDescription::Initialize(SessionDescription* description)[Offer SDP已创建]
7.WebRtcSessionDescriptionFactory::PostCreateSessionDescriptionSucceeded->
Post(this, MSG_CREATE_SESSIONDESCRIPTION_SUCCESS)[生成SDP已成功]->
CreateSessionDescriptionObserver::OnSuccess(SessionDescriptionInterface* desc)[回调给用户]
|OnFailure(const std::string& error)[错误返回]
设置本端SDP
注:用户主动调用PeerConnection::CreateOffer 通过回调OnSuccess拿到了SDP对象,之后需要设置本端SDP
1.PeerConnection::SetLocalDescription->
2.WebRtcSession::SetLocalDescription->WebRtcSession::ValidateSessionDescription[合法性校验]->
VerifyCrypto-ValidateBundleSettings
3.Post(MSG_SET_SESSIONDESCRIPTION_SUCCESS)[设置SDP已成功]->
SetSessionDescriptionObserver::OnSuccess()[回调给用户]
4.WebRtcSession::MaybeStartGathering->TransportController::MaybeStartGathering->
5.P2PTransportChannel::MaybeStartGathering
Offer对象转换成SDP字符串(需要发给对端)
1.JsepSessionDescription::ToString
2.SdpSerialize[webrtcsdp.cc]->
接收对端SDP
1.CreateSessionDescription(const std::string& sdp)->JsepSessionDescription::Initialize->
2.SdpDeserialize->ParseMediaDescription->
3.ParseContentDescription->ParseContent->
4.以数据通道为例AddSctpDataCodec->MediaContentDescriptionImpl::AddCodec[google-sctp-data 90000 ]
5.SessionDescription::AddContent
const char NS_JINGLE_RTP[] = "urn:xmpp:jingle:apps:rtp:1";
sctp 的data 对应得ContentInfos中的type必须为NS_JINGLE_DRAFT_SCTP
const char NS_JINGLE_DRAFT_SCTP[] = "google:jingle:sctp";
SCTP 去掉DTLS 出现如下错误:
(channel.cc:2195): Setting local data description
(channel.cc:867): Failure in SetLocalContent with action 2
(webrtcsession.cc:327): Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote data description recv parameters..
(conductor.cc:104): OnFailure Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote data description recv parameters..
将bool IsDtlsSctp(const std::string& protocol) {
// This intentionally excludes "SCTP" and "SCTP/DTLS".
return protocol.find(cricket::kMediaProtocolDtlsSctp) != std::string::npos;
//return protocol.find(cricket::kMediaProtocolSctp) != std::string::npos;//add by sjz
}
将kMediaProtocolDtlsSctp改为kMediaProtocolSctp
a=sctpmap:5000 webrtc-datachannel 1024
========================sdp================================
local:
v=0
o=- 8900646765147570832 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS stream_label
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 127 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:qiC7
a=ice-pwd:97xpdM9B0N8ox6jbS2PI+0G7
a=fingerprint:sha-256 A8:C7:B8:88:DE:B9:F6:70:A3:6E:9D:C7:FD:41:4C:1E:61:9F:B7:E1:92:5F:0A:4C:48:FD:9E:D6:B7:B2:C9:55
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:127 red/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:3486011156 cname:RS5piJcFSbw2XOoP
a=ssrc:3486011156 msid:stream_label audio_label
a=ssrc:3486011156 mslabel:stream_label
a=ssrc:3486011156 label:audio_label
m=video 9 UDP/TLS/RTP/SAVPF 100 101 116 117 96 97 98
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:qiC7
a=ice-pwd:97xpdM9B0N8ox6jbS2PI+0G7
a=fingerprint:sha-256 A8:C7:B8:88:DE:B9:F6:70:A3:6E:9D:C7:FD:41:4C:1E:61:9F:B7:E1:92:5F:0A:4C:48:FD:9E:D6:B7:B2:C9:55
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtpmap:101 VP9/90000
a=rtcp-fb:101 ccm fir
a=rtcp-fb:101 nack
a=rtcp-fb:101 nack pli
a=rtcp-fb:101 goog-remb
a=rtcp-fb:101 transport-cc
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=101
a=rtpmap:98 rtx/90000
a=fmtp:98 apt=116
a=ssrc-group:FID 3678723866 3268471119
a=ssrc:3678723866 cname:RS5piJcFSbw2XOoP
a=ssrc:3678723866 msid:stream_label video_label
a=ssrc:3678723866 mslabel:stream_label
a=ssrc:3678723866 label:video_label
a=ssrc:3268471119 cname:RS5piJcFSbw2XOoP
a=ssrc:3268471119 msid:stream_label video_label
a=ssrc:3268471119 mslabel:stream_label
a=ssrc:3268471119 label:video_label
{
"candidate" : "candidate:430735571 1 udp 2122260223 192.168.1.102 59637 typ host generation 0 ufrag pa+u network-id 3 network-cost 50",
"sdpMLineIndex" : 0,
"sdpMid" : "audio"
}
remote:
v=0
o=- 5895813448736697051 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 127 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:1MuA
a=ice-pwd:YAf0BbrY4nSGW5iQ3faC1MVC
a=fingerprint:sha-256 C3:A8:1B:82:A4:2C:93:4C:F6:67:9E:E4:F3:00:7B:D1:BB:56:4D:D4:C1:9E:34:E7:82:73:D9:D0:AF:02:D2:D2
a=setup:active
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:127 red/8000
a=rtpmap:126 telephone-event/8000
m=video 9 UDP/TLS/RTP/SAVPF 100 101 116 117 96 97 98
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:1MuA
a=ice-pwd:YAf0BbrY4nSGW5iQ3faC1MVC
a=fingerprint:sha-256 C3:A8:1B:82:A4:2C:93:4C:F6:67:9E:E4:F3:00:7B:D1:BB:56:4D:D4:C1:9E:34:E7:82:73:D9:D0:AF:02:D2:D2
a=setup:active
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=recvonly
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtpmap:101 VP9/90000
a=rtcp-fb:101 ccm fir
a=rtcp-fb:101 nack
a=rtcp-fb:101 nack pli
a=rtcp-fb:101 goog-remb
a=rtcp-fb:101 transport-cc
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=101
a=rtpmap:98 rtx/90000
a=fmtp:98 apt=116
============auido and data=============
v=0
o=- 1390922805343918830 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio data
a=msid-semantic: WMS
m=audio 9 RTP/AVPF 111 8
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:ShBp
a=ice-pwd:a7DnyIUa0qvcB4yEmzOZBC3c
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:8 PCMA/8000
m=application 9 SCTP 5000
c=IN IP4 0.0.0.0
b=AS:30
a=ice-ufrag:ShBp
a=ice-pwd:a7DnyIUa0qvcB4yEmzOZBC3c
a=mid:data
a=sctpmap:5000 webrtc-datachannel 1024
内容来自用户分享和网络整理，不保证内容的准确性，如有侵权内容，可联系管理员处理 
相关文章推荐
- WebRTC 初体验
- WebRTC 简介
- WebRTC起航篇，写在最前面的话
- SimplewebRTC使用指南
- simpleWebRTC：一种简单的开源网页音视频聊天室解决方案（一）
- 在ubuntu上编译WebRTC Android版本
- WebRTC现状及实现概要
- WebRTC 初览(1) 概述
- WebRTC 初览(2) getUserMedia 介绍一
- WebRTC 初览(2) getUserMedia 介绍二
- WebRTC环境搭建
- 高性能stun服务器搭建
- webrtc源码下载
- WebCallServer WebRTC with Asterisk
- Asterisk 11的Web RTC功能
- WebRTC手记初探——（编程实现过程）
- 最简单的WebRTC示例
- 搭建webrtc服务器(ubuntu14.04)
- mac 下 webrtc 实现入门