Answer SDP 中缺少什么（从网络浏览器到 android 设备）
What's missing in Answer SDP (From web browser to android device)
我已经自定义了 Apprtc 项目。我可以从一个用户呼叫，其他用户可以接听电话或拒接电话
当我从 android 调用网络浏览器时，我无法在 android 设备中看到网络浏览器的视频源，但我可以在网络浏览器中看到 android 的视频源只有.
网络浏览器版本：Chrome58（桌面版）
Android版本：棉花糖
提供 SDP：（来自 Android）
v=0
o=- 7916385280226465055 2 输入 IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 音频视频
a=msid-语义：WMS ARDAMS___
m=音频 9 UDP/TLS/RTP/SAVPF 111 103 9 102 0 8 105 13 126
c=输入 IP4 0.0.0.0
a=rtcp:9 输入 IP4 0.0.0.0
a=ice-ufrag:xKDP
a=ice-pwd:/hAtH4MAzGA/If6Fn+sT6Okj
a=ice-options:重新提名
a=指纹：sha-256
35:5A:08:8D:FA:18:41:B9:A6:E2:B4:9A:A7:EE:1E:61:CA:38:BC:5B:98:9F:D1:3E:1F:51:79:C8:F3:63:00:F8
a=设置：actpass
a=mid:音频
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-音频级别
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 传输-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:102ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126telephone-event/8000
a=ssrc:1281015102 cname:wYjcft96aVDGkQzC
a=ssrc:1281015102 msid:ARDAMS___ ARDAMSa0
a=ssrc:1281015102 mslabel:ARDAMS___
a=ssrc:1281015102 label:ARDAMSa0
m=视频 9 UDP/TLS/RTP/SAVPF 100 101 116 117 96 97 98
c=输入 IP4 0.0.0.0
a=rtcp:9 输入 IP4 0.0.0.0
a=ice-ufrag:xKDP
a=ice-pwd:/hAtH4MAzGA/If6Fn+sT6Okj
a=ice-options:重新提名
a=指纹:sha-256 35:5A:08:8D:FA:18:41:B9:A6:E2:B4:9A:A7:EE:1E:61:CA:38:BC:5B:98:9F:D1:3E:1F:51:79:C8:F3:63:00:F8
a=设置：actpass
a=mid:视频
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-方向
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm 冷杉
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 传输-cc
a=rtpmap:101 VP9/90000
a=rtcp-fb:101 ccm 冷杉
a=rtcp-fb:101 nack
a=rtcp-fb:101 nack pli
a=rtcp-fb:101 goog-remb
a=rtcp-fb:101 传输-cc
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96rtx/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 2034101263 3486873766
a=ssrc:2034101263 cname:wYjcft96aVDGkQzC
a=ssrc:2034101263 msid:ARDAMS___ ARDAMSv0
a=ssrc:2034101263 mslabel:ARDAMS___
a=ssrc:2034101263 label:ARDAMSv0
a=ssrc:3486873766 cname:wYjcft96aVDGkQzC
a=ssrc:3486873766 msid:ARDAMS___ ARDAMSv0
a=ssrc:3486873766 mslabel:ARDAMS___
a=ssrc:3486873766 label:ARDAMSv0
回答 SDP：（从 Web 浏览器）
v=0
o=mozilla...THIS_IS_SDPARTA-52.0.2 6548308332703463210 0 IN IP4 0.0.0.0
s=-
t=0 0
a=指纹:sha-256 E6:0F:6A:A6:35:E0:B3:8E:7A:0E:2E:20:A9:AB:0B:CA:1C:6D:33:6C:B6:D1:E4:2D:39:87:1E:93:4E:ED:BB:CF
a=group:BUNDLE 音频视频
a=ice-options:trickle
a=msid-语义：WMS *
m=音频 9 UDP/TLS/RTP/SAVPF 111 126
c=输入 IP4 0.0.0.0
a=recvonly
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-音频级别
a=fmtp:111 maxplaybackrate=48000;立体声=1;useinbandfec=1
a=fmtp:126 0-15
a=ice-pwd:8a4fad1c837809d3ee952922dbe2b927
a=ice-ufrag:ab799d79
a=mid:音频
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtpmap:126telephone-event/8000/1
a=设置：活动
a=ssrc:2269112214 cname:{b1e7d024-d327-4788-a5b1-a1b8291b5c8d}
m=视频 9 UDP/TLS/RTP/SAVPF 100
c=输入 IP4 0.0.0.0
a=recvonly
a=fmtp:100 max-fs=12288;max-fr=60
a=ice-pwd:8a4fad1c837809d3ee952922dbe2b927
a=ice-ufrag:ab799d79
a=mid:视频
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 ccm 冷杉
a=rtcp-fb:100 goog-remb
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=设置：活动
a=ssrc:1613714278 cname:{b1e7d024-d327-4788-a5b1-a1b8291b5c8d}
在peerconnection.cc中current_tracks变量未填写：
void PeerConnection::UpdateRemoteStreamsList(
const cricket::StreamParamsVec& streams,
bool default_track_needed,
cricket::MediaType media_type,
StreamCollection* new_streams) {
TrackInfos* current_tracks = GetRemoteTracks(media_type);
// Find removed tracks. I.e., tracks where the track id or ssrc don't match
// the new StreamParam.
auto track_it = current_tracks->begin();
while (track_it != current_tracks->end()) {
通过查看您的答案 SDP，它没有携带任何 stream/track。
怀疑的问题可能是，您在浏览器中创建答案之前没有添加流。
您可以通过打开 chrome://webrtc-internals/
检查 PeerConnection API 调用
PeerConnection API 调用在 browser/answered 端应如下所示
pc = new RTCPeerConnection({"iceServers": [{"urls": "stun:stun.l.google.com:19302"}]},
{"optional": [{"DtlsSrtpKeyAgreement": true}]
});
pc.setRemoteDescription(
new RTCSessionDescription(jsep),
function() {
console.log(' OFFER accepted ');
}, function(e) {
console.log(' OFFER Failed ', e);
});
pc.addStream(stream);
pc.createAnswer(function(answer) {
console.log('got answer', answer);
pc.setLocalDescription(answer,
function() {
console.log('set local description sucesses ');
}, function(e) {
console.log('set local description failed ', e);
});
// Send the answer to other user endpoint
}, function() {
console.log('Error: Unable to create answer');
}, {
'mandatory': {
'OfferToReceiveAudio': true,
'OfferToReceiveVideo': true,
}
});
}
所以你的答案 SDP 应该包含 a=sendonly 行而不是 a=recvonly.
您的浏览器 SDP 具有 a=recvonly 属性，这意味着本地流未添加到您的 Peerconnection。如果您的浏览器正在向远程发送 audio/video 轨道并希望接收远程流，那么它应该在 AnswerSDP 中有 a=sendrec。
扩展其他答案：只有在确保已获取本地流并将其添加到 RTCPeerConnection 后，您才应发送连接信号。
navigator.mediaDevices.getUserMedia({
audio: false, // request access to local microphone
video: true // request access to local camera
}).then(function (local_stream) {
// display preview from the local camera & microphone using local <video> MediaElement
var media_element = document.getElementById('local_video');
media_element.srcObject = local_stream;
media_element.play();
// add local camera stream to peer_connection ready to be sent to the remote peer
peer_connection.addStream(local_stream);
signal_init();
}).catch(console.log);
其中 signal_init 是您的 signaling/connection 回调。
我已经自定义了 Apprtc 项目。我可以从一个用户呼叫，其他用户可以接听电话或拒接电话
当我从 android 调用网络浏览器时，我无法在 android 设备中看到网络浏览器的视频源，但我可以在网络浏览器中看到 android 的视频源只有.
网络浏览器版本：Chrome58（桌面版） Android版本：棉花糖
提供 SDP：（来自 Android）
v=0 o=- 7916385280226465055 2 输入 IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 音频视频
a=msid-语义：WMS ARDAMS___
m=音频 9 UDP/TLS/RTP/SAVPF 111 103 9 102 0 8 105 13 126
c=输入 IP4 0.0.0.0
a=rtcp:9 输入 IP4 0.0.0.0
a=ice-ufrag:xKDP
a=ice-pwd:/hAtH4MAzGA/If6Fn+sT6Okj
a=ice-options:重新提名
a=指纹：sha-256 35:5A:08:8D:FA:18:41:B9:A6:E2:B4:9A:A7:EE:1E:61:CA:38:BC:5B:98:9F:D1:3E:1F:51:79:C8:F3:63:00:F8
a=设置：actpass
a=mid:音频
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-音频级别
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 传输-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:102ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126telephone-event/8000
a=ssrc:1281015102 cname:wYjcft96aVDGkQzC
a=ssrc:1281015102 msid:ARDAMS___ ARDAMSa0
a=ssrc:1281015102 mslabel:ARDAMS___
a=ssrc:1281015102 label:ARDAMSa0
m=视频 9 UDP/TLS/RTP/SAVPF 100 101 116 117 96 97 98
c=输入 IP4 0.0.0.0
a=rtcp:9 输入 IP4 0.0.0.0
a=ice-ufrag:xKDP
a=ice-pwd:/hAtH4MAzGA/If6Fn+sT6Okj
a=ice-options:重新提名
a=指纹:sha-256 35:5A:08:8D:FA:18:41:B9:A6:E2:B4:9A:A7:EE:1E:61:CA:38:BC:5B:98:9F:D1:3E:1F:51:79:C8:F3:63:00:F8
a=设置：actpass
a=mid:视频
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-方向
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm 冷杉
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 传输-cc
a=rtpmap:101 VP9/90000
a=rtcp-fb:101 ccm 冷杉
a=rtcp-fb:101 nack
a=rtcp-fb:101 nack pli
a=rtcp-fb:101 goog-remb
a=rtcp-fb:101 传输-cc
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96rtx/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 2034101263 3486873766
a=ssrc:2034101263 cname:wYjcft96aVDGkQzC
a=ssrc:2034101263 msid:ARDAMS___ ARDAMSv0
a=ssrc:2034101263 mslabel:ARDAMS___
a=ssrc:2034101263 label:ARDAMSv0
a=ssrc:3486873766 cname:wYjcft96aVDGkQzC
a=ssrc:3486873766 msid:ARDAMS___ ARDAMSv0
a=ssrc:3486873766 mslabel:ARDAMS___
a=ssrc:3486873766 label:ARDAMSv0
回答 SDP：（从 Web 浏览器）
v=0
o=mozilla...THIS_IS_SDPARTA-52.0.2 6548308332703463210 0 IN IP4 0.0.0.0
s=-
t=0 0
a=指纹:sha-256 E6:0F:6A:A6:35:E0:B3:8E:7A:0E:2E:20:A9:AB:0B:CA:1C:6D:33:6C:B6:D1:E4:2D:39:87:1E:93:4E:ED:BB:CF
a=group:BUNDLE 音频视频
a=ice-options:trickle
a=msid-语义：WMS *
m=音频 9 UDP/TLS/RTP/SAVPF 111 126
c=输入 IP4 0.0.0.0
a=recvonly
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-音频级别
a=fmtp:111 maxplaybackrate=48000;立体声=1;useinbandfec=1
a=fmtp:126 0-15
a=ice-pwd:8a4fad1c837809d3ee952922dbe2b927
a=ice-ufrag:ab799d79
a=mid:音频
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtpmap:126telephone-event/8000/1
a=设置：活动
a=ssrc:2269112214 cname:{b1e7d024-d327-4788-a5b1-a1b8291b5c8d}
m=视频 9 UDP/TLS/RTP/SAVPF 100
c=输入 IP4 0.0.0.0
a=recvonly
a=fmtp:100 max-fs=12288;max-fr=60
a=ice-pwd:8a4fad1c837809d3ee952922dbe2b927
a=ice-ufrag:ab799d79
a=mid:视频
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 ccm 冷杉
a=rtcp-fb:100 goog-remb
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=设置：活动
a=ssrc:1613714278 cname:{b1e7d024-d327-4788-a5b1-a1b8291b5c8d}
在peerconnection.cc中current_tracks变量未填写：
void PeerConnection::UpdateRemoteStreamsList(
const cricket::StreamParamsVec& streams,
bool default_track_needed,
cricket::MediaType media_type,
StreamCollection* new_streams) {
TrackInfos* current_tracks = GetRemoteTracks(media_type);
// Find removed tracks. I.e., tracks where the track id or ssrc don't match
// the new StreamParam.
auto track_it = current_tracks->begin();
while (track_it != current_tracks->end()) {
通过查看您的答案 SDP，它没有携带任何 stream/track。
怀疑的问题可能是，您在浏览器中创建答案之前没有添加流。
您可以通过打开 chrome://webrtc-internals/
PeerConnection API 调用在 browser/answered 端应如下所示
pc = new RTCPeerConnection({"iceServers": [{"urls": "stun:stun.l.google.com:19302"}]},
{"optional": [{"DtlsSrtpKeyAgreement": true}]
});
pc.setRemoteDescription(
new RTCSessionDescription(jsep),
function() {
console.log(' OFFER accepted ');
}, function(e) {
console.log(' OFFER Failed ', e);
});
pc.addStream(stream);
pc.createAnswer(function(answer) {
console.log('got answer', answer);
pc.setLocalDescription(answer,
function() {
console.log('set local description sucesses ');
}, function(e) {
console.log('set local description failed ', e);
});
// Send the answer to other user endpoint
}, function() {
console.log('Error: Unable to create answer');
}, {
'mandatory': {
'OfferToReceiveAudio': true,
'OfferToReceiveVideo': true,
}
});
}
所以你的答案 SDP 应该包含 a=sendonly 行而不是 a=recvonly.
您的浏览器 SDP 具有 a=recvonly 属性，这意味着本地流未添加到您的 Peerconnection。如果您的浏览器正在向远程发送 audio/video 轨道并希望接收远程流，那么它应该在 AnswerSDP 中有 a=sendrec。
扩展其他答案：只有在确保已获取本地流并将其添加到 RTCPeerConnection 后，您才应发送连接信号。
navigator.mediaDevices.getUserMedia({
audio: false, // request access to local microphone
video: true // request access to local camera
}).then(function (local_stream) {
// display preview from the local camera & microphone using local <video> MediaElement
var media_element = document.getElementById('local_video');
media_element.srcObject = local_stream;
media_element.play();
// add local camera stream to peer_connection ready to be sent to the remote peer
peer_connection.addStream(local_stream);
signal_init();
}).catch(console.log);
其中 signal_init 是您的 signaling/connection 回调。