What's missing in Answer SDP (From web browser to android device)
Question
I've customized Apprtc project. i can call from an user and other user can answer call or reject call
When I call from android to web browser, I can't See video source of web browser in android device but I can see video source of android in web browser only.
Web browser version: Chrome 58 (Desktop version) Android version: Marshmallow
Offer SDP: (From Android)
v=0 o=- 7916385280226465055 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS ARDAMS___
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9 102 0 8 105 13 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:xKDP
a=ice-pwd:/hAtH4MAzGA/If6Fn+sT6Okj
a=ice-options:renomination
a=fingerprint: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=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:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:1281015102 cname:wYjcft96aVDGkQzC
a=ssrc:1281015102 msid:ARDAMS___ ARDAMSa0
a=ssrc:1281015102 mslabel:ARDAMS___
a=ssrc:1281015102 label:ARDAMSa0
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:xKDP
a=ice-pwd:/hAtH4MAzGA/If6Fn+sT6Okj
a=ice-options:renomination
a=fingerprint: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=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=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 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 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
Answer SDP: (From Web Browser)
v=0
o=mozilla...THIS_IS_SDPARTA-52.0.2 6548308332703463210 0 IN IP4 0.0.0.0
s=-
t=0 0
a=fingerprint: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 audio video
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 9 UDP/TLS/RTP/SAVPF 111 126
c=IN IP4 0.0.0.0
a=recvonly
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:111 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:126 0-15
a=ice-pwd:8a4fad1c837809d3ee952922dbe2b927
a=ice-ufrag:ab799d79
a=mid:audio
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtpmap:126 telephone-event/8000/1
a=setup:active
a=ssrc:2269112214 cname:{b1e7d024-d327-4788-a5b1-a1b8291b5c8d}
m=video 9 UDP/TLS/RTP/SAVPF 100
c=IN 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:video
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 goog-remb
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=setup:active
a=ssrc:1613714278 cname:{b1e7d024-d327-4788-a5b1-a1b8291b5c8d}
In peerconnection.cc current_tracks variable Not be filled:
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()) {
3 answers
solution1
1 2017-05-10 10:20:41
By looking into your answer SDP, it is not carrying any stream/track.
Suspected issue could be, you are not adding the stream before creating the answer in the browser.
You can check the PeerConnection API calls by opening chrome://webrtc-internals/
PeerConnection API calls should be as following in the browser/answered side
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,
}
});
}
So your Answer SDP should contain a=sendonly lines instead of a=recvonly .
solution2
1 ACCPTED 2017-05-10 10:33:52
Your browser SDP has a=recvonly attribute which means local stream is not added to your Peerconnection. If your browser is sending audio/video track to remote and wants to receive remote streams then it should have a=sendrec in AnswerSDP.
solution3
0 2017-09-26 08:04:28
Expanding on other answers: you should send your connect signal only after making sure your local stream has been fetched and added to your RTCPeerConnection.
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);
Where signal_init is your signaling/connection callback.
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