problem with SDP when adding datachannel.

[Tore Lading]

Hi All,

I have created two peers in Android- works fine.

But as soon as I add in a datachannel on the offer side like this 

DataChannel.Init dcInit = new DataChannel.Init();
dataChannel = mPeerConnection.createDataChannel("dataChannel", dcInit);
customDataChannelObserver.setDataChannel(dataChannel);
dataChannel.registerObserver(customDataChannelObserver);
debugToast("Camera created datachannel");

I get a onSetFailure with the very little saying error:

SessionDescription is NULL. 

when ever I try to set the local descriptor when creating the offer. The session descriptor is definetly not null, but looks like this below....

Seems there is a bug whenever creating a SDP with a datachannel... anyone has some ideas or is this a bug to be fixed ?

SDP with datachannel:

v=0
o=- 5683493735642657276 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1 2
a=extmap-allow-mixed
a=msid-semantic: WMS WebRTC-stream-audio WebRTC-stream-video
m=audio 9 RTP/AVPF 111 63 103 104 9 102 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:2FHB
a=ice-pwd:YNZx91ajAZCZ9VZbtl/HN6dz
a=ice-options:trickle renomination
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:WebRTC-stream-audio 101
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
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:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:531177122 cname:kl+UO1igPDu+qoJb
a=ssrc:531177122 msid:WebRTC-stream-audio 101
a=ssrc:531177122 mslabel:WebRTC-stream-audio
a=ssrc:531177122 label:101
m=video 9 RTP/AVPF 96 97 98 99 35 36 100 101 127 124 123 122 125
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:2FHB
a=ice-pwd:YNZx91ajAZCZ9VZbtl/HN6dz
a=ice-options:trickle renomination
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:WebRTC-stream-video 100
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:35 AV1/90000
a=rtcp-fb:35 goog-remb
a=rtcp-fb:35 transport-cc
a=rtcp-fb:35 ccm fir
a=rtcp-fb:35 nack
a=rtcp-fb:35 nack pli
a=rtpmap:36 rtx/90000
a=fmtp:36 apt=35
a=rtpmap:100 H264/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640c1f
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:124 rtx/90000
a=fmtp:124 apt=127
a=rtpmap:123 red/90000
a=rtpmap:122 rtx/90000
a=fmtp:122 apt=123
a=rtpmap:125 ulpfec/90000
a=ssrc-group:FID 2120596826 2749532084
a=ssrc:2120596826 cname:kl+UO1igPDu+qoJb
a=ssrc:2120596826 msid:WebRTC-stream-video 100
a=ssrc:2120596826 mslabel:WebRTC-stream-video
a=ssrc:2120596826 label:100
a=ssrc:2749532084 cname:kl+UO1igPDu+qoJb
a=ssrc:2749532084 msid:WebRTC-stream-video 100
a=ssrc:2749532084 mslabel:WebRTC-stream-video
a=ssrc:2749532084 label:100
m=application 9 SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=ice-ufrag:2FHB
a=ice-pwd:YNZx91ajAZCZ9VZbtl/HN6dz
a=ice-options:trickle renomination
a=mid:2

[Philipp Hancke]

  m=audio 9 RTP/AVPF

indicates that you seem to disable encryption. Which then presumably leads to

  m=application 9 SCTP webrtc-datachannel

which is not supported.

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[Tore Lading]

Yes I believe this fixed tre problem, thanks

To view this discussion on the web visit https://groups.google.com/d/msgid/discuss-webrtc/CADxkKiJ9CgNSKPfh_VV231Udeb_U_5PjtWVM7QDe%3DU4XW9NS0A%40mail.gmail.com.

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Tore Lading