<div dir="ltr">Hi,<div><br></div><div>I've made a simple Android Java application utilizing JAIN SIP, <a href="http://webrtc.org" target="_blank">webrtc.org</a> android library and connected to FreeSwitch via UDP.</div><div><br></div><div>But when I send SDP from SIP/Chrome/Firefox phone to my JAIN SIP client this SDP is not managed well by FreeSwitch for establishment WebRTC PeerConnection.</div><div><br></div><div>When I call `peerConnection.setRemoteDescription(new SDPObserver(), sdp);` in my Android Application with the SDP from FreeSwitch I get:<br></div><div><div><br class="">"onSetFailure Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."<br></div></div><div><br></div><div><div>At the same time the calls between Chrome/Firefox(<a href="http://tryit.jssip.net/">http://tryit.jssip.net/</a>) and SIP-phone (e.g. linphone) greatly managed by FreeSwitch and I have pure audio flow.</div></div><div><br></div><div>Here is initial SDP from Chrome (<a href="http://tryit.jssip.net/">http://tryit.jssip.net/</a>):<br></div><div><br></div><div><div>v=0</div><div>o=- 6887715720880489867 2 IN IP4 127.0.0.1</div><div>s=-</div><div>t=0 0</div><div>a=group:BUNDLE audio video</div><div>a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU</div><div>m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126</div><div>c=IN IP4 192.168.122.1</div><div>a=rtcp:38359 IN IP4 192.168.122.1</div><div>a=candidate:<a href="tel:4062413514" value="+14062413514" target="_blank">4062413514</a> 1 udp <a href="tel:2122260223" value="+12122260223" target="_blank">2122260223</a> 192.168.122.1 38359 typ host generation 0</div><div>.......<br></div><div>a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0<br></div><div>a=ice-ufrag:bwrCv9yS8rCY12Az</div><div>a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP</div><div>a=ice-options:google-ice</div><div>a=<b>fingerprint</b>:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2</div><div>a=setup:actpass</div><div>a=mid:audio</div><div>a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level</div><div>a=extmap:3 <a href="http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time" target="_blank">http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time</a></div><div>a=sendrecv</div><div>a=rtcp-mux</div><div>a=rtpmap:111 opus/48000/2</div><div>a=fmtp:111 minptime=10</div><div>a=rtpmap:103 ISAC/16000</div><div>a=rtpmap:104 ISAC/32000</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:106 CN/32000</div><div>a=rtpmap:105 CN/16000</div><div>a=rtpmap:13 CN/8000</div><div>a=rtpmap:126 telephone-event/8000</div><div>a=maxptime:60</div><div>a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB</div><div>a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU 4e8f212e-746a-47bb-bc62-4a42d4e9e84e</div><div>a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU</div><div>a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e</div><div>m=video 38359 RTP/SAVPF 100 116 117 96</div><div>c=IN IP4 192.168.122.1</div><div>a=rtcp:38359 IN IP4 192.168.122.1</div><div>a=candidate:<a href="tel:4062413514" value="+14062413514" target="_blank">4062413514</a> 1 udp <a href="tel:2122260223" value="+12122260223" target="_blank">2122260223</a> 192.168.122.1 38359 typ host generation 0</div><div>............</div><div>a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0<br></div><div>a=ice-ufrag:bwrCv9yS8rCY12Az</div><div>a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP</div><div>a=ice-options:google-ice</div><div>a=<b>fingerprint</b>:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2</div><div>a=setup:actpass</div><div>a=mid:video</div><div>a=extmap:2 urn:ietf:params:rtp-hdrext:toffset</div><div>a=extmap:3 <a href="http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time" target="_blank">http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time</a></div><div>a=recvonly</div><div>a=rtcp-mux</div><div>a=rtpmap:100 VP8/90000</div><div>a=rtcp-fb:100 ccm fir</div><div>a=rtcp-fb:100 nack</div><div>a=rtcp-fb:100 nack pli</div><div>a=rtcp-fb:100 goog-remb</div><div>a=rtpmap:116 red/90000</div><div>a=rtpmap:117 ulpfec/90000</div><div>a=rtpmap:96 rtx/90000</div><div>a=fmtp:96 apt=100</div></div><div><br></div><div><br></div><div>Here is SDP received from FreeSwitch in JAIN SIP via UDP:</div><div><br></div><div><div>v=0</div><div>o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253</div><div>s=FreeSWITCH</div><div>c=IN IP4 192.168.131.253</div><div>t=0 0</div><div>m=audio 16390 RTP/AVP 111 0 8 101 13</div><div>a=rtpmap:111 opus/48000/2</div><div>a=fmtp:111 minptime=10</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>m=video 16388 RTP/AVP 100</div><div>a=rtpmap:100 VP8/90000</div></div><div><br></div><div><br></div><div>I suppose that FreeSwitch wants to see WebRTC connection only on the WebSocket ports and it doesn't know that my UDP client is actually WebRTC client. </div><div><br></div><div>So I'm wondering if it possible to connect SIP client to the WebSocket port via TCP using standard SIP client and never upgrade connection to WebSocket?</div><div><div><br></div><div><div dir="ltr"><div>Regards,</div><div>Oleg</div></div></div>
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