[Freeswitch-users] JAVA & WebRTC & JAIN SIP - no WebSockets

[Michael Jerris]

you need to tell freeswitch to send a webrtc compatible SDP.

https://wiki.freeswitch.org/wiki/Variable_media_webrtc <https://wiki.freeswitch.org/wiki/Variable_media_webrtc>


> On Mar 5, 2015, at 3:51 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
> 
> Hi,
> 
> I've made a simple Android Java application utilizing JAIN SIP, webrtc.org <http://webrtc.org/> android library and connected to FreeSwitch via UDP.
> 
> 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.
> 
> When I call `peerConnection.setRemoteDescription(new SDPObserver(), sdp);` in my Android Application with the SDP from FreeSwitch I get:
> 
> "onSetFailure Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
> 
> At the same time the calls between Chrome/Firefox(http://tryit.jssip.net/ <http://tryit.jssip.net/>) and SIP-phone (e.g. linphone) greatly managed by FreeSwitch and I have pure audio flow.
> 
> Here is initial SDP from Chrome (http://tryit.jssip.net/ <http://tryit.jssip.net/>):
> 
> v=0
> o=- 6887715720880489867 2 IN IP4 127.0.0.1
> s=-
> t=0 0
> a=group:BUNDLE audio video
> a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
> m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126
> c=IN IP4 192.168.122.1
> a=rtcp:38359 IN IP4 192.168.122.1
> a=candidate:4062413514 <tel:4062413514> 1 udp 2122260223 <tel:2122260223> 192.168.122.1 38359 typ host generation 0
> .......
> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0
> a=ice-ufrag:bwrCv9yS8rCY12Az
> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
> a=ice-options:google-ice
> a=fingerprint: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
> 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 <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
> a=sendrecv
> a=rtcp-mux
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> 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:126 telephone-event/8000
> a=maxptime:60
> a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB
> a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU 4e8f212e-746a-47bb-bc62-4a42d4e9e84e
> a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
> a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e
> m=video 38359 RTP/SAVPF 100 116 117 96
> c=IN IP4 192.168.122.1
> a=rtcp:38359 IN IP4 192.168.122.1
> a=candidate:4062413514 <tel:4062413514> 1 udp 2122260223 <tel:2122260223> 192.168.122.1 38359 typ host generation 0
> ............
> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0
> a=ice-ufrag:bwrCv9yS8rCY12Az
> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
> a=ice-options:google-ice
> a=fingerprint: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
> 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 <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
> a=recvonly
> a=rtcp-mux
> 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=rtpmap:116 red/90000
> a=rtpmap:117 ulpfec/90000
> a=rtpmap:96 rtx/90000
> a=fmtp:96 apt=100
> 
> 
> Here is SDP received from FreeSwitch in JAIN SIP via UDP:
> 
> v=0
> o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253
> s=FreeSWITCH
> c=IN IP4 192.168.131.253
> t=0 0
> m=audio 16390 RTP/AVP 111 0 8 101 13
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> m=video 16388 RTP/AVP 100
> a=rtpmap:100 VP8/90000
> 
> 
> 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. 
> 
> 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?
> 
> Regards,
> Oleg
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