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

[Michael Jerris]

Always nice to hear that we are magic!  

> On Mar 6, 2015, at 5:05 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
> 
> thank you very much Michael, it magically works.
> 
> On Thu, Mar 5, 2015 at 6:51 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> 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 <mailto: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?

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