[Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise

[Andrew Pogrebennyk]

Hi,
this works for us so you have provide some logs or do you have any idea
what exactly is not working?

For example, if i try to call from jssip to peer or legacy client, the
offer from jssip webrtc client has RTP/SAVPF:
> Apr 15 15:13:23 sp1 rtpengine[9260]: [09b5ibtgmeu8v24713iv] Dump for 'offer' from 127.0.0.1:37622: { "sdp": "v=0#015#012o=mozilla...THIS_IS_SDPARTA-37.0.1 4135386994961230335 0 IN IP4 0.0.0.0#015#012s=-#015#012t=0 0#015#012a=sendrecv#015#012a=fingerprint:sha-256 AE:B8:19:A2:CD:A7:E6:C9:91:EA:5A:2F:AD:0B:2B:1A:E7:EF:C9:48:33:3F:3D:DC:85:2B:85:C4:A9:46:F9:8A#015#012a=group:BUNDLE sdparta_0 sdparta_1#015#012a=ice-options:trickle#015#012m=audio 9 RTP/SAVPF 109 9 0 8#015#012c=IN IP4 0.0.0.0#015#012a=candidate:0 1 UDP 2122252543 10.15.20.121 47502 typ host#015#012a=candidate:2 1 UDP 2122055935 172.17.42.1 58678 typ host#015#012a=c ...
but the server changes it to RTP/AVP:
> Apr 15 15:13:23 sp1 rtpengine[9260]: [09b5ibtgmeu8v24713iv] Response dump for 'offer' to 127.0.0.1:37622: { "sdp": "v=0#015#012o=mozilla...THIS_IS_SDPARTA-37.0.1 4135386994961230335 0 IN IP4 10.15.20.185#015#012s=-#015#012t=0 0#015#012m=audio 30538 RTP/AVP 109 9 0 8#015#012c=IN IP4 10.15.20.185#015#012a=rtpmap:109 opus/48000/2#015#012a=rtpmap:9 G722/8000/1#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=ssrc:4069823830 cname:{d6e19c01-43b4-4fa8-b841-7cc82c0e367d}#015#012a=sendrecv#015#012a=rtcp:30539#015#012m=video 30560 RTP/AVP 120 126 97#015#012c=IN IP4 10.15.20.185#015#012a=fmtp:120 max-fs=12288;max-fr=60#015#012a=fmtp:126 profi ...

In the parameters of the offer in /var/log/ngcp/rtp.log you will see
also the flag "transport-protocol": "RTP/AVP", which is telling
rtpengine to do that protocol change, provided that you have set the
transport_protocol preference on the asterisk peer correctly.

Try to read the log and find your configuration error and if you can't
please provide the output from:
select * from peer_preferences where attribute="transport_protocol";
/var/log/ngcp/rtp.log (with log_level=7 in
/etc/default/ngcp-rtpengine-daemon)
and /var/log/ngcp/kamailio-proxy.log

On 04/15/2015 06:40 AM, H Yavari wrote:
> Hi,
> 
> I can configure RTP setting for peers but this is not work for me. For
> example I have an asterisk as a peer and I selected RTP/AVP. but SDP in
> calls from webrtc clients to peer is : UDP/TLS/RTP/SAVP. So asterisk
> reject the calls.
> 
> 
> But about dynamically:
> why I can't make a call between sip extensions and webrtc clients that
> are registered on ngcp? I found in log that SDP for callee doesn't
> change automatically.

as i said in the previous email, if callee is on the websocket the proxy
tell rtpengine to transcode to "transport-protocol":
"UDP/TLS/RTP/SAVPF", even if the "transparent" is selected in the
preference.

Andrew