[OpenSIPS-Users] WebRTC - OpenSIPS and OverSIP, no audio.

[Vlad Paiu]

Hello,

What are the two end-parties user agents ? Are they both sipML5 ? Or at 
least do they both support WEBRTC ?
Could you please provide a full SIP trace with the entire call flow ?

Best Regards,
Vlad

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 07/11/2013 09:29 PM, Ewgeny wrote:
> Hi !
>
> I've configured Oversip and sipml5 with my existing Opensips server 
> according to that manual: 
> http://www.opensips.org/Documentation/Tutorials-WebSocket.
> Registrations are routed from WS to Opensips - no problems. Also SIP 
> INVITE's are working - means calls are routed from ws 10080 -> oversip 
> 6789 -> 5060 opensips.
> But there are no audio at all :(
> I use Chromium version 25. And i see that browser ask for using my 
> microphone and speaker devices.
>
> INVITE sip:380522308888 at xx.xx.xx.xx SIP/2.0
> Record-Route: <sip:xx.xx.xx.xx:6789;transport=udp;lr;ovid=f99f817f>
> Record-Route: 
> <sip:dafe310c04 at xx.xx.xx.xx:10080;transport=ws;lr;ovid=f99f817f>
> Via: SIP/2.0/UDP 
> xx.xx.xx.xx:6789;branch=z9hG4bK49f1d2e5baecf6fed6568e1f90b8525f764e6ebe;rport 
>
> Via: SIP/2.0/WS 
> df7jal23ls0d.invalid;branch=z9hG4bKXG9SxHTImCb9AowUIP5XvJK65BewNTDs;rport
> From: "username"<sip:username at xx.xx.xx.xx>;tag=bpfsf8BOnkTUOhyGBTo9
> To: <sip:380522308888 at xx.xx.xx.xx>
> Contact: 
> "username"<sip:username at df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=username;ha1=502c7613944a73ceb9f07e56b1111fa5;+g.oma.sip-im;+sip.ice;language="en,fr" 
>
> Call-ID: 7933eaf8-247d-25a3-c358-5afb92d66262
> CSeq: 14347 INVITE
> Content-Type: application/sdp
> Content-Length: 2277
> Max-Forwards:  9
> Proxy-Authorization: Digest 
> username="username",realm="xx.xx.xx.xx",nonce="51deec76a1dbbe823d92e",uri="sip:380522308888 at xx.xx.xx.xx",response="9a26ec9b5b2afafd4c",algorithm=MD5 
>
> User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
> Organization: Doubango Telecom
> P-Asserted-Identity: <sip:username at xx.xx.xx.xx>
>
> v=0
> o=- 1653835076 2 IN IP4 127.0.0.1
> s=Doubango Telecom - chrome
> t=0 0
> a=group:BUNDLE audio
> a=msid-semantic: WMS zoWEYFAmSYL6shOiAg3B17hWgT0xbyXQPjjL
> m=audio 36315 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126
> c=IN IP4 188.xxx.xxx.190
> a=rtcp:36315 IN IP4 188.xxx.xxx.190
> a=candidate:352176934 1 u
>
>
>
>
> Why it is no audio ?
> I'm still don't understand through which ports are RTP goes on WS ???
>
>
>
>
>
>
>
> Best Regards,
> Ewgeny
>
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