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

Ewgeny evoip at ukr.net
Thu Jul 11 20:29:25 CEST 2013

Hi !

I've configured Oversip and sipml5 with my existing Opensips server 
according to that manual: 
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>
<sip:dafe310c04 at xx.xx.xx.xx:10080;transport=ws;lr;ovid=f99f817f>
Via: SIP/2.0/UDP 

Via: SIP/2.0/WS 
From: "username"<sip:username at xx.xx.xx.xx>;tag=bpfsf8BOnkTUOhyGBTo9
To: <sip:380522308888 at xx.xx.xx.xx>
"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>

o=- 1653835076 2 IN IP4
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,

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