[OpenSIPS-Users] Audio problem

[Justin L]

Here is the INVITE:

INVITE sip:13101234567 at ask00-rvn SIP/2.0
Record-Route: <sip:10.1.3.130;lr;ftag=c020195b;did=d08.3a8259b2>
Via: SIP/2.0/UDP 10.1.3.130;branch=z9hG4bK88c5.4ae45bf5.0
Via: SIP/2.0/UDP 172.16.100.159:21874
;received=172.16.100.159;branch=z9hG4bK-d87543-2376e4785757b07b-1--d87543-;rport=21874
Max-Forwards: 69
Contact: <sip:2000 at 172.16.100.159:21874>
To: "13101234567"<sip:13101234567 at ask00-rvn>
From: "20000"<sip:2000 at ask00-rvn>;tag=c020195b
Call-ID: NDg4Y2Y0ZWU5MGM4NjhiNWVlZGNiZTc1ZGQxMjlhYzc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 529

v=0
o=- 8 2 IN IP4 172.16.100.159
s=CounterPath X-Lite 3.0
c=IN IP4 172.16.100.159
t=0 0
m=audio 39148 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 3 : 5AkMoAfO yRnFlRIn 172.16.100.159 39148
a=alt:2 2 : 7PbWVKqn VccqHBD1 192.168.2.59 39148
a=alt:3 1 : TXSbExav /8BXXCL+ 192.168.176.152 39148
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv


2009/10/21 Raúl Alexis Betancor Santana <rabs at dimension-virtual.com>

> On Wednesday 21 October 2009 23:13:36 Justin L wrote:
> > Hi,
> >
> > I have a question related to my load balancing configuration of opensips.
> >
> > I have an X-Lite softphone that connects to Opensips server, which
> > transfers the INVITE request to one of the asterisk boxes.
> > All of them are behind firewall on the same network. Then asterisk calls
> to
> > my cell phone through the voip provider.
> >
> > The SIP balancing works fine and I get the call, but there is no audio.
> The
> > firewall should be configured correctly to transfer the SIP and RTP
> ports.
> >
> > Since I just started to use opensips it sounds to me like a very basic
> > problem, that many people probably have faced.
> > Could you please recommend me a  way to troubleshoot this issue?
> >
> > Thanks a lot,
> >
> > Justin.
>
> Some SIP trace would be nice to begin ...
>
> --
> Raúl Alexis Betancor Santana
> Dimensión Virtual
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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