[OpenSIPS-Users] Audio problem

[Bogdan-Andrei Iancu]

Hi Justin,

a trace means all SIP messages from that call (not only the INVITE) :).

Also, "audio problem" means there is not audio at all or means you have 
one way audio ?

Regards,
Bogdan

Justin L wrote:
> 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 
> <http://sip:2000@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 
> <mailto: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
>
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