[asterisk-users] No audio after a reinvite changing codec

[Larry Moore]

On 15/06/2011 8:15 PM, Matteo Campana wrote:
> HI list,
> no idea?? :)
>

There not much substance in the information provided for an assessment 
to be made.

I would suggest you capture the network traffic between UAC (g711) & 
Asterisk UAS ensuring the snap length is large enough to capture the 
whole packet and do the same with traffic between Asterisk UAC & 
Provider then use Wireshark and its telephony feature to analyse VoIP 
calls, check the flows, you may discover the problem this way!

Larry.

> M.
>
> On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana 
> <matteo.campana at gmail.com <mailto:matteo.campana at gmail.com>> wrote:
>
>     Hi all,
>     we have a problem with a reinvite sent by our SIP provider to
>     change audio codec due to the recognition of a fax tone.
>     After that the SIP call session has been established (INVITE and
>     200 OK) we have the following codec situation:
>
>     UAC                                        ASTERISK UAS | ASTERISK
>     UAC                              PROVIDER
>     g711 <---------------------->           g711      |       g729
>     <--------------------------->  g729
>                             rtp                                       
>                                             rtp
>
>     After a while, we have the reinvite sent by the SIP provider with
>     g711 in the SDP.
>     So asterisk need to change audio codec from g729 to g711 and
>     correctly we see on debug the following line:
>     "Oooh, we need to change our audio formats since our peer supports
>     only g729" and asterisk send back 200 OK to the provider.
>     At this point we have one way rtp audio:
>
>     UAC                                        ASTERISK UAS | ASTERISK
>     UAC                              PROVIDER
>     g711      ---------------------->           g711      |       g711
>         --------------------------->  g711
>                             rtp                                       
>                                             rtp
>
>     So the problem is that UAC does not hear audio at all.
>     Any idea?
>
>     (Asterisk version: 1.4.33.1).
>
>     Thanks in advance,
>     Matteo 
>
>
>
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