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

[Matteo Campana]

HI list,
no idea?? :)

M.

On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana <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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