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

[Eric Wieling]

We experience the same thing.  The solution we use is to not change codecs in the middle of a call.   I assumed it was an issue with our upstream.

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Larry Moore
> Sent: Thursday, June 16, 2011 10:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] No audio after a reinvite changing codec
>
> 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> 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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