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
<[email protected] <mailto:[email protected]>> 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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