[asterisk-users] Determining the used codec for the IP Trunk (SIP Trunk)

[bilal ghayyad]

Dear Alex;

I am asking about:

What is the configuration that I can do it to let the
traffic between the two Asterisk PBX and another IP BX
to be g729 or G711 or g723?

In other words, how can I let the ued codec for the IP
Trunk between my Asterisk and the other IP PBX to be
g729 and not g711? Ofcourse, I am assuming that the
other side also supporting g729.

Regards
Bilal 

> Where I determine the codec to be used for the SIP
Trunk (between 
> Asterik and another SIP softswitch)?

   Are you asking positively how to determine which
codec is being 
negotiated between those two elements, or,
normatively, which one is
best to use?

   If the former question, you can look at the SDP
(Session Description
Protocol) payload in the INVITEs (and other messages
part of the INVITE
transaction).  Find the 'rtpmap' lines.  They look
like this:

         a=rtpmap:18 G729/8000

   There may be multiple such lines indicating that
endpoint's support
for all of them.  If so, the only way to determine
which one is
 actually
going to be used is to look at the RTP stream with a
protocol analyser.
It should be able to tell you.  Wireshark/Ethereal
certainly can.

   If you're asking which codec should be used, it
depends on the
 desired 
application, whether the trunk is running over a LAN,
WAN, or over the 
public Internet, whether QoS is involved, etc.



 
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