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

[Alex Balashov]

On Tue, 3 Jul 2007, bilal ghayyad wrote:

> 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.

-- Alex

--
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : +1-678-954-0670
Direct : +1-678-954-0671