[Asterisk-Users] AArgh, * and the 7960

[Vic Cross]

On Thu, 20 May 2004, Iain Stevenson wrote:

> The Catch 22 is I don't have access to access to a source of repeatable
> (ie recorded) content accessed through IAX.  That would help in
> producing traces for the ATA and 7960 for comparison.

The payload (i.e. audio) of the RTP stream is not relevant, at least in my 
experience.  All the information you need is in the RTP header -- sequence 
numbers (not a problem, that I've seen) and timestamps.

If you have two SIP phones, a FWD account and an IAXtel account, you have 
all you need to test SIP-IAX2-SIP.  From one SIP phone, use your FWD 
account to call your IAXtel number, and pick up the incoming call on your 
other SIP phone.  To avoid looping issues (multiple hops through your * 
box), make the source (FWD) end a SIP client defined directly to FWD, the 
IAXtel end your * box, and hang your destination SIP client off *.  
Subject to the bandwidth you have available upstream, this should be an 
adequate test and allow you to capture everything you need.  Capture 
everything in and out of the * box if you can, as this will give the 
greatest amount of information and good correlation between the IAX2 
traffic and the SIP traffic that goes to your SIP destination.

Hope this is helpful (and not restating the bleeding obvious)...

Cheers,
Vic Cross