[Asterisk-Users] AArgh, * and the 7960

[Rich Adamson]

> > 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.

I might add to Vic's comments that simply signing up for an FWD IAX account
is enough for testing in most cases. They provide a consistent source of
audio in the forms of a milliwatt generator, data-time annoucements, and
other automated sources of audio to generate the rtp stream. Some of those
sources may have other issues, but they are sufficiently stable to observe
sequence numbers, timestamps, etc.

It is a royal pain in the butt to manually walk through 2,000 packets
calculating timestamp differences, inspecting sequence numbers, etc. I'm
in the process of writing a small app to read the ethereal packet capture
files and do that stuff on request.

Rich