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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
