Hello all,
Newbie here.
I’m testing an implementation of a SIP B2BUA, and I need a SIP server acting as a proxy/registrar in order to test against. I was originally thinking that Asterisk would be a good product to use to test against, but upon digging further, I understand that Asterisk itself is a B2BUA. Would this render it inappropriate to use as a test platform? I’m not so much interested in performance testing as I am in being able to generate a variety of SIP call flows (scenarios). I just don’t understand yet if Asterisk will do what I want it to.
Basically, I want to be able to have SIP phones call through the B2BUA under test and register with the Asterisk server, then undergo various scenarios, such as call forwarding, hold, transfer, conferencing, etc. I don’t particularly want the media generated by the SIP phones to go through the Asterisk, rather, I’d prefer the media to go directly from one SIP phone to the other (as is standard in SIP phone calls).
I like the idea of having the SIP server being under my control so that I can (a) take charge of the call flows myself, and (b) capture packet traffic and extract call flows for documentation. Using Asterisk would certainly be a lot simpler and cheaper than trying to get a real softswitch into our lab.
Anyone have any tips for me?
Thanks
Jeff