[Kamailio-Users] basic SIP forwarding with Asterisk

[Klaus Darilion]

Jeff Brower schrieb:
> Klaus-
> 
>> So you want to do transcoding in rtpproxy using a DSP card? I do not
>> know - better ask on the rtpproxy mailing list (or Maxim directly - I
>> think he has a non-open source solution).
> 
> Ya we have -- and it works, no problem.  We've tested already with Kamailio + rtpproxy.
> 
>> Anyway - why not do the transcoding in Asterisk?
> 
> Because Asterisk is too limited.  It can't do enough channels for G729, and doesn't have good options for codecs like
> EVRC and GSM-AMR.
> 
> But anyway my question is about SIP with Kamailio + Asterisk, not RTP.  Is there a way that Kamailio can "pass thru"
> SIP messages from Asterisk?  Or does each call have to be relayed; i.e Asterisk sets up a call to Kamailio, then
> Kamalio sets up a call to the endpoint?

What is the difference between "pass thru" and relaying? Kamailio is a 
proxy, that means it receives a SIP message from somebody and sends it 
(slighty modified) to somebody. This forwarding can be done stateless or 
transaction statefull.

Unless you use the dialog module, Kamailio does not care about "set-up" 
calls, so if you have 100000000 millions if calls set up, Kamailio is 
idle as it only cares about transactions (INVITE, BYE ...), not about 
ongoing calls.

regards
klaus