[Kamailio-Users] basic SIP forwarding with Asterisk

[Jeff Brower]

Klaus-

> 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.

Thanks for your reply.  Yes you're right... I think we just need to try it stateless and measure the performance.

-Jeff