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?
I know that we can get it to work one way or another, but I'm worried about channel capacity if both Asterisk and Kamailio run on the same server. "Duplicating" calls does not seem efficient.
-Jeff
Jeff Brower schrieb:
All-
Can we use Asterisk combined with Kamailio as follows:
__________ ___________ | | | |SIP ___| |___ SIP ___| Kamailio |___ SIP | | | rtpproxy | | Asterisk | | | | | | | | | RTP ___| |___ RTP ___| DSP card |___ RTP (G711) |__________| (G711) |___________| (G729, G723, GSM-AMR, EVRC)
We've already implemented an rtpproxy interface to the DSP card, which has its own GbE port. Our question is whether we can we perform some type of basic SIP forwarding or "SIP pass-thru", but still invoke rtpproxy for call setup and tear-down and/or when media attributes for the call change?
We're getting a lot of requests from customers who -- for whatever reasons -- need to use or continue to use Asterisk, but need to also add transcoding, ec, encryption, and other compute-intensive requirements that Asterisk doesn't support (or at least doesn't support at higher capacity or without going unstable).
Thanks.
-Jeff
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