On Friday 23 October 2009 16:01:41 Jeff Brower wrote:
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.
It's possible to know witch DSP card are you using ?
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?
Kamailio as SIP proxy, doesn't "setup a call", it just route SIP messages,
so
by definition it works as "SIP-Passthrought", so what is the problem you
have ?
Well hopefully no problem and I'm a worry-wart :-) We're very sensitive to call
setup and tear-down times...
especially we're concerned about running Asterisk and Kamailio on the same server.
Doing RTP stuff with a DSP card
gives a huge increase in call capacity, but if we can't maintain fast setup and
tear-down rates, then we defeat the
purpose.
-Jeff
Raúl Alexis Betancor Santana
Dimensión Virtual
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users