[Kamailio-Users] basic SIP forwarding with Asterisk

[Jeff Brower]

Raúl Alexis-

> 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 ?

  http://www.signalogic.com/sigc5561_ptmc.htm

in PCI and PCIe formats.

>> 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
>
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