[Kamailio-Users] basic SIP forwarding with Asterisk

[Klaus Darilion]

Hi Jeff!

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).

Anyway - why not do the transcoding in Asterisk?

regards
klaus

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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