[Asterisk-Users] SIP <->h.323

[Bernie Hoeneisen]

Hi Ryan!

Interesting what experience you have made in this issue.
We have setup the alternative channel for H.323 (the * built in
chan_h323), and we are now in a testing phase.


I was wondering (in case no transcoding is needed), how your setup treats
the RTP streams. Do the RTP streams go end-to-end or always via Asterisk?

Another question I'd be interested in: Have you also gained some
experience with bridging _video_ calls between H.323 and SIP?


cheers,
 Bernie

PS: I'd be glad, if I also could get the relevant config files from you.


On Fri, 13 Aug 2004, Ryan Wilkins wrote:

> Yes, it can.. I'm doing it at my home.  My current setup is
> Asterisk-1.0-RC2 using the oh323 driver.  I have a SIP connection to
> Broadvoice talking to Asterisk.  I have a e-tel (now Qtelnet) H.323 VoIP
> telephone adapter as my end point talking to Asterisk.
>
> For processing sake, you may want to keep your codec the same all the way
> through.  Originally I ran G.711u on the SIP connection and G.711a on the
> H.323 connection.  It worked just fine but the logs always said something
> about transcoding between u-law and a-law.  I reset the H.323 link to
> G.711u and now it says nothing about transcoding.  In theory you would
> lose a bit of audio quality in the translation process.  In reality I
> don't really know.
>
> email me privately if you want a sample config.
>
> Ryan Wilkins
>
>
>  On Fri, 13 Aug 2004, Yiannis Costopoulos, Web2Net Solutions Ltd. wrote:
>
> > 	is there a definite answer if asterisk can pass calls between SIP
> > and h.323 protocols?