AW: SIP and H323

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Hi Megan

We faced a similar problem some times ago.
As you did, we also found out, asterisk is not suitable as a h323/sip
gateway (poor voice quality, h323 implementation very unreliable and lacking
a lot of basic features).
We also evaluated yate but decided against it. It misses some very important
functionality/features regarding codecs. Overall yate looks like a project
in it's startup phase - yet not ready for a productive environment (but with
the potential to grow). There are also some more, comercial products but
none of them were able to fullfill our requirements (like h235 or any codec
to any codec translation).
So we ended up in developing our own h323/sip signaling gateway that
interoperates with gnugk and ser.

Cheers
Frank


> -----Ursprüngliche Nachricht-----
> Von: openh323gk-users-admin@xxxxxxxxxxxxxxxxxxxxx 
> [mailto:openh323gk-users-admin@xxxxxxxxxxxxxxxxxxxxx] Im 
> Auftrag von Megan Willigs
> Gesendet: Montag, 27. Juni 2005 02:58
> An: openh323gk-users@xxxxxxxxxxxxxxxxxxxxx
> Betreff:  SIP and H323
> 
> Hi everybody.
> I'm working in a VoIP enterprise, our network was developed 
> over H323. 
> At the moment we have a few SIP phones, those are accessing 
> to our H323 network through an Asterisk server in order to 
> interconnect both networks.
> We want to migrate from Asterisk to SER, and perform the 
> translation (between SIP and H323) with Yate, but Yate 
> doesn't work as the documentation refers.  Anyone have 
> experience configuring Yate to perform the transation? 
> (betwen SIP and H323). Is there another option to perform 
> this translation?.
> 
> Thanks in advance.
> Megan Willigs
> 
> 
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> 





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