[Asterisk-Dev] Call and video bandwidths

[John Martin]

Hi Kevin, thanks for the response and I can see how Asterisk is
constructed now. It seems a shame, however, that there is no support for
fmtp media options yet which I think will limit Asterisk's appeal as a
video PBX. 

Is fmtp something that is scheduled to be worked on? We would be
interested in helping out if there was a consensus that it was worth
pursuing.

Thanks again for you time,
Regards, John

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: 06 September 2005 00:16
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] Call and video bandwidths

John Martin wrote:

> it's in beta. We can see that video support is now much better, but,
as
> far as we can see call and video bandwidth capabilities/prefs are not
> passed through Asterisk to the called party.

Asterisk is not a SIP proxy, so it negotiates the two call legs 
independently. The capabilities presented to the called endpoint are 
based on how the administrator configures Asterisk, with a small amount 
of input based on the originating endpoint's capabilities.
_______________________________________________
Asterisk-Dev mailing list
Asterisk-Dev at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-dev
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev