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List:       asterisk-dev
Subject:    Re: [asterisk-dev] Asterisk application and SIP channel
From:       Russell Bryant <russell () digium ! com>
Date:       2010-01-28 15:04:37
Message-ID: 4B61A785.8030408 () digium ! com
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On 01/28/2010 09:00 AM, Olle E. Johansson wrote:
> 
> 28 jan 2010 kl. 15.51 skrev Russell Bryant:
> 
> > On 01/28/2010 02:18 AM, nedo nodo wrote:
> > > I'm new of Asterisk. I'm studying how improve an application of Asterisk
> > > (app_konference). Now I need a lot of RTP information like audio/video
> > > ports, ip address etc. The application receive from Asterisk only
> > > ast_channel data structure that doesn't contain RTP information. How can I
> > > retrieve these information?
> > 
> > Strictly speaking, this violates Asterisk architecture.  Applications
> > are supposed to work independently of what channel type is in use.
> > However, there is one interface that could be extended to provide the
> > information you need.  Take a look at the CHANNEL() function defined in
> > funcs/func_channel.c.  Through this dialplan function, channel drivers
> > can provide channel type specific information.  chan_sip already
> > provides a number of things.  You could extend it to suit your needs and
> > execute CHANNEL() from the application.
> 
> Agreed.
> 
> The question here is why you need RTP information in app_konference. We're a \
> multiprotocol PBX and non-RTP protocols also support video (like IAX2). It's better \
> to try to stick to the multiprotocol approach as long as you can and avoid \
> protocol-specific features in applications.

Furthermore, we would much rather you work on adding whatever 
functionality you would like to add to the conferencing code that is in 
Asterisk for all to benefit, instead of a third party module.  :-)

-- 
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org

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