[asterisk-users] directrtp with SIP + H.323

[Olle E. Johansson]

24 feb 2010 kl. 01.22 skrev Kristian Kielhofner:

> On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis <support at ocg.ca> wrote:
>> We're creating a SIP gateway for a client that will take one leg of a call
>> in via SIP, and out the other side via H.323.  To minimize load on the
>> gateway, we would like to have the RTP stream bypass the gatewayy altogether
>> (directrtp/reinvite).  Is this possible with these to protocols?
>> 
>> Thanks
> 
> Yate claims it can do this:
> 
> http://yate.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
> 
There are two ways - either by reinvites, which according to Kevin won't work with H323, or by doing it right in the call setup. If we did that, we would stumble into the same problem as we have with this function in SIP - which goes all back to the media negotiation framework (see http://www.voip-forum.com/asterisk/2010-02/asterisk-18-lts-wishlist-1-media-negiotiation-framework/ ).

Asterisk currently just communicates an answered call as "answered" over the bridge without any attributes. This is the reason why the code has been marked "experimental" for many releases and no one has solved it. In order for this to work, you either need exactly the same codec attributes or a way to handle the ANSWER control frame (like John Martin did in the videocaps branch).

The hooks are all there if you want to experiment with this in the H.323 channel. It's certainly possible. But it is not a function I would support generally (which is why the directrtp call setup function remains experimental).

/O