[asterisk-dev] Asterisk 12 Project Update

[Joshua Colp]

Gunnar Hellstrom wrote:
> Hi,
> Thanks for a good overview.
>
> I have a couple of questions on what is included in current plans for
> Asterisk 12.
>
> 1. Real-time text.
> Is the support for real-time text with presentation according to T.140
> and RTP transport according to RFC 4103 included in the plans.
> It is in current versions 1.6, 1.8 etc, and described in
> http://www.voip-info.org/wiki/view/Real-time+text and works well for
> rapid smooth text communication.

It's not on the current list but the design of the new SIP channel 
driver makes it trivial to do. If someone would be interested in 
tackling it the places to look would be res_sip_sdp_rtp.c and chan_gulp.c

> 2. Calls without audio.
> A long time ago, Asterisk had problems with calls without audio. E.g.
> only video.
> Many SIP implementations have the problem that they have audio-centric
> roots and are not verified for audio-less calls.
> I did not see any use case for calls without audio among the video call
> cases and suggest that such cases are added and the case kept in mind
> during the development.

As is the case with a few things in the new SIP channel driver the 
limitation is not the channel driver, it's the Asterisk core. It is 
really designed around expecting an audio stream to always be present.

> 3. Complete SDP parameter handling for video.
> Video calling through Asterisk requires good and comprehensive handling
> and conveyance of video parameters between the call legs to achieve
> interoperability and good video quality between the parties. Are there
> plans for that?

This was done for Asterisk 11 and is available there using chan_sip. 
This functionality is also already used for the new SIP channel driver.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org