Olle has posted some more information on the IAX2 transfers of media I spoke of yesterday:
To update you on recent changes in svn trunk, I can inform you that we now have ever smarter ways to handle media streams in Asterisk than we do in 1.2 for the IAX2 and SIP protocols.
IAX2: Splitting signalling and media apart
Starting with the IAX2 protocol, we now have the ability to transfer media streams to go directly between IAX2 servers and keep the signalling path. Before, when Asterisk did a native transfer to optimize the IAX2 call path, we lost all tracks of the call and could not get a CDR. With this patch, by Mark, we now have a hybrid solution that releases the media but keeps IAX2 signalling.
This is a very new feature, so I don't expect the various non-asterisk IAX2 clients out there to support it yet. When they do, it will mean a huge change in the number of calls your server can handle. For now, this optimizes calls in Asterisk IAX2 "clusters".
SIP: Removing the media immediately, not as an afterthought
Mark and Kevin have been working on various ways to optimize the setup of a SIP call where Asterisk has no reason to stay in the media stream. Asterisk will now setup the call directly between the two devices instead of accepting the call, staying in the stream and then, as a sudden afterthought, send re-invites to release the media stream.
An additional new feature, inspired by a community patch on the bug tracker, is that we now also release calls if SIP INFO dtmf is used. Since the DTMF is not handled in the RTP media stream, we can release the call (unless there is another reason to stay in the media path, like NAT support).
These changes optimize your Asterisk a great deal and will hopefully make Asterisk scale a bit more. Your development team is as always focused on scaling issues, trying to go where no software PBX has gone before, explore new telephony territories... VoiP trekking... Well, enough of that. Sorry, got sidetracked.
Asterisk 1.4 - I see a shape, an outline
The work with Asterisk 1.4 is going into the final stages. We are working hard to commit the changes that are ready and finalize the 1.4 release. If you visit the bug tracker, you already see patches that we've marked "post 1.4" since we feel they're not ready. The next release is not that far away, so it's not a big thing. We won't wait over 1 year like we did between 1.0 and 1.2.
This weekend, I'm leaving for my Training in New York. Next training is in Stockholm, Sweden in June, after that we're launching the Asterisk SIP Masterclass in Chicago in July - with a gold team teaching: Ed Guy, Terry Wilson and myself.
While I'm travelling around, you can spend all your free time testing Asterisk 1.4 for us.
We need your help, now. Download svn trunk and test in your environment!
On behalf of the community - thank you for testing!
* Olle E. Johansson - oej at edvina.net
* Asterisk Training http://edvina.net/training/