[Asterisk-Users] Development news :: Smarter medialess calls!

[Sean Cook]

Olle,

Is there a poster of you that I can put up on my wall ;)

Regards,

Sean

Olle E Johansson wrote:
> Friends,
> 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!
>
> SIP greetings!
> /Olle
>
> ---
> * Olle E. Johansson - oej at edvina.net
> * Asterisk Training http://edvina.net/training/
>
>
>
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