asterisk users - May 2006 - Development news :: Smarter medialess calls!

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@edvina.net
* Asterisk Training http://edvina.net/training/

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@edvina.net
> * Asterisk Training http://edvina.net/training/
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users