[asterisk-dev] Avoiding RTP flows (new topic)

[Alex Balashov]

Let me be a little clearer:

Sure, you can get Asterisk out of the media path by just passing through 
the SDP in such a way that an RTP session is set up between the two 
endpoints without involving it at all.  That's all good and fine, and 
well within the capacity of what is then effectively a signaling-only 
[SIP] B2BUA to do.  If you want to do this with a B2BUA instead of a 
proxy, you can already do that with something like Yate.

What you can't do with Asterisk is defy the fundamental purpose of its 
architecture:  it's a PBX.  It is designed as a media endpoint and has a 
heavyweight event loop with a lot of different processes and threads 
being invoked in order to provide a relatively high-level application 
environment.  You will not get the kind of switching setup capacity and 
integration paths with Asterisk that you can get with a proxy dedicated 
solely for that purpose and requiring no bridging of diverse call legs.

Besides that, the integration paths and interfaces offered by the 
OpenSER technology stack are designed on a correspondingly low level of 
abstraction that is commensurate with the one on which the proxy 
operates - it provides a relatively thin layer of (transactional) 
insulation from the underlying application-layer protocol's mechanics.

So, what you're asking is not possible not so much because someone is 
myopic and can't wrap their head around the possibility of passing 
through media without p2p bridging or re-INVITEs, but rather because 
you're conflating concepts.  It's like asking to put the milk back in 
the cow.

Venefax wrote:

> I think that in my company and in many companies, if we could use Asterisk
> the way Openser works, we would save tons of money in training and learning
> curve. Human talent is far more expensive than iron. I wonder why we cannot
> change Asterisk a little so directrpt=yes does in fact work.



-- 
Alex Balashov
Evariste Systems
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