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

[Alex Balashov]

Asterisk *cannot* *work* the way that a *proxy* works.

It is not a proxy.  The difference here is an essential, a conceptual 
one that translates into the very basis of the respective programmatic 
architectures and runtime characteristics.

Asterisk can be optimised.  It can't be ontologically redefined.

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.
> 
> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Alex Balashov
> Sent: Thursday, April 23, 2009 3:32 AM
> To: Asterisk Developers Mailing List
> Subject: Re: [asterisk-dev] Avoiding RTP flows (new topic)
> 
> Olle E. Johansson wrote:
> 
>> The SIP proxy will always be faster and more  
>> lightweight,
>> unless you add a lot of call stateful functionality to it.
>>There are multiple proposals up for improving the media negotiation in  
Asterisk
and we've spent quite a lot of time discussing this on the mailing
list and
during asterisk developer meetings. The videocaps branch has been around
for many years now, which is one of the proposals for improved media
negotiation, even though it's not focusing on the directrtp issue,
it's one way
of solving it.

>> Personally, I don't see why you want to make one piece of software into
>> a different piece of software. OpenSER/Kamailio has one role in the
>> SIP network, Asterisk a very different role and they work well together.
> 
> I would second that, and imagine that the difference in the resources 
> Asterisk allocates to a call is far greater than just dialog statekeeping.
> 


-- 
Alex Balashov
Evariste Systems
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