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

[Venefax]

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.
> 
> 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
Web    : http://www.evaristesys.com/
Tel    : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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