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

[Olle E. Johansson]

23 apr 2009 kl. 09.01 skrev Venefax:

> I wonder what would it take so we in Asterisk can guarantee that RTP  
> flows
> outside in a SIP-to-SIP call. I mean, if the codecs are the same,  
> how can we
> be sure that there will never be a packet-to-packet bridging or  
> anything
> else like that. If we did this, then Openser would be irrelevant.  
> Any ideas?
>
The behaviour today is not random. If you configure properly, Asterisk  
won't
be involved in the call.

As a side note: The directrtp= option only works if you have complete  
control
of codecs on the answering end. If not, it will fail miserably, and  
that's the
reason it's still marked experimental.

But you still have to remember that Asterisk still allocates much more  
resources
for each call, as Asterisk is a call stateful PBX, compared with a  
transaction
stateful SIP proxy. 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.

/O