[asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.

[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?


-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Olle E.
Johansson
Sent: Thursday, April 23, 2009 1:37 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Symmetric RTP behaviour and 'nat' peer option.


23 apr 2009 kl. 00.35 skrev Venefax:

> Agreed. I wish to see that behavior.
> Maybe that is why directrtp=yes does not work.

In the very old chan_sip2 project I had an option called  
symmetricrtp=yes that turned on symmetric RTP regardless of the SIP  
situation, this was for Asterisk behind proxys. That was one of the  
few changes that did not get approved for merging into Asterisk at the  
time.

Maybe it's time to do something again. Overloading nat= doesn't make  
sense, as the nat setting affects both SIP and RTP.

Always running symmetric RTP is not a good thing, not all devices can  
handle that.

I need to take a look at this.

/O

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