[Asterisk-Users] RTP session traversing Asterisk server ...

[Dave Packham]

Check out this bug

http://bugs.digium.com/bug_view_page.php?bug_id=0000005

its a know problem.  I have played with the canreinvite stuff to no end and have never gotten my Cisco Phones to do P2P RTP.  I am going to try free world dialup to see if it does P2P with my Cisco Phones  then it might just be a message thing on * server.

Dave Packham


>>> danfernandez00 at hotmail.com 7/28/2003 4:16:16 PM >>>
On  your sip.conf for each sip endopoint set canreinvite = yes.

That way the rtp stream won t go through *. The only problem though is for
ATA 186. They need canreinvite = No when they are in a NAT environment.



----- Original Message -----
From: "Low, Adam" <ALow at Prioritytelecom.com>
To: <asterisk-users at lists.digium.com>
Sent: Monday, July 28, 2003 11:29 AM
Subject: [Asterisk-Users] RTP session traversing Asterisk server ...


>
> I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would
expect the RTP session should ideally be between the two end points of the
call, in my case the AS5300 and the 7940 which are connected on the same
VLAN as the Asterisk server.
>
> When I sniff the packets on the VLAN I find that all RTP packets are being
relayed by the Asterisk server causing increased load on the server and
ultimately a higher latency between the two end points.
>
> Is this a typical operation of Asterisk or is this possibly due to the
fact that some of the phones (not those used in the tests) are running NAT
and Asterisk relays all RTP packets ?
>
> Adam
>
>
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