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

[Dan]

Hi,

Cisco 7940/60 does P2P with FWD.

BR,
Dan


----- Original Message ----- 
From: "Dave Packham" <dave.packham at utah.edu>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, July 29, 2003 5:30 AM
Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server...


> 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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