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

[Dave Packham]

can you share the SIP conf entries that you are using to get this to work?   I have played with the canreinvite and reinvite entries but cannot make my 7960's do P2P  I am running the 5.1 SIP code on the phones.   

Dave


>>> ALow at Prioritytelecom.com 7/29/2003 3:13:54 AM >>>
Thanks all,

I spent some time on this last night with packet sniffer in hand, the 'canreinvite' option makes sense and seems to work well for me (running latest * CVS release) when used between 79xx phones and the AS5300 gateway although I get some somewhat expected problems with 79xx that are NAT'd behind ADSL/cable connections.

I don't seem to be hitting the bug that Dave mentioned below ...

> -----Original Message-----
> From: Dave Packham [mailto:dave.packham at utah.edu] 
> Sent: 29 July 2003 04:30
> To: asterisk-users at lists.digium.com 
> 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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