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

[Low, Adam]

Sure, nothing special though:

[4840]
type=friend
username=4840
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband

[4842]
type=friend
username=4842
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband



> -----Original Message-----
> From: Dave Packham [mailto:dave.packham at utah.edu] 
> Sent: 29 July 2003 15:43
> To: asterisk-users at lists.digium.com; ALow at Prioritytelecom.com
> Subject: RE: [Asterisk-Users] RTP session traversing Asterisk 
> server ...
> 
> 
> 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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This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person