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

[Dave Packham]

OK calls thru the * server are looped and calls with the same phones thru Free WOrld Dialup are P2P.....  same configs...

Anyone have any ideas?  I know its a bug but we need to fix this one.... I think its pretty big one.  it would HAMMER the scalability of * servers

Dave

>>> ALow at Prioritytelecom.com 7/29/2003 8:01:41 AM >>>
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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