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

Tue Jul 29 13:22:39 MST 2003

Dave,

You can use a sniffer to view the contact field in the INVITE Message that
the Originating Phone sends to *.  Then look at the INVITE Message that *
sends to the remote phone and compare the contact filed.  You will see that
the IP Address is changed to reflect the IP of *.  If you want pure P2P then
that address needs to remain the same.  I have not seen how you can do that
with *.

Ricardo

----- Original Message -----
From: "Dave Packham" <dave.packham at utah.edu>
To: <asterisk-users at lists.digium.com>; <jtodd at loligo.com>;
<ALow at Prioritytelecom.com>
Sent: Tuesday, July 29, 2003 3:00 PM
Subject: RE: [Asterisk-Users] RTP session traversing Asterisk server...

> 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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