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

Fri Aug 1 14:20:26 MST 2003

Ricardo,

You are right about the contact field in the INVITE message.  It does
display the address or our Asterisk proxy.  It seems to me that this
field is used for endpoints to exchange future SIP messages among
themselves and not to set up the RTP stream.  I have found that the SDP
Connection (c) field in the invite also reflects the IP of the Asterisk
box after the message leaves the proxy. The 200 OK reflects the same
symptoms.  I think that this is the reason the RTP stream is being set
up between the endpoint and the server.  Do you think that the contact
field and connection field being incorrect may be related?  You also
have mentioned that you have not seen a way to configure this with
Asterisk. How about other SIP proxies such as VOCAL?

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
ricvil at telesip.net
Sent: Tuesday, July 29, 2003 2:23 PM
To: andrew.reich at utah.edu
Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server...

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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