[Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Sean P. Robertson]

I agree with you about warning that some SIP phones, especially the Cisco
phones, do not handle this well.

Also,  should "can reinvite=yes" in the example be "canreinvite=yes" without
the space or will it work either way?

It might also be useful to note in the documentation that these settings
make Endpoint-to-Endpoint RTP possible because, initially the RTP is routed
by the Asterisk regardless of settings.  If these settings are set to "yes",
very quickly after the call is established and the OK's are sent, the
Asterisk sends another INVITE message to each endpoint in which the RTP
connection address is changed to be the address of the opposite endpoint
rather than the address of the Asterisk.  If the endpoints can handle the
new INVITE (reinvite) then the RTP stream starts flowing directly between
the endpoints.

Sean
----- Original Message -----
From: "Steven Critchfield" <critch at basesys.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, September 10, 2003 12:01 PM
Subject: Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets


> On Wed, 2003-09-10 at 10:42, Olle E. Johansson wrote:
> > Eric Wieling wrote:
> >
> > > That would be reinvite= and canreinvite= in the user entry for each
SIP
> > > endpoint.  Asterisk will allow the endpoints to talk directly to each
> > > other if both those settings are = yes (the default, I think) AND both
> > > endpoints use the same protocol (SIP) AND the same codec.
> >
> > I tried to document this on the Wiki. Did I get it right? Please check!
> > http://tinyurl.com/mvny
> >
> > (If it's wrong, just change it, it's a Wiki)
>
> Minor nitpick, second paragraph 4th word should probably be "two" not
> "to".
>
> Also you may want to point out that there are SIP phones that do not
> react well to the reinvites like the ata186.
>
> --
> Steven Critchfield  <critch at basesys.com>
>
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