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

Eric Wieling eric at fnords.org
Tue Sep 9 13:14:54 MST 2003


Codecs are always an issue.  Best to put disallow=all and
allow=whatevercodecyouwant in each [sipuser] entry.  You can't have
Asterisk do codec translation (transcoding) bewteen g729 and some other
codec unless you have the g729 licenses (US$10/channel from Digium).

Transcoding would be required for access to ANY of the asterisk sound
files, voicemail and PSTN via Zap interfaces.

On Tue, 2003-09-09 at 14:51, Ernest W. Lessenger wrote:
> At 02:38 PM 9/9/2003 -0500, you 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.
> 
> So Asterisk will allow it... and if I set both to no, asterisk would act as 
> a true proxy, using the most bandwidth efficient codec available for each 
> leg of the call (i.e. GSM for x-lite and g.729 for Cisco et al)?
> 
> Thanks,
> --Ernest
> 
> 
> >On Tue, 2003-09-09 at 13:04, Sean P. Robertson wrote:
> > > I have seen this asked in the archives several times, but do not see a
> > > definitive answer anywhere. Is there a way to tell the Asterisk to act like
> > > a "normal" SIP Proxy, handling only the SIP messages, and letting the 
> > RTP go
> > > point-to-point?
> > > ----- Original Message -----
> > > From: "Sean Figgins" <sfiggins at mail.celicas.org>
> > > To: <asterisk-users at lists.digium.com>
> > > Sent: Tuesday, September 09, 2003 1:40 PM
> > > Subject: Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet
> > >
> > >
> > > > On Mon, 8 Sep 2003, Jim Mercer wrote:
> > > >
> > > > > > Can we bribe you? :)
> > > > >
> > > > > sure, pay my rent for 3 months and give me a 50" plasma TV to play in
> > > the
> > > > > background.
> > > >
> > > > Is that all?  That sounds rather cheap, compared to the things direction
> > > > that I'd have to go if I wanted to stick to the cisci CM route, with
> > > > licenses for every endpoint that I want to connect.
> > > >
> > > > Realistically...  I just can not comprehend how to get stuff to work
> > > > correctly with Linux.  I used to be a Linux nut years ago, but once I
> > > > found FreeBSD with it's ports collection, I wondered why anyone ever
> > > > bothered with Linux and it's completely messed up software install
> > > > requirements.
> > > >
> > > > Right now, under RedHat 9.0, I have * running, but no hardware, and I
> > > > can't figure out how to get h.323 operational so I can talk to my cisco
> > > > gateway with the PRI interface...  I'm only guessing that FreeBSD 
> > would be
> > > > much easier for non-programmers like myself.
> > > >
> > > > -Sean
> > > >
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >--
> >BTEL Consulting
> >850-484-4535 x2111 (Office)
> >504-595-3916 x2111 (Experimental)
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> >
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