[Asterisk-Users] Re: how can I test canreinvite effectivness?

[Tom Schroer]

> Subject: Re: [Asterisk-Users] how can I test canreinvite effectivness?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 	<asterisk-users at lists.digium.com>
> Message-ID: <200410142241.49603.denis at isolve.com.br>
> Content-Type: text/plain;  charset="iso-8859-1"
> 
> Try IPTRAF or TCPDUMP.
> 
> Denis.
> 
> Em Qui 14 Out 2004 19:12, Matthew Boehm escreveu:
> > I'm not running X or any kind of GTK/GUI abilities on our asterisk 
> > server. I need some sort of ability to test wether sip 
> canreinvite is 
> > working.
> >
> > If it is, then the network usage should be 
> minimal/nonexistant because 
> > all voice packets should be going phone-to-phone.
> >
> > If it is not, then network usage would be high because all voice 
> > packets would be going phone-to-asterisk-to-phone
> >
> > Does anyone know of a nice ncurses or terminal based 
> realtime network 
> > usage app?
> >
> > Or is there some other way in asterisk I can tell if the phones are 
> > talking to each other directly?
> >
This may be brute force and there may be more elegant methods, but I
just monitor on the server with "tethereal -R rtp" and if I see packets
then * is not releasing the media stream.  The problem is that I have
found that this can impair call quality if you leave it up, so I only do
it to spot check.  Also, I do an ethereal trace on the UA and look at
the source/destination address of the rtp stream and that should tell
you as well if the rtp is released.


> > Thanks,
> > Matthew
> >
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