Matthew Boehm
2004-Oct-14 15:12 UTC
[Asterisk-Users] how can I test canreinvite effectivness?
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? Thanks, Matthew
William Suffill
2004-Oct-14 16:04 UTC
[Asterisk-Users] how can I test canreinvite effectivness?
Ntop.org probably could fit you needs from the console.
Denis Galvão
2004-Oct-14 18:41 UTC
[Asterisk-Users] how can I test canreinvite effectivness?
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? > > Thanks, > Matthew > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Tom Schroer
2004-Oct-15 06:03 UTC
[Asterisk-Users] Re: how can I test canreinvite effectivness?
> Subject: Re: [Asterisk-Users] how can I test canreinvite effectivness? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <200410142241.49603.denis@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 > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Tom Schroer
2004-Oct-15 06:06 UTC
[Asterisk-Users] Re: how can I test canreinvite effectivness?
> Subject: Re: [Asterisk-Users] how can I test canreinvite effectivness? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <200410142241.49603.denis@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 > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >