Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan -- Andy Tan andytan@fastmail.fm -- http://www.fastmail.fm - Does exactly what it says on the tin
Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan -- Andy Tan andytan@fastmail.fm -- http://www.fastmail.fm - mmm... Fastmail...
"Out of the Box" probably not but with an AGI script this is very doable: You can have a script that monitors active calls and the Codecs that are in use. The script will have to do some math to calculate the bandwidth in use and then using the variables in Asterisk, Namely SIP_CODEC. If you are using SIP. There has not been a Variable coded for the other Technologies at this time. Alex> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andy Tan > Sent: Tuesday, April 11, 2006 9:00 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Bandwidth Management > > Hi, > > understand that the bandwidth utilized for each call is > dependent on the codec used, wonder if Asterisk can monitor > the total bandwidth utilized and restrict/reject new calls > when the resource is insufficient to support them reliably? > > Regards > Andy Tan > -- > Andy Tan > andytan@fastmail.fm > > -- > http://www.fastmail.fm - mmm... Fastmail... > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
On 4/11/06, Andy Tan <andytan@fastmail.fm> wrote:> Hi, > > understand that the bandwidth utilized for each call is dependent on the > codec used, wonder if Asterisk can monitor the total bandwidth utilized > and restrict/reject new calls when the resource is insufficient to > support them reliably? > > Regards > Andy TanTo the best of my knowledge, Asterisk does not have such a feature at the current time. -Rusty
Hi Alex, thanks for the suggestion. Did some checks, and thought that I could set a global variable to track the utilized bandwidth. Wish that there are plans for support to include variables like SIP_CODEC in other protocols. Regards On Tue, 11 Apr 2006 12:50:56 -0400, "Alexander Lopez" <Alex.Lopez@OpSys.com> said:> "Out of the Box" probably not but with an AGI script this is very > doable: > > You can have a script that monitors active calls and the Codecs that are > in use. The script will have to do some math to calculate the bandwidth > in use and then using the variables in Asterisk, Namely SIP_CODEC. If > you are using SIP. There has not been a Variable coded for the other > Technologies at this time. > > Alex > > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andy Tan > > Sent: Tuesday, April 11, 2006 9:00 AM > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] Bandwidth Management > > > > Hi, > > > > understand that the bandwidth utilized for each call is > > dependent on the codec used, wonder if Asterisk can monitor > > the total bandwidth utilized and restrict/reject new calls > > when the resource is insufficient to support them reliably? > > > > Regards > > Andy Tan > > -- > > Andy Tan > > andytan@fastmail.fm > > > > -- > > http://www.fastmail.fm - mmm... Fastmail... > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Andy Tan andytan@fastmail.fm -- http://www.fastmail.fm - And now for something completely different
I think this belongs to the development mail-list. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jean-Michel Hiver Sent: Wednesday, April 12, 2006 12:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bandwidth Management Andy Tan a ?crit :>Hi Alex, > >thanks for the suggestion. > >Did some checks, and thought that I could set a global variable to >track the utilized bandwidth. > >Wish that there are plans for support to include variables like >SIP_CODEC in other protocols. > >Actually this sounds like a really nice idea. It would be cool to have a way to start using less intensive bandwith codecs for new calls when bandwith reaches a certain threshold. For example: - 0-40% bandwith: g711 - 40-60% bandwith: g729 - 60%-80% bandwith: g723 - 80%-100% bandwith: drop new calls, or maybe use lpc10 It wouldn't help in SOHO usage but when using Asterisk as a call termination gateway, it would help making the most out of available bandwith. g711 is certainly better than g729 when you have the bandwith, and i'm pretty sure that even lpc10 sounds better when on non-saturated bandwith compared with g729 with some packet loss... How would you go about implementing this? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Brought over from -users, Please reply to the -dev list.
I agree, lets move the discusstion over to that list as it has to be discussed
there. After we reach an accord on how it should be done we will open up a issue
on Mantis.
I see this as being two distinctive parts that would need to be tied together:
First: We need to make the selection of CODECS technology agnostic, There
currently exist a facility for CODEC selection (SIP_CODEC) in the sip channel
but not in others.
Second: Discuss is this sould be an outside application that is called from
within Asterisk or if it should become a function
Set(CODEC=${OPTIMALCODEC(quality)})
available options could be:
quality
bandwidth
license
Any comments.
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wai Wu
> Sent: Wednesday, April 12, 2006 10:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Bandwidth Management
>
> I think this belongs to the development mail-list.
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
> Jean-Michel Hiver
> Sent: Wednesday, April 12, 2006 12:05 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Bandwidth Management
>
> Andy Tan a ?crit :
>
> >Hi Alex,
> >
> >thanks for the suggestion.
> >
> >Did some checks, and thought that I could set a global variable to
> >track the utilized bandwidth.
> >
> >Wish that there are plans for support to include variables like
> >SIP_CODEC in other protocols.
> >
> >
> Actually this sounds like a really nice idea. It would be
> cool to have a way to start using less intensive bandwith
> codecs for new calls when bandwith reaches a certain threshold.
>
> For example:
>
> - 0-40% bandwith: g711
> - 40-60% bandwith: g729
> - 60%-80% bandwith: g723
> - 80%-100% bandwith: drop new calls, or maybe use lpc10
>
> It wouldn't help in SOHO usage but when using Asterisk as a
> call termination gateway, it would help making the most out
> of available bandwith. g711 is certainly better than g729
> when you have the bandwith, and i'm pretty sure that even
> lpc10 sounds better when on non-saturated bandwith compared
> with g729 with some packet loss...
>
> How would you go about implementing this?
>
> Cheers,
> Jean-Michel.
>
> --
> Jean-Michel Hiver - http://ykoz.net/
> D?couvrez la R?union des Technologies IP & Telecom
> TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
>
>
> _______________________________________________
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