[asterisk-dev] RE: [Asterisk-Users] Bandwidth Management

[Alexander Lopez]

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 at lists.digium.com 
> [mailto:asterisk-users-bounces at 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 at lists.digium.com 
> [mailto:asterisk-users-bounces at 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 écrit :
> 
> >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/
> Découvrez la Réunion des Technologies IP & Telecom
> TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
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> 
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