[Asterisk-Users] Bandwidth Management

[Wai Wu]

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/
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