We've recently deployed about 4 asterisk boxes to clients behind m0n0wall
> Is there anyone out there who has a m0n0wall Traffic Shaper
> ruleset for VOIP usage? If there's no-one updating the
> manual with this information then I am willing to take this
> on, even though we don't use VOIP here ourselves yet, and
> I'll keep updating this section as we start to play ourselves.
I'll happily post my traffic shaping rules to the list once I've cleared up
the P2P limiting rules I also have running.
But a couple of points:
1) Bandwidth requirements can vary massively depending on what VoIP systems
you're running. For example, an Asterisk server running g729 over IAX will
use about 21kbps for the first call, but only about 9kbps extra for
additional calls *provided* you have a timing device (use ztdummy if you
don't have a digium card installed) and of course the remote IAX gateway
supports trunking. By contrast, a g711 SIP call will be about 80kbps per
call, with each call using exactly the same bandwidth as the first call.
Even a g729 SIP call will use about 32kbps per call. Where possible use an
IAX gateway if there are bandwidth considerations.
So on a 256kbps link, the number of concurrent calls could vary anywhere
between 3 and over 20 depending on codec and signalling protocol used.
2) Decide on how you want call quality to degrade (as gracefully as
possible). If you enable the jitter buffer, your calls won't stutter as much
when bandwidth becomes scarce, but you will start to pick up an obvious
echo. In this case it might be better to write something into your dialplan
to refuse outbound calls if the number of concurrent calls exceeds a certain
number, and route subsequent calls out via PSTN.
C.M. Bagnall, Director, Minotaur I.T. Limited
Tel: (07010) 710715 Mobile: (07811) 332969 Skype: minotaur-uk
ICQ: 13350579 AIM: MinotaurUK MSN: msn at minotaur dot cc Y!: Minotaur_Chris
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