[Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?

[Bill Jennings]

Brian,

I run a fixed-wireless ISP using 802.11.  I have one customer who could
not get phone service at one of his locations, but could at another.
Since both locations have wireless internet service, I installed an
asterisk box at each location with three fxo cards at one location, and
a channel bank at the other.

The link between the two locations is a simplex repeater (similar to an
access point, but operating in IBSS mode) which is shared by a dozen or
so of my other customers.

It took a while before I could get the sound quality to an acceptable
level, but that was mainly choosing the correct echo cancellor.

The latency is not a problem, as all of the routers (linux boxes) honor
the TOS flags, and asterisk sets them correctly.

FWIW, my $.02

Bill Jennings
Big River Wireless

Brian Capouch (brianc at palaver.net) wrote:
> Just wondering if anyone out there has done any work, or knows where any 
> work is being done, to try to honor the latency requirements of this 
> VOIP stuff and push out SIP and RTP traffic, etc., "ahead of the crowd."
> 
> I'm doing my VOIP behind wireless, so it is particularly important.  I 
> am getting ready to do some digging, and don't want to re-invent the wheel.
> 
> Thx.
> 
> B.
> 
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