In the paper: "To generate a 64 Kbps flow, 64 bytes packets were sent every 8 milliseconds."
This is not how the G.711 codec works...it's normally a 20 msec packetization rate resulting in ~220 byte frames (sans 802.11 headers, I believe). I've seen too much undergraduate start with faulty assumptions such that the results are bogus. The only thing these guys have proven is that lots of small packets cause latency and jitter when the link becomes saturated. No surprise there. Regards, Frank -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Osmon Sent: Tuesday, May 22, 2007 11:21 PM To: [email protected] Subject: Re: [c-nsp] VoIP without QoS On Tue, May 22, 2007 at 01:47:15PM -0500, Dan wrote: > We have a voip system we have been running in our department now for > about a year. Only 12 phones, connected through various wireless links > with throughput of up to 40mbit. Speed is definitely not an issue for > us, but we notice glitches with the quality on an ongoing basis. We are > currently implementing qos and are wondering what is the best way? Most common wireless solutions don't really like a lot of small packets. They tend to have too much overhead in the protocols, and you if you're pushing small packets, you hit a pps limit long before anything comes close to using up the bandwidth you think is available. While trying to research why VOIP over wireless networks, I ran into this paper: http://www.it.uc3m.es/~acuevasr/publicaciones/LCS06.pdf _______________________________________________ cisco-nsp mailing list [email protected] https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/ _______________________________________________ cisco-nsp mailing list [email protected] https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/
