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

Brian Capouch brianc at palaver.net
Mon May 12 16:14:45 MST 2003


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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