[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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