Has anyone played around with QoS or TOS relative to * and sip phones?http://www.voip-info.org/wiki-Asterisk+sip+tos
I was just doing a little real-time research and noticed our C7960's mark IP packets with "low delay" and "high throughput" (presumably due to tos_media: 5 in the SIPDefault config file), and rtp packets flowing "from" asterisk back to the sip phone are not marked at all.
Is there a * config parameter to enable such a function?
/O
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