From 17afebc1a66f6cb114abfbd0a0490f7e45b3bdc6 Mon Sep 17 00:00:00 2001 From: Olle Johansson Date: Sun, 16 Dec 2007 10:51:53 +0000 Subject: HUGE improvements to QoS/CoS handling by IgorG - Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/unistim.conf.sample | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'configs/unistim.conf.sample') diff --git a/configs/unistim.conf.sample b/configs/unistim.conf.sample index 4a6c61abc..649737317 100644 --- a/configs/unistim.conf.sample +++ b/configs/unistim.conf.sample @@ -4,6 +4,13 @@ [general] port=5000 ; UDP port +; +; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters. +;tos=cs3 ; Sets TOS for signaling packets. +;tos_audio=ef ; Sets TOS for RTP audio packets. +;cos=3 ; Sets 802.1p priority for signaling packets. +;cos_audio=5 ; Sets 802.1p priority for RTP audio packets. +; ;keepalive=120 ; in seconds, default = 120 ;public_ip= ; if asterisk is behind a nat, specify your public IP ;autoprovisioning=no ; Allow undeclared phones to register an extension. See README for important -- cgit v1.2.3