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: http://svn.digium.com/svn/asterisk/trunk@93163 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 7 insertions, 0 deletions

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