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

8 files changed, 32 insertions, 11 deletions

diff --git a/configs/dundi.conf.sample b/configs/dundi.conf.sample
index fa3d255d5..f62210659 100644
--- a/configs/dundi.conf.sample
+++ b/configs/dundi.conf.sample
@@ -27,7 +27,7 @@
 ;bindaddr=0.0.0.0
 ;port=4520
 ;
-; See doc/qos.tex for a description of the tos parameter.
+; See qos.tex or Quality of Service section of asterisk.pdf for a description of the tos parameter.
 ;tos=ef
 ;
 ; Our entity identifier (Should generally be the MAC address of the
diff --git a/configs/h323.conf.sample b/configs/h323.conf.sample
index 08849f690..5be321f33 100644
--- a/configs/h323.conf.sample
+++ b/configs/h323.conf.sample
@@ -4,7 +4,10 @@
 [general]
 port = 1720
 ;bindaddr = 1.2.3.4 	; this SHALL contain a single, valid IP address for this machine
-;tos=ef
+;
+; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
+;tos_audio=ef		; Sets TOS for RTP audio packets.
+;cos_audio=5		; Sets 802.1p priority for RTP audio packets.
 ;
 ; You may specify a global default AMA flag for iaxtel calls.  It must be
 ; one of 'default', 'omit', 'billing', or 'documentation'.  These flags
diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample
index 01ccfb7f6..2441f2cf4 100644
--- a/configs/iax.conf.sample
+++ b/configs/iax.conf.sample
@@ -225,7 +225,7 @@ forcejitterbuffer=no
 ;
 ;authdebug=no
 ;
-; See doc/qos.tex for a description of the tos parameters.
+; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
 ;tos=ef
 ;cos=5
 ;
diff --git a/configs/iaxprov.conf.sample b/configs/iaxprov.conf.sample
index 8f979b533..06891d785 100644
--- a/configs/iaxprov.conf.sample
+++ b/configs/iaxprov.conf.sample
@@ -53,7 +53,7 @@ codec=ulaw
 ;
 flags=register,heartbeat
 ;
-; See doc/qos.tex for a description of this parameter.
+; See qos.tex or Quality of Service section of asterisk.pdf for a description of this parameter.
 ;tos=ef
 ;
 ; Example iaxy provisioning
diff --git a/configs/mgcp.conf.sample b/configs/mgcp.conf.sample
index c4b5cba59..104891e8a 100644
--- a/configs/mgcp.conf.sample
+++ b/configs/mgcp.conf.sample
@@ -5,8 +5,11 @@
 ;port = 2427
 ;bindaddr = 0.0.0.0
 
-; See doc/qos.tex for a description of the tos parameters.
-;tos=ef
+; 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.
 
 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
 ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 61cdf114a..d8e25e642 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -66,16 +66,16 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 				; and multiline formatted headers for strict
 				; SIP compatibility (defaults to "no")
 
-; See doc/qos.tex for a description of these parameters.
+; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
 ;tos_sip=cs3                    ; Sets TOS for SIP packets.
 ;tos_audio=ef                   ; Sets TOS for RTP audio packets.
 ;tos_video=af41                 ; Sets TOS for RTP video packets.
 ;tos_text=af41                  ; Sets TOS for RTP text packets.
 
-;cos_sip=4                      ; Sets CoS for SIP packets.
-;cos_audio=6                    ; Sets CoS for RTP audio packets.
-;cos_video=5                    ; Sets CoS for RTP video packets.
-;cos_text=0                     ; Sets CoS for RTP text packets.
+;cos_sip=3                      ; Sets 802.1p priority for SIP packets.
+;cos_audio=5                    ; Sets 802.1p priority for RTP audio packets.
+;cos_video=4                    ; Sets 802.1p priority for RTP video packets.
+;cos_text=3                     ; Sets 802.1p priority for RTP text packets.
 
 ;maxexpiry=3600			; Maximum allowed time of incoming registrations
 				; and subscriptions (seconds)
diff --git a/configs/skinny.conf.sample b/configs/skinny.conf.sample
index afbbc8c39..26a6db6c7 100644
--- a/configs/skinny.conf.sample
+++ b/configs/skinny.conf.sample
@@ -28,6 +28,14 @@ keepalive=120
 ;allow=all		; see doc/rtp-packetization for framing options
 ;disallow=
 
+; 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.
+;tos_video=af41		; Sets TOS for RTP video packets.
+;cos=3			; Sets 802.1p priority for signaling packets.
+;cos_audio=5		; Sets 802.1p priority for RTP audio packets.
+;cos_video=4		; Sets 802.1p priority for RTP video packets.
+
 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
 ;jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
                              ; skinny channel. Defaults to "no". An enabled jitterbuffer will
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