QOS In CUBE

[Suchi]

Hi,

I need to mark signalling & RTP traffic in dialpeers pointing to ITSP through SIP trunk,

UCM---SIP Trunk ----CUBE---SIP Trunk ----ITSP

What is the appropriate commands/method ?

Please advise 

[Mohammed al Baqari]

use the command ip qos dscp media x, ip qos dscp signaling y, ip qos dsco video rsvp-none z

[Suchi]

Hi Mohmmed,

Thanks & i configured each dial peer as below commands, but i cant see the ip qos dscp ef media in dial peer config but i can see the signalling config

ip qos dscp cs3 signaling

ip qos dscp ef media

What is the reason that i cant see the media config & how can i verify the qos marking has been applied?

Thanks

[Mohammed al Baqari]

Can you post show version output along with dailpeer config? If you type ip
qos dscp ? when is the output.

During active call you can use show call active voice br. Look for tos
value as it should be 0xb8 for EF.

Another method, you can configure qos policy to match dscp ef & voice
subnet and apply it to egress interface. Then you can see if there are hits
in the voice class of the qos policy.

[Suchi]

HI,

Please see below output, seems OK

<ID>: <CallID> <start>ms.<index> (<start>) +<connect> pid:<peer_id> <dir> <addr> <state>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> dscp:<packets violation> media:<packets violation> audio tos:<audio tos value> video tos:<video tos value>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec> <textrelay> <transcoded

media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>

long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>
LostPacketRate:<%> OutOfOrderRate:<%>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
<codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
<codec> (payload size)
Tele <int> (callID) [channel_id] tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops>
speeds(bps): local <rx>/<tx> remote <rx>/<tx>
Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>
bw: <req>/<act> codec: <audio>/<video>
tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
0 : 53275 1083574000ms.1 (10:44:26.448 AEDT Sun Nov 5 2017) +9280 pid:1000 Answer +61285918298 active
dur 00:01:42 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.21.217.210:24720 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw TextRelay: off Transcoded: No ICE: Off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
LostPacketRate:0.00 OutOfOrderRate:0.00

0 : 53276 1083574010ms.1 (10:44:26.458 AEDT Sun Nov 5 2017) +9270 pid:3005 Originate 0458927976 active
dur 00:01:42 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 192.0.6.6:16400 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw TextRelay: off Transcoded: No ICE: Off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
LostPacketRate:0.00 OutOfOrderRate:0.00

Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2

[Mohammed al Baqari]

Yup, seems that audio is already marked as EF. 

As mentioned earlier you can configure service policy matching dscp ef and voice subnets. Then assign the service policy to exist interface based on routing table to confirm. But the above output shows that marking is taking place.