CUBE remarks audio rtp from af41 to ef. Why ? And how to fix it ?

[jevgep]

Scenario:

Video call from IP phone 1 to IP phone 2.

IP phone 1 (video)--->CUCM --> CUBE ---> CME---IP phone 2(video/DN4000)

 --------------------------------------------------------------------

Problem:

CUBE remarks audio RTP packets from AF41 to EF

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Description:

Wireshark trace shows that both parts of video call(audio and video  RTP packets) when they come out from IP phone 1 are marked with DSCP AF41.

When audio RTP stream leaves CUBE it is marked as EF not AF41. Video RTP  stream remains marked as AF41

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Config on CUBE(media flow-trough mode):

dial-peer voice 4000 voip
destination-pattern 4...$
rtp payload-type cisco-codec-fax-ack 111
rtp payload-type cisco-codec-video-h264 97
session protocol sipv2
session target ipv4:1.1.1.1
incoming called-number 4...$
voice-class sip pass-thru content sdp
!

--------------------------------------------------------------------------

Dial-peer DSCP settings:

cube#show dial-peer voice 4000 | in DSCP
ip media DSCP = ef, ip media rsvp-pass DSCP = ef
ip media rsvp-fail DSCP = ef, ip signaling DSCP = af31,
ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
ip video rsvp-fail DSCP = af41,

[Mohammed al Baqari]

If you want to retain af41 for audio, then apply this command on outbound
dialpeer in cube.

dial-peer voice x voip
ip qos dscp af41 media

for video the command is ip qos dscp af41 video rsvp-none

[jevgep]

Hi Mohammed,
thanks for response.
Have already tried to play with "ip qos dscp af41 media". It seem doesn't work.
As per "ip qos dscp af41 video rsvp-none" - this is a default for voice Dial-peers.

[jevgep]

If my understanding is correct CUBE should(?) treat a video call as a whole(video+audio RTP) and apply if there is a need -qos marking accordingly AF41.

Maybe I am wrong ?

[Mohammed al Baqari]

during active call please share the output of sh call act vo br and show call act vid br

[jevgep]

cube#show call active voice brief
<ID>: <CallID> <start>.<index> +<connect> pid:<peer_id> <dir> <addr> <state>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>

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>
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: 0
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 0
--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------

cube#show call active video brief
<ID>: <CallID> <start>.<index> +<connect> pid:<peer_id> <dir> <addr> <state>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>

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>
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
video: h320:<type> tx:<video codec> <video pkts>/<video bytes> rx:<video codec> <video pkts>/<video bytes>
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 : 1258 06:53:40.338 CDT Thu Nov 2 2017.1 +4440 pid:4000 Answer 2002 active
dur 00:00:31 tx:2740/562728 rx:2569/138788
IP 10.1.110.19:24914 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms pass-through TextRelay: 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

0 : 1259 06:53:40.358 CDT Thu Nov 2 2017.1 +4400 pid:4000 Originate 4002 active
dur 00:00:31 tx:2569/107960 rx:2740/595608
IP 1.1.1.1:16694 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms pass-through TextRelay: 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


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]

If you apply ip qos dscp af41 and make audio call, is it marking the packets with af41. I don't see why your call is marking audio as ef while you have configured af41 and you are matching the right dialpeer.

[jevgep]

!
dial-peer voice 4000 voip
destination-pattern 4...$
rtp payload-type cisco-codec-fax-ack 111
rtp payload-type cisco-codec-video-h264 97
session protocol sipv2
session target ipv4:1.1.1.1
incoming called-number 4...$
voice-class sip pass-thru content sdp
!
Above dialpeer works as incoming/outgoing for calls which destination are 4...$
When incoming video call(2 RTP streams: video RTP/audio RTP, both streams marked as AF41) hits that dialpeer CUBE remarks audio part of video call(audio RTP) from AF41 to EF. In my opinion it is not correct. In my opinion because audio RTP stream belongs to video call it should remain marked as AF41 when it leaves CUBE.