Skype4B 2015 + CUCM 10.5 - Trunk to VIS ok, But there is NO VIDEO between calls only audio!

rfurukawasr3 · ‎06-24-2016

Hello all,

We here are deploying integration with Skype4B 2015 Server + CUCM 10.5 , following Microsoft instructions: (https://technet.microsoft.com/en-us/library/dn951430.aspx).

The Trunk SIP TCP with VIS OK is "UP", But there is NO VIDEO between calls only audio ! Someone had these problems in integration?

Thanks for help me !

Suresh Hudda · ‎06-24-2016

Is proper region is assigned to sip trunk and phone with appropriate bandwidth. 2ndly try to check MTP required on sip trunk and reset it and then check video call once.

Suresh

rfurukawasr3 · ‎06-24-2016

Thanks for your reply Suresh Hudda

The bandwidth is appropriate on region for video: "64 kbps (G.722, G.711) 6000 kbps" like we are using today with VCS-C + Lync2013 ! and "MTP required" option we try both ways on SIP Trunk (check and uncheck)! We try also following other examples like on these links:

http://rahmanny.blogspot.com.br/2015/12/cisco-mcu-skype-for-business_24.html

and

http://www.wavecoreit.com/blog/serverconfig/how-to-setup-and-configure-the-integrationsimulring-between-cisco-unified-communications-manager-10-5-and-lyncskype-for-business-server/

But the calls for SIP Trunk X VIS, there is only audio !!!

Deepak Mehta · ‎06-24-2016

Please can you attach cucm logs for this call with sip tracing enabled.thanks

llobregatil · ‎06-27-2016

Hi Deepal Mehta

I am friend work of the rfurukawasr3, annex i send the trace logs of comunication between skype VIS and CUCM

Calling CUCM SX 10 - Directory number 7925

Called Skype VIS - Directory number 4999

Best regards!

Leandro

Deepak Mehta · ‎06-27-2016

Hi Leandro,

We can see SX10 sending invite to CUCM with video attributes however CUCM replies with media inactive and rtp port 0 for video .Resulting in no video.This is irrespective of which side is calling .

To support video channels, the MTP uses pass-through mode.You may want to make sure The SDP Session-level Bandwidth Modifier for Early is set to TIAS only for skype trunk also set DTMF prerference to no pref.

Call-ID: d9d654c9d43c7759ce9146a13969144a
CSeq: 100 INVITE
Contact: <sip:d1315fbe-f682-43ba-51ae-aee2cc234b73@172.31.98.102:32993;transport=tcp>
From: "SX-10 TESTE" <sip:7925@192.168.205.145>;tag=645906ff80d68f50
To: <sip:*104999@192.168.205.145>
Max-Forwards: 70
Route: <sip:192.168.205.145;lr>
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
User-Agent: TANDBERG/529 (ce8.0.1.1e47efe)
Supported: replaces,100rel,timer,gruu,path,outbound,X-cisco-serviceuri,X-cisco-callinfo,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-sis-7.1.1,norefersub,extended-refer,sdp-anat
Recv-Info: x-cisco-conference
Session-Expires: 1800
Allow-Events: dialog
Remote-Party-ID: "SX-10 TESTE" <sip:7925@192.168.205.145>;privacy=off;id-type=subscriber;screen=yes;party=calling
Content-Type: application/sdp
Content-Length: 2355

v=0
o=tandberg 19 2 IN IP4 172.31.98.102
s=-
c=IN IP4 172.31.98.102
b=AS:3072
t=0 0
m=audio 16512 RTP/AVP 108 104 105 9 18 8 0 101 114
b=TIAS:64000
a=rtpmap:108 MP4A-LATM/90000
a=fmtp:108 profile-level-id=24;object=23;bitrate=64000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:114 opus/48000/2
a=fmtp:114 maxaveragebitrate=48000
a=sendrecv
m=video 16514 RTP/AVP 97 126 96 34
b=TIAS:3072000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428014;max-br=2500;max-mbps=244800;max-fs=8160;max-smbps=244800;max-fps=3000
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428014;max-br=2500;max-mbps=244800;max-fs=8160;max-smbps=244800;max-fps=3000
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=30000
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=10000
a=label:11
a=answer:full
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 28312 UDP/BFCP *
a=setup:actpass
a=confid:1
a=userid:19

================================================
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.31.98.102:32993;branch=z9hG4bKed281476a873cda28d4c90b2ce11f41d;rport
From: "SX-10 TESTE" <sip:7925@192.168.205.145>;tag=645906ff80d68f50
To: <sip:*104999@192.168.205.145>;tag=12217959~e9ac6d02-23b2-4275-98c6-677e800f14e7-112530948
Date: Fri, 24 Jun 2016 19:37:19 GMT
Call-ID: d9d654c9d43c7759ce9146a13969144a
CSeq: 100 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Server: Cisco-CUCM10.5
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= to; gci= 6-1714051; isVoip; call-instance= 1
Send-Info: conference, x-cisco-conference
Session-Expires: 1800;refresher=uas
Require: timer
Remote-Party-ID: <sip:4999@192.168.205.145>;party=called;screen=yes;privacy=off
Remote-Party-ID: <sip:*104999@192.168.205.145;user=phone>;party=x-cisco-original-called;privacy=off
Contact: <sip:*104999@192.168.205.145:5060;transport=tcp>;automata;actor="attendant";text;audio;video;application
Content-Type: application/sdp
Content-Length: 701

v=0
o=CiscoSystemsCCM-SIP 12217959 1 IN IP4 192.168.205.145
s=SIP Call
c=IN IP4 10.129.55.146
b=AS:64
t=0 0
m=audio 27282 RTP/AVP 0 101
b=TIAS:64000
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 31 34 96 97
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=rtpmap:97 H264/90000
a=content:main
a=inactive
m=application 0 UDP/BFCP *
c=IN IP4 0.0.0.0
m=video 0 RTP/AVP 31 34 96 97>>>>>>>>>>>>>>>>
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=rtpmap:97 H264/90000
a=content:slides
a=inactive>>>>>>>>>>>>>>>>>>>>>>>
m=application 0 RTP/AVP 96
a=rtpmap:96 H224/0
a=inactive
m=application 0 UDP/UDT/IX *

llobregatil · ‎07-04-2016

Hi Deepak,

Sorry for delay, I set the configuration according annex but the problem persists, can you help me?

Trace logs.

Calling - 7905

Called - evandrobucci

Deepak Mehta · ‎07-04-2016

Thanks , i will go through the traces again however can you make sure MTP is configured as "codec pass-through " in the IOS router.

This is needed for video calls(pass through supports H264,BFCP,FECC).

rfurukawasr3 · ‎07-07-2016

Thanks for help us Deepak Mehta !

After some changes in options on CUCM, we can see "VideoOutgoing" on calls status from "SX10-Teste" to Skype4B but no video yet, today we'll make other tests with Microsoft team. But calls from another way (Skype4b VIS -> VTC Cisco) we see only audio packets! We have some doubts about this integration:

The video calls are possible only from Cisco to Skype (one way)? Its one of limitations about this integration? I found this information bellow on (https://insidemstech.com/tag/vis-and-vtc/).

"VIS only supports one way call from VTCs to VIS, calls from VIS to VTS is not supported"

Thanks !

Deepak Mehta · ‎07-07-2016

As per my understanding If we are using H.264 on both sides, which i can see is the case then it should work , VIS doesn’t support trans coding of video streams .

Can you share the trace again with calling and called number and time of calls.thanks

rfurukawasr3 · ‎07-21-2016

Now the video calls are OK! We try to uncheck option "MTP required" on SIP Trunk (like Microsoft documents said), and we saw source IP for call was "SX10 Codec" in SB4 log's Server, we see some "deny" by ACL's for this source, after permit that way the calls is working!

But after search about limitations in this integration a lot of pages, we saw a lot of comments: " S4B VIS only supports one way call from VTCs Cisco to SB4 VIS, calls from SB4 VIS to VTCs Cisco is not supported" ! That's correct , the video calls and screen sharing works only when Cisco call to Skype user !

Suresh Hudda and Deepak Mehta Thanks for help us !!!