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VIDEO CALLING NOT WORKING

Dear Experts,

 

I have some issue about Video calling, from Side A to Side B 

From Side A I am able to see the phone sending the streams to the other end correct IP and also receiving the streams by the other end.

But the 9971 phone I see the phone sending audio and video streams but unable to receive from the other end. Audio call was working before with the MTP required option checked, because once MTP is invoked for the call, the phones will stream to the MTP IP address and not directly to the other phone. however if we check that option video will not work.

 

Best Regards,

 

 

3 Replies 3

Leo Salcie Tejeda
Rising star
Rising star

Hi Mohammed,

Could please attach call flow and scheme?

 

Regards

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Dear Leo,

Thanks for your support, Please find the below captures during troubleshooting.

pinging both sides is ok no issue Audio calling no issue,except video calling, there is no port restrictions from the Firewall. do we need to do something from Service provider Router.

 

02882727.002 |11:28:16.533 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.4.0.16 on port 52147 index 33 with 1833 bytes:

[35858,NET]

INVITE sip:7080@10.4.7.1;user=phone SIP/2.0

Via: SIP/2.0/TCP 10.4.0.16:52147;branch=z9hG4bK4b0f9acf

From: "9888" <sip:9888@10.4.7.1>;tag=a45630ba4c7b000b7762fd18-75fe34c5

To: <sip:7080@10.4.7.1>

Call-ID: a45630ba-4c7b0004-623a23fb-00278e48@10.4.0.16

Max-Forwards: 70

Date: Thu, 19 Mar 2015 08:28:16 GMT

CSeq: 101 INVITE

User-Agent: Cisco-CP9951/9.4.1

Contact: <sip:e8c3d692-a51f-1192-12bf-0f6dd9e28382@10.4.0.16:52147;transport=tcp>;video

Expires: 180

Accept: application/sdp

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO

Remote-Party-ID: "9888" <sip:9888@10.4.7.1>;party=calling;id-type=subscriber;privacy=off;screen=yes

Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-8.0.1

Allow-Events: kpml,dialog

Content-Length: 775

Content-Type: application/sdp

Content-Disposition: session;handling=optional

 

v=0

o=Cisco-SIPUA 5137 0 IN IP4 10.4.0.16

s=SIP Call

t=0 0

m=audio 28354 RTP/AVP 0 8 18 102 9 116 124 101

c=IN IP4 10.4.0.16

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:102 L16/16000

a=rtpmap:9 G722/8000

a=rtpmap:116 iLBC/8000

a=fmtp:116 mode=20

a=rtpmap:124 ISAC/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

m=video 21762 RTP/AVP 126 97

c=IN IP4 10.4.0.16

b=TIAS:1000000

a=rtpmap:126 H264/90000

a=fmtp:126 profile-level-id=42801E;packetization-mode=1;level-asymmetry-allowed=1

a=imageattr:* recv [x=640,y=480,q=0.50]

a=rtpmap:97 H264/90000

a=fmtp:97 profile-level-id=42801E;packetization-mode=0;level-asymmetry-allowed=1

a=imageattr:* recv [x=640,y=480,q=0.50]

a=sendrecv

 

CUCM responds with a trying

 

02882733.001 |11:28:16.572 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.4.0.16 on port 52147 index 33

[35859,NET]

SIP/2.0 100 Trying

 

Digit analysis happens in CUCM and trunk is located to route the call out

 

02882754.013 |11:28:16.617 |AppInfo  |Digit analysis: analysis results

02882754.014 |11:28:16.617 |AppInfo  ||PretransformCallingPartyNumber=9888

|CallingPartyNumber=9888

|DialingPartition=

|DialingPattern=7XXX

|FullyQualifiedCalledPartyNumber=7080

 

H323 setup sent to the CUCM cluster 8 and the other end responds with a proceeding and alerting

 

02882790.010 |11:28:16.728 |AppInfo  |Out Message -- H225SetupMsg -- Protocol= H225Protocol

 

02882792.002 |11:28:16.765 |AppInfo  |In  Message -- H225CallProceedingMsg -- Protocol= H225Protocol

 

02882800.002 |11:28:16.821 |AppInfo  |In  Message -- H225AlertMsg -- Protocol= H225Protocol

 

CUCM 10 in turn sends ringing to the originating phone

 

02882827.001 |11:28:16.832 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.4.0.16 on port 52147 index 33

[35860,NET]

SIP/2.0 180 Ringing

Via: SIP/2.0/TCP 10.4.0.16:52147;branch=z9hG4bK4b0f9acf

From: "9888" <sip:9888@10.4.7.1>;tag=a45630ba4c7b000b7762fd18-75fe34c5

To: <sip:7080@10.4.7.1>;tag=11809~2db5803f-12ff-4fb8-a736-43acd13857cb-31501439

 

Connect message from trunk for this call

 

02882840.002 |11:28:18.977 |AppInfo  |In  Message -- H225ConnectMsg -- Protocol= H225Protocol

 

After this both sides negotiates capabilities and then the CUCM sends 200 OK to the originating phone with the SDP details provided by the the other end

 

02883035.001 |11:28:19.525 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.4.0.16 on port 52147 index 33

[35862,NET]

SIP/2.0 200 OK

Via: SIP/2.0/TCP 10.4.0.16:52147;branch=z9hG4bK4b0f9acf

From: "9888" <sip:9888@10.4.7.1>;tag=a45630ba4c7b000b7762fd18-75fe34c5

To: <sip:7080@10.4.7.1>;tag=11809~2db5803f-12ff-4fb8-a736-43acd13857cb-31501439

Date: Thu, 19 Mar 2015 08:28:16 GMT

Call-ID: a45630ba-4c7b0004-623a23fb-00278e48@10.4.0.16

CSeq: 101 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= 1-197; isVoip; call-instance= 1

Send-Info: conference, x-cisco-conference

Remote-Party-ID: "mohammed " <sip:7080@10.4.7.1>;party=called;screen=yes;privacy=off

Contact: <sip:7080@10.4.7.1:5060;transport=tcp>

Content-Type: application/sdp

Content-Length: 360

 

v=0

o=CiscoSystemsCCM-SIP 11809 1 IN IP4 10.4.7.1

s=SIP Call

c=IN IP4 10.2.7.39

b=TIAS:384000

b=CT:384

b=AS:384

t=0 0

m=audio 27632 RTP/AVP 9 101

a=ptime:20

a=rtpmap:9 G722/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

m=video 26422 RTP/AVP 97

b=TIAS:320000

a=rtpmap:97 H264/90000

a=fmtp:97 profile-level-id=42001E

a=content:main

 

 

Thanks,

Hello,

Refer this document for the configuration:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/feature/guide/Video-trans-conf.html

Also make sure that you have enabled the video camera on the IP Phone

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