04-06-2015 03:58 AM - edited 03-17-2019 02:33 AM
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,
04-06-2015 06:43 AM
Hi Mohammed,
Could please attach call flow and scheme?
Regards
04-06-2015 10:17 PM
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,
04-06-2015 11:03 PM
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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