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Terminologies

 

  • Frame: A video is a series of images played at a very high rate for human eye to recognize it as continuous motion. These images are called video frames. The higher the frame rate, the better the quality
  • Resolution: This defines the images size in term of number of pixels
  • Compressed Video Frames: These are results of compressing the original video frames to reduce the overall size of the video. Encoding takes place after compressing frames
  • Video Compression Format (Codec): This defines the encoding of the analog video into digital format

 

Basics

 

When using CUCM for video calls, the following is supported:

 

  1. Video Streaming and Video features such as Far-end Camera Control (FECC)
  2. SCCP, H323, and SIP Signaling
  3. CDRs generated for video calls
  4. E-LCAC is supported for video calls

 

Video calls include multiple RTP streams in each direction. The call can include the following stream types:

 

  • Audio RTP Stream based on audio codecs
  • Video RTP Stream based on video codecs
  • Far-end camera control (FECC)  RTP Stream based on video codecs
  • BFCP RTP Stream based on video codecs

 

Note: Each RTP stream will use different port which ranges between 16384 and 32767

 

Call control for video calls operate the same way as the call control for audio calls using the same set of messages. The difference that we will have additional attributes in the SDP or TCS messages indicating video parameters.

Some of the endpoints that support video calls:

 

  • Cisco Unified IP Phone 9971
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 8941
  • Cisco Unified IP Phone 8945
  • Cisco Unified IP Phone 7985
  • Telepresence Endpoint (General)

 

Codecs

 

Common Video Codecs are:

 

  • H261 (old codec)
  • H263/H263+ (newer codec for IP Video)
  • H264/AVC (high quality video codec)
  • Cisco VT (Fixed Rate Codec)

 

The bandwidth for video stream can = # pixels per frame x 3 colors per pixel x 8 bits per color x # frames per second

 

It is very important to note that unlike audio RTP, video RTP bandwidth is variable during the same call due to many reasons such as static backgrounds, different colors, etc.

 

From CUCM side, the bandwidth of video calls equals the sum of the audio bandwidth and the video bandwidth. The total bandwidth does not include L2 overhead.

 

Example

 

A 384-kb/s video call may comprise G.711 at 64 kb/s (for audio) plus 320 kb/s (for video). This sum does not include overhead. If the audio codec for a video call is G.729 (at 24 kb/s), the video rate increases to maintain a total bandwidth of 384 kb/s. If the call involves an H.323 endpoint, the H.323 endpoint may use less than the total video bandwidth that is available. Regardless of protocol, the endpoint may always choose to send at less than the max bit rate for the call.

 

H.261 and H.263 codecs exhibit the following parameters and typical values:

 

  • Bit rates range from 64 kb/s to a few mb/s. These bit rates can exist in any multiple of 100 b/s. H.261 and H.263 can function with bit rates lower than 64 kb/s, but video quality suffers in such cases.
  • Resolution:
    • One-quarter Common Interchange Format (QCIF) (Resolution equals 176x144.)
    • Common Interchange Format (CIF) (Resolution equals 352x288.)
    • 4CIF (Resolution equals 704x576.)
    • Sub QCIF (SQCIF) (Resolution equals 128x96.)
    • 16CIF (Resolution equals 1408x1152.)
    • Custom Picture Format
  • Frame Rate: 15 frames per second (fps), 30 fps
  • Annexes: F, D, I, J,K, L, P, T, N

 

Enable Audio-Only Device for Video Call

 

You can install Cisco Unified Video Advantage (CUVA) on a PC which is connected to non-video phone. This will make the PC act as camera device for the phone and provide video calling. This association can occur before a call is made or during a call (mid-call).

 

This feature is supported for both SIP and SCCP phones

Comments
Samuel T Mathai
Level 1
Level 1

I got new cisco 8865 phones and tried to make calls through cube & its failed. It works when I disabled video on phone. I see this messages on debugs.

 

 

SIP: Trying to parse unsupported attribute at media level
SIP: Trying to parse unsupported attribute at media level
SIP: Trying to parse unsupported attribute at media level
SIP: Trying to parse unsupported attribute at media level

 

 

martyn.rees
Level 4
Level 4

On the SIP trunk between CUCM and the CUBE do you have "Retry Video Call as Audio" selected? If not you will need to enable this and then the call will fail back to audio only if that is all that is supported.

What version of IOS are you running on the CUBE?

Hi,

 

On CUBE, try to put the commands

voice service voip

sip

pass-thru headers unsupp

pass-thru content unsupp

Samuel T Mathai
Level 1
Level 1

tried both recommended.. same..

Samuel T Mathai
Level 1
Level 1

call flow is like this.

CUCM-> CUBE-> ITSP

Attaching SIP messages from cube.

Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.199.223.98:5060;branch=z9hG4bK106618895C
From: "Frank Gully" <sip:71XXXX34@10.250.0.6>;tag=3461F888-18B7
To: <sip:83XXX17@10.250.0.6>;tag=1573359561-1444680627953
Call-ID: 1D77E85B-705411E5-ABA69F53-9A6EB380@64.199.223.98
CSeq: 101 INVITE
Timestamp: 1444680627
Require: timer
Session-Expires: 1800;refresher=uas
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported: timer
Accept: application/media_control+xml,application/sdp
Contact: <sip:83XXXXX17@10.250.0.6:5060;broadworks=BWSIG2i095m8o2or176;transport=udp>
Content-Type: application/sdp
Content-Length: 293

v=0
o=- 1462251052 1 IN IP4 10.250.0.6
s=-
c=IN IP4 10.250.0.6
t=0 0
m=audio 20370 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 0 RTP/AVP 100 126 97
a=rtpmap:100 H264/90000
a=rtpmap:126 H264/90000
a=rtpmap:97 H264/90000

SIP: Trying to parse unsupported attribute at media level
SIP: Trying to parse unsupported attribute at media level
SIP: Trying to parse unsupported attribute at media level
SIP: Trying to parse unsupported attribute at media level
113594926: Oct 12 20:10:28.110: //7040632/462601000000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:83XXXX17@10.250.0.6:5060;broadworks=BWSIG2i095m8o2or176;transport=udp SIP/2.0
Via: SIP/2.0/UDP 64.199.223.98:5060;branch=z9hG4bK106618915CD
From: "Franly" <sip:713XXX34@10.250.0.6>;tag=3461F888-18B7
To: <sip:83XXX17@10.250.0.6>;tag=1573359561-1444680627953
Date: Mon, 12 Oct 2015 20:10:27 GMT
Call-ID: 1D77E85B-705411E5-ABA69F53-9A6EB380@64.199.223.98
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


113594927: Oct 12 20:10:28.114: //7040631/462601000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.5.211.10:5060;branch=z9hG4bK6d0f1b6feb6b
From: "Frank Gully" <sip:7134356934@10.5.211.10>;tag=39952~00b439cd-2477-79c1-b23b-ba0ce4762c8d-40420577
To: <sip:83XXX17@10.5.211.250>;tag=3461F914-24F0
Date: Mon, 12 Oct 2015 20:10:28 GMT
Call-ID: 46260100-61c113b3-3ac5-ad3050a@10.5.211.10
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:83XXX17@10.5.211.250:5060;transport=tcp>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.3.3.M4
Session-Expires:  1800;refresher=uas
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 295

v=0
o=- 1462251052 1 IN IP4 10.250.0.6
s=-
c=IN IP4 10.5.211.250
t=0 0
m=audio 19088 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 0 RTP/AVP 100 126 97
a=rtpmap:100 H264/90000
a=rtpmap:126 H264/90000
a=rtpmap:97 H264/90000

113594928: Oct 12 20:10:28.114: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

PaeTec_Cyrusone#ACK sip:83XXXX17@10.5.211.250:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.5.211.10:5060;branch=z9hG4bK6d111aa061f3
From: "Franly" <sip:71XXXX34@10.5.211.10>;tag=39952~00b439cd-2477-79c1-b23b-ba0ce4762c8d-40420577
To: <sip:83XXX17@10.5.211.250>;tag=3461F914-24F0
Date: Mon, 12 Oct 2015 20:10:27 GMT
Call-ID: 46260100-61c113b3-3ac5-ad3050a@10.5.211.10
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

 

Samuel T Mathai
Level 1
Level 1

I tried to configure region from phone to SIp trunk with none fo videos.. still didnt working..

Vivek Batra
VIP Alumni
VIP Alumni

Can you please share the CUBE output of debug ccsip messages for the failed call? Logs you shared before are partial.

- Vivek

Samuel T Mathai
Level 1
Level 1

I opened tac and they told me to put zero or none configure in video bandwidth in region. it worked. Thanks.

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