Cisco TelePresence

• Jabber Video will display the video from the MCU in the wrong aspect ratio when using the H.263 codec. Under normal circumstances H.264 or H.263+ is used in preference to H.263. (CSCtx91864)

No issues found.

A number of Conductor XC2.2 features are not supported in version 4.3

Tested with 4.3(2.32): A number of Conductor features are not supported in version 4.3.

• Audio may be distorted if G722.1 codec is negotiated. (CSCty01044)
• Audio may be distorted on low bandwidth calls. (CSCtx91864)

No issue found

Tested to work with the Cisco MCU 4500 series registered to VCS version X7.2, X7.1, or X7.0, if the system running TX software is registered to Cisco Unified Communications Manager Version 9.1(1) or 8.6(2a)SU3.

For more information about products that are compatible with systems running CTS 1.10 software, refer to the Cisco TelePresence System Software Version Compatibility - 1.10 Releases document at the following URL: http://www.cisco.com/en/US/docs/telepresence/cts_admin/1_10/compatibility/cts_compatibility_1_10.html

No issues

SIP and H.323

No issues

Tested to work with the Cisco MCU 4500 series registered to VCS version X7.2, X7.1, or X7.0, if the system running TX software is registered to Cisco Unified Communications Manager Version 9.1(1) or 8.6(2a)SU3.

For more interoperability information about products running TX6.0 software, refer to the Cisco TelePresence TX System Software Version Compatibility and Interoperability with Other Devices—6.0.x Releases at the following URL: http://www.cisco.com/en/US/docs/telepresence/tx_sw/6_0/compatibility/tx_sw_compatibility_6_0.html

• If a call is made between an endpoint and an MCU, with the endpoint using SIP and the MCU using H.323, and a second endpoint joins the conference and sends a content stream (any protocol) to the MCU, the first endpoint display will freeze or go black when the second endpoint stops streaming content. This is because the first endpoint is not aware that streaming has ended and is expecting to receive further content.
• An endpoint, for example CTS, may end up in a half encrypted call where the call leg from CTS > CUCM > VCS is unencrypted and the call leg from VCS > MCU is encrypted.

• Due to inaccurate timestamps sent by this endpoint, lip synchronisation cannot be guaranteed.Bandwidth must be set to 768k or lower in order for encryption to work.

• The MCU does not support FECC with this endpoint when calling using SIP.

• After using hold and resume, CTS endpoints connected via CUCM do not receive audio from the MCU (CSCtr60976).
• In networks in which not all call legs are encrypted, calls may drop after using hold and resume. When using encryption it is recommended that all call legs are encrypted (CSCtr52491).

• Due to inaccurate timestamps sent by this endpoint, lip synchronisation cannot be guaranteed.
• The MCU does not support SIP with this endpoint.

• Encrypted SIP calls are not supported between the MCU and this endpoint (CSCtx91859).
• SIP content is not supported between the MCU and this endpoint.
• G.722.1 Annex C is not supported between the MCU and this endpoint.

• In a conference which is sending a content stream and the content stream changes to a different content provider, endpoints will receive data in a different codec. In the case of the Polycom HDX, the endpoint ignores all incoming packets. This also occurs if the content provider itself changes codecs, for example, if the computer providing the content changes resolution. (CSCtw49873 / Polycom: VIDEO-93014)

• Video negotiation with the MGC can fail when using H.264 in calls of 384kbps or less. To avoid this either use a higher bandwidth or disable H.264 when dialling the MGC using custom codec selection. (CSCts46389)

• Due to inaccurate timestamps sent by this endpoint, lip synchronisation cannot be guaranteed.
• In some circumstances this endpoint can fail to display video during SIP calls.
• On low bandwidth H.323 calls, the endpoint may not be able to decode audio.

• Encrypted SIP calls are not supported between the MCU and this endpoint.

• Audio between the MCU and this endpoint is not supported on interworked calls if “Audio Mode:” setting on the endpoint is set to either “MPEG4 Audio” or “Auto”.

Tested H.323 and H.323-SIP interworking via VCS. No issues found.

• Does not support two-way video at less than 410kbps.
• The Sony HG-90 only supports video in 1280x720 resolution. If the HG-90 is dialled in to a conference on the MCU, it will become an audio only call if the MCU is set to transmit video in 4:3 ratio only.

• H.323 call to an encryption-required conference results in an audio only call.
• The MCU does not support FECC with this endpoint when calling using SIP.