Cisco TelePresence

Tested SIP, H.323, CUCM to VCS/Conductor SIP interoperability and CUCM to VCS H.323 interworking.

• Using hold/resume functionality on a Cisco IP Video Phone E20 stops the endpoint from receiving content from the Cisco TelePresence MCU or Cisco TelePresence Server. The endpoint receives content in the main video. (Issue identifier CSCty98376)

Tested SIP and H.323 to SIP interworking.

• Transition from video to content or content to video may lead to transient video issues. (Issue identifier: CSCud64312)

Tested SIP and H.323 to SIP interworking.

• In some cases increasing resources for a call will result in low frame rate from the endpoint. (Issue identifier CSCud79739)

Tested CUCM to VCS H.323 interworking and CUCM to VCS/Conductor SIP interoperability. No issues found.

Found with TelePresence Server 3.0(2.46). Fast update requests may not be forwarded by Conductor to endpoints in Cisco UCM deployments with Conductor versions prior to XC2.0.2(CSCue89279).

No issues found.

Tested H.323 and SIP interoperability.

• Audio only calls to a TCS will not be recorded.

CSCud79767: Content from CUCM-registered C-series endpoints can take 10 seconds to arrive at TS. Fixed in TS 2.42

Tested TIP interoperability:

• When on a secure call using TLS, using audio add-in from a CTS endpoint is not supported in this release, and will cause the call to fail. While in that state, placing the call on hold and then resuming it resolves this issue. Audio add-in is supported for non-secure calls. (Issue identifier CSCub87163)
• In some circumstances, 1080p video to a Cisco TelePresence System endpoint may show some artefacts. The issue is more likely to occur when using a mixture of 1080p and lower resolution endpoints in a conference. (Issue identifiers CSCuf07863 and CSCuf03190). The video artefacts occur less frequently at 1080p if the endpoint is running software version 1.10, and do not occur if the TelePresence Server is sending 720p to the endpoint. This issue has been observed on Cisco TelePresence System 500-37, Cisco TelePresence 1000, Cisco TelePresence 1100, Cisco TelePresence System 1300 65, and Cisco TelePresence System 3000 Series endpoints.

Tested CUCM to VCS H.323 interworking.

• Cisco recommends that you do not interwork CTS calls. Under rare circumstances video issues may be encountered.

Tested CUCM to VCS/Conductor SIP, CUCM to VCS H.323 interworking, SIP, and H.323 interoperability.

• Under some circumstances there is an echo from the EX90 when sending it stereo audio when on speaker. To work around this, turn off the directional audio on the TelePresence Server or use a headset. (Issue identifier CSCud79705)
• When EX60 endpoints connect to the TelePresence Server via a VCS which is interworking the calls from SIP to H.323, then sending content to that EX endpoint will fail. To work around this issue use either H.323 or SIP throughout the call to avoid interworking. (Issue identifier CSCud81537)

Increasing resources for a call that leaves the TS as H.323 and is interworked by the VCS to SIP can cause the call to fail. (Issue identifier CSCud64471)

Tested H.323 interoperability.

• PIN entry and layout changing will not function as the T1 and T3 do not support DTMF/ FECC to theTelePresence Server.
• Note that some T1 and T3 functionality is not supported when the TS operates in Remotely Managed mode. For further details, refer to page 2 of the v3.0 release notes.

 

No issues found.

The following features are not supported when interoperating with this version of CUCM:

• Autodetection of CTS endpoints
• Trunking to CUCM

No issues found.

Calls to CTS or TX endpoints running software version 1.8 or higher may fail due to defect CSCtr70893, particularly in encrypted calls. This is resolved in Cisco Unified Communications Manager software revision 8.6.2.21900-5.

No issues found.

No issues found.

Tested CUCM to VCS/Conductor SIP interoperability.

• Video optimization will not work in interworked calls. (Issue identifier CSCtx16122)

Tested CUCM to VCS/Conductor SIP interoperability and CUCM to VCS H.323 interworking.

• CUPC calls will use low video resolutions. (Issue identifier CSCud81937)

Tested CUCM to VCS/Conductor SIP interoperability; no issues found.

Tested H.323 and SIP interoperability:

• TLS encrypted SIP calls are not supported between this endpoint and the TelePresence Server. (Issue identifier CSCtx91859)
• The endpoint does not support SIP content.
• H.263 codec is not supported between this endpoint and the TelePresence Server, H.264 is enabled by default.
• H.261 codec is not supported by the endpoint in SIP calls (not default codec), use H.264 codec (enabled by default).
• SIP calls out from the TelePresence Server to a LifeSize endpoint may exhibit garbled audio, in both directions, if the calling parties use the G.722.1.C codec. This is unlikely to occur unless the call cannot fall back on another codec. Try using a different codec to work around this issue.

Tested H.323 interoperability.

• SIP calls to the TelePresence Server from this endpoint are not supported.

Tested SIP and H.323 interworking interoperability, no issues found.

Tested H.323 and SIP interoperability.

• HDX does not support redial missed/rejected calls due to an endpoint limitation.

Tested H.323 and CUCM to VCS/Conductor SIP interoperability:

• OTX does not support redial missed/rejected calls due to an endpoint limitation.
• Low bandwidth calls are not supported

Tested H.323 and SIP interoperability.

• VVX does not support redial missed/rejected calls due to an endpoint limitation.

Tested H.323 and SIP interoperability:

• XT1000-series will adversely alter aspect ratio of H.263+ video (not default codec). H.264 is used by default.
• Dialling out to the endpoint from the TelePresence Server may result in content not working. Dialling in works around this issue. (Issue identifier CSCtz21165)
• Making a 60fps call at less than 2 Mbps is not supported. As a work around, use a higher bandwidth or disable 60fps on the endpoint.

Tested H.323 interoperability:

• At low bandwidths this endpoint may not handle audio properly. You can mitigate this by disabling AAC codec for this endpoint.