Only products that meet rigorous and extensive testing requirements and conform to the specifications and test plans will receive qualification.

While the specifications are based on industry standards, this program also defines:

• Specific requirements for interoperability with Office Communications Server & Exchange Server Voice Mail
• Specific requirements for interoperability with Office Communications Online Services for SIP Trunking Service Providers
• Testing requirements for qualifying interoperability with Office Communications Server & Exchange Server Voice Mail
• Installation, set-up and configuration requirements via a Quick Start Guide
• Release Notes with any known issues
• Documented support process between Microsoft and the vendor
• Enterprise-class standards for audio quality, reliability, and scalability

The scope of the qualification is for environments where either Office Communications Server or Exchange Server Voice Mail utilizes a SIP/PSTN Gateway, IP-PBX or SIP Trunking Service for communication with the PSTN. Additionally, qualification is available for Microsoft Office Communications Online Service environments utilizing Office Communications Server with Exchange Online Unified Messaging.

The testing focus of the program is designed to ensure that vendors providing interoperability with Microsoft unified communications solutions do so in a consistent and supportable manner, including SIP and signaling support used with the Mediation Server role of Office Communications Server and the Voice Mail role of Exchange Server.

• Configuration requires setting “Alternate Route Timer(sec)” value from default of 10 sec to 30 sec. The configuration should show “Alternate Route Timer(sec): 30″ in the corresponding SIP signaling group.
• When an call is ringing to the Office Communicator user, the caller (either on an Avaya station or a PSTN line routed through the PBX) will not get ring back tone. This issue has been resolved by Avaya with the 5.x software releases.
• Quality of Experience reports will not contain information regarding jitter and packet loss.
• Comfort noise generation is not supported. As a result, comfort noise is not played on Office Communicator.
• ISDN Failover is not supported from an OCS perspective. If the Avaya PBX is being used for PSTN connectivity and multiple T1′s are being utilized, an OC client will not retry a call based on a T1 being unavailable. It may be possible to configure the Avaya to not assign outbound calls from OCS to an unavailable T1, but this configuration was not tested.

• The PRACK message sent by CUCM 4.2(3) is malformed by missing the MAXFORWARDS header.  As a result, this configuration requires PRACK to be disabled.  By default, PRACK is disabled in CUCM 4.2(3)
• For Office Communications Server 2007, this support requires update package for Communications Server 2007 Mediation Server: August 2008.
• OCS 2007 may not appropriately normalize the PAI in the 200 OK or UPDATE, resulting in OC displaying a non RFC3966 formatted global number and in failed RNL on OC. When calling from OC 2007 to a Cisco phone number, after the caller gets connected, the name of the person on the Cisco phone may not be shown on Communicator, and instead OC may display the E.164 number (without a “+”) for the person on the Cisco phone. This is resolved in OCS 2007 R2
• When calling from OC 2007 to a Cisco phone number, where the Cisco extension is disconnected or out of service, the Cisco IP-PBX may not notify OC 2007 in a timely manner. This has been remediated in OCS 2007 R2.

• Inbound early media/PRACK is not supported on the Siemens PBX. As a result, in some situations the initial audio of a call may be clipped as the call signaling is being set up.
• Certain Siemens IP phones may not render audio for inbound calls. If this occurs, a phone configuration change to support both symmetric and asymmetric RTP will be needed.
• For Hold/Un-hold to function properly, OpenScape needs the RTP config parameter Srx/Sip/ZeroIpOnHold set to false.
• For full SDP versioning support, OpenScape needs the RTP config parameter Srx/Sip/CompareSdpBody set to true.
• Quality of Experience reports will not contain information regarding jitter and packet loss for PSTN calls coming through the Siemens IP PBX.