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Unified Communications Open Interoperability Program – Office Communications Server

Find out more about the Microsoft Unified Communications Open Interoperability Program for enterprise telephony infrastructure, including finding qualified SIP-PSTN gateways, IP-PBXs and SIP Trunking Services.

This page details infrastructure and services qualified with Office Communications Server 2007 and Office Communications Server 2007 R2. For information on Lync Server 2010, please see Unified Communications Open Interoperability Program for Lync Server 2010.

Overview

The qualification program for SIP/PSTN Gateways, IP-PBXs and SIP Trunking Services ensures that customers have seamless experiences with setup, support, and use of qualified telephony infrastructure and services with Microsoft's unified communications software and Microsoft Office Communications Online (BPOS-Dedicated).

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.

To find out more about how Microsoft recognizes the expertise and impact of partners in the technology marketplace, visit the Microsoft Partner Program site.

Vendor Process

If you are interested in joining the Microsoft Unified Communications Open Interoperability Program, please go to the Vendor Process for Microsoft Unified Communications Open Interoperability Program page and follow instructions to join and participate in the program.

Direct SIP: Gateways and IP-PBXs Qualified for Office Communications Server

Listed below are gateway and IP-PBX products and necessary firmware combinations that have been independently qualified. It is recommended that you visit the vendor's Web site for the latest information regarding PSTN/PBX, protocol, capacity, country support and documentation including a Quick Start Guide, release notes and known issues.

— "Qualified" +S —"Qualified with SRTP & TLS"

VendorConfigurationTested ProductCommunications Server Version
2007 R22007
AastraIP-PBXMX-ONE V.4.0+S 
AculabBasic HybridApplianX Gateway for Office Communications Server 2007, V1.0.0    
AltigenIP-PBXMAXCS, 6.5.0.1000 
AudioCodesBasic GatewayMediant 1000, 5.60A.013.005+S
AudioCodesBasic GatewayMediant 1000, 5.20A.043 
AudioCodesBasic GatewayMediant 2000, 5.60A.013.005+S
AudioCodesBasic GatewayMediant 2000, 5.20A.043 
AudioCodesBasic HybridMediant 1000 Hybrid, 5.60A.013.005+S
AudioCodesBasic HybridMediant 1000 Hybrid, 5.20A.043 
AudioCodesBasic HybridMediant 2000 Hybrid, 5.60A.013.005+S
AudioCodesBasic HybridMediant 2000 Hybrid, 5.20A.043 
AvayaIP-PBXAvaya Aura Session Manager 5.2 with Avaya Aura Communication Manager 5.2.1 SP1 
AvayaIP-PBXCS 1000, 5.50.12+S
AvayaIP-PBXCS 1000, 6.00R / 6.00.18+S
AvayaBasic HybridSecure Router 4134, 10.2.1 
AvayaBasic HybridSecure Router 4134, 10.1.0 
AvayaBasic GatewaySecure Router 2330, 10.2.1 
CiscoBasic Gateway**   2851 Integrated Services Router, IOS 12.4(15)T 
CiscoBasic Gateway**2851 Integrated Services Router, IOS 12.4(24)T 
CiscoBasic Gateway**3845 Integrated Services Router, IOS 12.4(15)T 
CiscoBasic Gateway**3845 Integrated Services Router, IOS 12.4(24)T 
CiscoIP-PBXUnified Communication Manager 7.1.3 
CiscoIP-PBXUnified Communication Manager 8.0.1 
DialogicBasic GatewayDMG2000, 6.0.128+S
DialogicBasic GatewayDMG2000, 5.1.142 
DialogicBasic HybridDMG4000, 1.5.102 
DialogicBasic HybridDMG4000, Dialogic Diva SIPcontrol 2.1.0.33+S
Ferrari electronic AGBasic GatewayOfficeMaster Gate, 3.1 
Ferrari electronic AGBasic GatewayOfficeMaster Gate, 3.2+S
Ferrari electronic AGBasic HybridOfficeMaster Hybrid Gate, 3.2+S
GenbandIP-PBXCS2000/C20, CVM14 
Huawei TechnologiesIP-PBXSoftCo, V100R002
InnovaphoneIP-PBXIP6000, V7.00 ocs-certified 09-7034301 
Media5Basic GatewayMediatrix 3000 Series, DGW 2.0 
MitelIP-PBX3300, 9.0.0.41 
NECBasic GatewaySV70 OCS-GW-A, MG-16SIPC 
NETBasic GatewayVX1200, 4.7v88+S
NETBasic GatewayVX1200, 4.4.2.v31 
Nuera CommunicationsBasic GatewayGX-1K, 5.60A.013.005+S
Nuera CommunicationsBasic GatewayGX-1K, 5.20A.043 
Nuera CommunicationsBasic GatewayGX-2K, 5.60A.013.005+S
Nuera CommunicationsBasic GatewayGX-2K, 5.20A.043 
PattonBasic GatewaySmartNode 4600 Series Gateways, R5.6 and newer 
QuintumBasic GatewayTenor DX, P107-06-00-OCSR2-03
QuintumBasic GatewayTenor DX, P105-19-10-MS-01 
QuintumBasic HybridTenor Hybrid Gateway 60, P107-06-00-OCSR2-03
SeltatelIP-PBXSAMoffice 4, 2.8.0 
Tango NetworksBasic GatewayAbrazo (qualified with Audiocodes Mediant 2000 5.20A.043), 3.3 
TeldatBasic GatewayVyda-2M 10.7.55+S 
VegaStreamBasic GatewayVega400, 8.282S029 

** See partner’s site for known issues and support notes

Other Supported Products

Other supported products are listed by request of the Vendor as the same qualified firmware may be supported across several different products. While these have not been specifically tested, the Vendor does fully support this configuration for the listed Qualification Level. Please contact the Vendor for more information on these products.

Gateways or IP-PBXsOther Supported Products
AudioCodes Mediant 1000MediaPack 11x, Mediant
Cisco 2851 Integrated Services Router   2800 Series
Cisco 3851 Integrated Services Router3800 Series
Dialogic DMG2000DMG1000 and DMG2000 Series
Dialogic DMG4000Dialogic 4000 Media Gateway Series and Dialogic Diva SIPcontrol
Mediatrix 3000 SeriesMediatrix 4100 Series, Mediatrix 4400 Series
NET VX1200VX Series, VX900, VX1800
Patton SmartNode SeriesSmartNode SN4940/50/60, SN45xx, SN49/44/4300 and SN5400 Series
Quintum Tenor DXTenor AS, AF, AX, BX, DX and CMS Series
Teldat Vyda-2MVyda Series and Atlas Series
VegaStream Vega400Vega50 Europa Vega 5000

SIP Trunking Services Qualified for Microsoft Office Communications Server 2007 R2

SIP Trunking enables connectivity to the Public Switched Telephony Network (PSTN) directly over SIP. SIP Trunking services are offerings from IP Telephony Service Provider partners that offer PSTN origination, termination and emergency services using the SIP protocol. An enterprise can use SIP Trunking to connect their on-premise voice network implemented by Microsoft Office Communications Server 2007 R2 or to provision PSTN termination capability for Office Communications Online (BPOS-Dedicated).

Listed below are SIP Trunking Services that have been independently qualified to meet the UCOIP requirements along with those services who meet the additive requirements for Office Communications Online (BPOS-Dedicated).

CarrierService NameOffice 365 Dedicated
AT&TAT&T IP Flexible Reach ServiceQualified for Office Communications Online (BPOS-Dedicated)
BT Global ServicesBT Onevoice 
EtherspeakLyncTrunk 
Global CrossingSIP Trunking ServicesQualified for Office Communications Online (BPOS-Dedicated)
IntelePeerIntelePeer SIP TrunkingQualified for Office Communications Online (BPOS-Dedicated)
InterouteInterouteOneQualified for Office Communications Online (BPOS-Dedicated)
IP DirectionsOCS Telephony Services 
JajahJAJAH SIP Trunking 
Level 3Level 3 Communications SIP Trunking 
Orange Business ServicesSIP Trunking 
PaetecPaetec SIP Trunking Services 
SotelSoTel IP Services 
SprintSprint Global MPLS, SIP Trunking   Qualified for Office Communications Online (BPOS-Dedicated)
SwisscomSwisscom VoIP Gate 
TelenorTelenor Samordnet kommunikasjon
(Unified Communication)
 
ThinkTelOCS Connect 
Verizon Business
IP Trunking ServicesQualified for Office Communications Online (BPOS-Dedicated)

Supported IP-PBXs for Microsoft Office Communications Server

The following IP-PBXs are supported by Microsoft but have not gone through the formal OIP qualification process nor was the testing requested by the vendor. Sufficient internal testing has been performed by Microsoft such that specific configurations are supported by Microsoft (where applicable with known limitations). These configurations utilize the commercially available production SIP trunk interface of the IP-PBX vendor but may not be supported by the IP-PBX vendor. In addition, IP-PBX vendor-provided complete documentation for installation and set-up, release notes, or documented support processes may not be available. Wherever possible, Microsoft will endeavor to provide documentation for installation and set-up.

IP-PBX VendorTested ProductSupported ConfigurationSoftware Versions Tested2007 R22007
Avaya
Communications Manager SIP Enablement ServicesDirect SIP4.0 

Known Limitations:

  • 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.
Cisco
Cisco Unified Communications ManagerDirect SIP4.2(3)_SR3a
4.2(3)_SR4b 
5.1(1b)
5.1(3e) 
6.1(1b)
6.1(3a) 

Known Limitations:

  • 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.
Siemens
Enterprise
Communications
OpenScapeDirect SIP3.1R3 

Known Limitations:

  • 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.

Dual Forking Qualified for Microsoft Office Communications Server

Listed below are IP-PBX and firmware combinations that have been independently qualified. Please contact the vendor for more information on these products.

IP-PBX Vendor   Tested Product   Qualification Level   Software Version TestedOther Supported Products   2007
NortelCS 1000

Dual Forking***

Dual Forking
with RCC***

Call Server X2105.00W

Signaling Server 5.00.31

Multimedia Communications Manager 3.0.1.77   

 

CS 1000 Series

*** If you are deploying Dual Forking, you must also install the following updates for Microsoft Office Communications Server 2007:

Qualified Gateways for Exchange Server Voice Mail

The current list of supported gateways is maintained here.

Gateway vendors interested in being listed on this page please follow the vendor process for Microsoft Unified Communications Open Interoperability Program mentioned above.

 

Qualified PBXs for Exchange Server Voice Mail

The current list of supported IP-PBXs is maintained here.

IP-PBX vendors interested in being listed on this page please follow the vendor process for Microsoft Unified Communications Open Interoperability Program mentioned above.

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