Planning Voice Deployments for Microsoft Office Communications Server 2007 R2

Nireshen BeerbulSenior ConsultantMicrosoft

Scope

Voice with Office Communications Server

Enterprise Voice PlanningDeployment ScenariosSizing and Topology ConsiderationsCall Routing and Management Interoperability With Existing Telephony Infrastructure

Out of ScopeIntegrations that don’t involve Voice

Client-based plug-insPBX-based controls hosted inside CommunicatorFeatures, Experience, Capability vary widely

Remote Call ControlOffice Communicator controls a PBX station setConsider a third party Gateway instead of an upgrade: Corebridge, Genesys, Estos

Agenda

Prerequisites

Enterprise Voice Elements

Deployment Scenarios

Recommendations & Next steps

Your current state?

No voice with Office Communications Server…yetHow do I get started with a pilot?Validate the technology & business case

Pilot completed and successfulHow do I move the pilot to production?Breadth - more & different users & cases

Initial deployment completed and successfulHow do I grow production to scale?Business critical, multi-site communications

Prerequisites to Deployment

Windows Server 2003 Domain functional levelAD used to store global settings & groupsSingle-forest & multiple-forest environments

Exchange Server 2007 with Service Pack 1Unified Messaging, Missed Call Notification, Auto Attendant, Outlook Voice Access

Assess your IP Network for real-time trafficBandwidthLatency & other network affects

Your IP NetworkBandwidth

Provision for streams of 45Kbps audio, 300Kbps videoModeling calling patterns: Intra/inter-office, external

Other network affectsReduce delay: < 150ms E2E delay considered excellent (G.114)Some jitter (< 30ms) & loss (< 10%) can be handled

Prioritizing Media with DiffServAudio: Expedited Forwarding; Video: Assured ForwardingCentralized policy enforcement for Vista PCs

Controlling UsageManage the size of conferences, allowed mediaSet port range for media, per session BW limits, video quality

Your (future) Voice Network

Enterprise Voice Elements

Mediation Server: Intermediate signaling and call flowManage innovative elements of the SIP transaction: Inside, TLS/SRTP – Outside, TCP/RTPTranscode media flows from G.711 to RTAudio and SIRENAct as an ICE Client for PSTN-originated callsProvide quality metrics back to monitoring server

Upstream telephony elementsSIP/PSTN GatewayIP-PBXSIP Trunking Service

UC Open Interoperability Program (UCOIP)Qualification program for telephony infrastructure – SIP/PSTN Gateways, IP-PBXs & SIP Trunking ServiceGoal for seamless interoperability with Office Communications Server and Exchange ServerEnsure Customers have positive experiences withSetup, Support, and Use of qualified devicesAllows for scalable qualification of vendors

http://technet.microsoft.com/UCOIP

SIP/PSTN Gateways IP-PBXs SIP Trunking Service

Aculab, Audiocodes, Cisco, Dialogic, Ferrari, NEC, NET, Nortel, Nuera, Quintum, Tango Networks, Vegastream

InnovaphoneMitelNortelSeltatel

InterouteGlobal CrossingSprint

Tested IP-PBXs

PBX vendors qualify their latest versionsCustomers want support for existing versions

Where possible, Microsoft may test IP-PBXsTo date: Cisco Unified Communications Manager

Listed with qualified infrastructure on UCOIP

OCS 2007 OCS 2007 R2

CUCM 4.X 4.2(3)_SR3a 4.2(3)_SR3a4.2(3)_SR4b

CUCM 5.X 5.1(1b) 5.1(1b)5.1(3e)

CUCM 6.x 6.1(1b) 6.1(1b)6.1(3a)

Voice Deployment Continuum

Voice Capabilities with an existing IP-PBXLong-term interoperability for all end-usersCombined experience of Communicator & PBX Phone

Mixed environment with PBXSome users will be on the PBX, others will moveMix expected to change over time

Office Communications Server for VoiceEnable a temporary transition statePrepare for your PBX-less future

Deployment Scenarios

Overlay PBX: Shared Dial PlanDual Forking: PBX rings to phone & CommunicatorMultiple Dial Plans: multiple numbers per user

Networked PBX: Split Dial PlanMediation Server located behind the PBX Connect using SIP/PSTN Gateway or Direct SIP

Direct to PSTN: Owned Dial PlanCalls are sent/received directly with PSTNSIP Trunking from carrier or circuits to Gateway

Deployment Scenarios

Overlay PBX: Shared Dial Plan

Networked PBX: Split dial plan

Direct to PSTN: Owned dial plan

Overlay PBX dual forking

Calls from PSTN & PBX phones are forked by IP-PBX to Office Communications ServerAny calls made by Office Communicator are also forked to IP-PBXProvides a blended, single number experience for end-usersEach infrastructure elementimplements equivalent dial planPBX upgrade required for Direct SIP plus Dual-Forking qualification

Overlay PBXmultiple dial plans

Each user has two numbers – one for Office Communicator & one for PBXNo forking

Users configure personal call forwarding between systemsConfigure with SIP/PSTN Gateway or Direct SIP to IP-PBXIf using Gateway, trunk expense rises as size of deployment growsFundamentally a transition over time, assume PBX phone goes away

Deployment Scenarios

Overlay PBX: Shared Dial Plan

Networked PBX: Split Dial Plan

Direct to PSTN: Owned Dial Plan

Networked PBXusing a SIP/PSTN Gateway

Users are moved off the PBX Calls delivered from PSTN to PBXand routed to SIP/PSTN GatewayWide availability of Gateways for geography & circuit configurationsDialing behavior preserved for calls between all users Fast & inexpensive to deploy forpilot & smaller productionDouble-trunking through the PBX increases cost with scale

Networked PBXDirect SIP

Users are moved off the PBXCalls delivered from PSTN to IP-PBXMediation Server connects directly to SIP interface on supported IP-PBXBut still a server to server trunk – not client to client due to lack of ICE negotiation, security, etc.May require additional software, licenses or upgrades to the IP-PBXAs production deployments grow, Direct SIP has OA&M advantages

Deployment Scenarios

Overlay PBX: Shared Dial Plan

Networked PBX: Split Dial Plan

Direct to PSTN: Owned Dial Plan

Direct to PSTNIP-IP Gateways

Mediation Server sits behind same SIP/PSTN Gateway used by IP-PBXSupported configuration as long as the Gateway is qualified with OCSGateway routes based on DID or trunk group, may require configuration on the carrierSome Gateways support doing an AD Query for routing determinationIncreased flexibility and negligible impact to trunking costs when moving users from PBX

Direct to PSTNusing a SIP/PSTN Gateway

Separate PSTN interconnect infrastructure from PBXNumber routing change or new numbers provisioned by Carrier Requires zero PBX changes, eventually move trunks from PBX to SIP/PSTN GatewayInternal calls between user groups routed through PSTN

Direct to PSTNSIP Trunking

Connecting Mediation Server to SIP Trunking ServiceNo on-premise third party products (SBCs, etc.) required Uses nailed up VPN to Service Provider for securityBrings telephony trunking into datacenter consolidation strategy Still early days…not all carriers signed on to support modern (SIP Connect 1.1) standards

More OptionsAll of these can be deployed in a mixed fashionScenarios can change as deployment maturesTrunking both to IP-PBX and PSTN cloudsUse Gateways for Pilot, Direct SIP to HQ IP-PBX, keep some users on PBX but move others

Pilot Deploy Scale

User Populations to Consider

Mobile & RemoteGreat pilot users to validate capabilityDiscontinuous number range

Headquarters / Single SiteExpect a mix of Communicator & PBX for coverage of all enterprise telephony featuresMost IW users can use Enterprise Voice exclusively

Branch OfficeGateways for Least Cost Routing & Local number terminationWAN Survivability via Cell Phone & Internet

Deployment Factors to ConsiderDeployment

Goal

With Existing IP-PBX

Mixed environment

OCS for Voice

Company Size

250

1000

5000

10,000

50,000

Current Stage

No Voice Deployed

Pilot Successful

Initial Deployment Successful

User Population

Mobile & Remote

Employee

HeadquartersSingle Site

Branch Office

Deployment Scenario

Overlay PBX

Networked PBX

Direct to PSTN

Interop

SIP/PSTN Gateway

IP-PBX

SIP Trunking Service

Recommendationsno OCS voice yet – heading to pilot

Start thinking about scaleArchitecture: HA, DR, Security, ManagementValidation & Testing of headsets, devices, etc.

Build a solid infrastructure foundationAddress any outstanding issues with these elementsAnything with AD, DNS or Certificates will surface

Telephony integration for rapid successGateways maximize flexibilitySIP Trunking maximizes environmental simplicity

Recommendations successful Pilot , heading to production

Scale thinking pays off – now take the next stepLook at traffic flows on LAN/WANManaging usage as appropriateCosts of Least Cost Routing vs. PSTN / Carrier

Respect the UsersDeploy MonitoringEnd-user training resources

Telephony integration for scaleIP-PBXs: Direct SIP behind or alongsideTDM PBXs: Direct to carrier

Moving to OCS Voice"I'm done with my PBX!"

Deploy a gateway to the PSTN or SIP TrunkingGive everyone OCS VoiceRemove desk phones for those who don't need them - give others the choiceUsers on PBX can do personal call forwarding from Desk phone to OCS phoneTurn off the PBX

What's next?

Move your pilot forward to production

Know that upgrading your IP-PBX is not the first step on the road to Unified Communications

Experience the end-user capability anywhere! FREE Hosted Trial at https://r2.uctrial.com

Check out the administration experience!FREE OCS VMs at http://microsoft.com/VHD

Voice Resources

Programs & StandardsUnified Communications Open Interoperability ProgramMicrosoft Office Protocol Documentation

White PapersIntegrating Telephony with Office Communications Server 2007 and 2007 R2Microsoft Quality of Experience

DocumentationVoIP ArchitectureConfiguring Voice Quality of Service

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