Jamie Stark Senior Product Manager Microsoft Corporation UNC311 Scope Voice with Office Communications Server Enterprise Voice Planning Deployment Scenarios Sizing and Topology Considerations Call Routing and Management Interoperability With.
Download ReportTranscript Jamie Stark Senior Product Manager Microsoft Corporation UNC311 Scope Voice with Office Communications Server Enterprise Voice Planning Deployment Scenarios Sizing and Topology Considerations Call Routing and Management Interoperability With.
Jamie Stark Senior Product Manager Microsoft Corporation UNC311 Scope Voice with Office Communications Server Enterprise Voice Planning Deployment Scenarios Sizing and Topology Considerations Call Routing and Management Interoperability With Existing Telephony Infrastructure Out of Scope Integrations that don’t involve Voice Client-based plug-ins PBX-based controls hosted inside Communicator Features, Experience, Capability vary widely Remote Call Control Office Communicator controls a PBX station set Consider 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…yet How do I get started with a pilot? Validate the technology & business case Pilot completed and successful How do I move the pilot to production? Breadth - more & different users & cases Initial deployment completed and successful How do I grow production to scale? Business critical, multi-site communications Prerequisites to Deployment Windows Server 2003 Domain functional level AD used to store global settings & groups Single-forest & multiple-forest environments Exchange Server 2007 with Service Pack 1 Unified Messaging, Missed Call Notification, Auto Attendant, Outlook Voice Access Assess your IP Network for real-time traffic Bandwidth Latency & other network affects Your IP Network Bandwidth Provision for streams of 45Kbps audio, 300Kbps video Modeling calling patterns: Intra/inter-office, external Other network affects Reduce delay: < 150ms E2E delay considered excellent (G.114) Some jitter (< 30ms) & loss (< 10%) can be handled Prioritizing Media with DiffServ Audio: Expedited Forwarding; Video: Assured Forwarding Centralized policy enforcement for Vista PCs Controlling Usage Manage the size of conferences, allowed media Set port range for media, per session BW limits, video quality Your (future) Voice Network Enterprise Voice Elements Mediation Server: Intermediate signaling and call flow Manage innovative elements of the SIP transaction: Inside, TLS/SRTP – Outside, TCP/RTP Transcode media flows from G.711 to RTAudio and SIREN Act as an ICE Client for PSTN-originated calls Provide quality metrics back to monitoring server Upstream telephony elements SIP/PSTN Gateway IP-PBX SIP Trunking Service UC Open Interoperability Program (UCOIP) Qualification program for telephony infrastructure – SIP/PSTN Gateways, IP-PBXs & SIP Trunking Service Goal for seamless interoperability with Office Communications Server and Exchange Server Ensure Customers have positive experiences with Setup, Support, and Use of qualified devices Allows for scalable qualification of vendors SIP/PSTN Gateways IP-PBXs SIP Trunking Service Aculab, Audiocodes, Cisco, Dialogic, Ferrari, NEC, NET, Nortel, Nuera, Quintum, Tango Networks, Vegastream Innovaphone Mitel Nortel Seltatel Interoute Global Crossing Sprint http://technet.microsoft.com/UCOIP Tested IP-PBXs PBX vendors qualify their latest versions Customers want support for existing versions Where possible, Microsoft may test IP-PBXs To date: Cisco Unified Communications Manager OCS 2007 OCS 2007 R2 CUCM 4.X 4.2(3)_SR3a 4.2(3)_SR3a 4.2(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) Listed with qualified infrastructure on UCOIP Voice Deployment Continuum Voice Capabilities with an existing IP-PBX Long-term interoperability for all end-users Combined experience of Communicator & PBX Phone Mixed environment with PBX Some users will be on the PBX, others will move Mix expected to change over time Office Communications Server for Voice Enable a temporary transition state Prepare for your PBX-less future Deployment Scenarios Overlay PBX: Shared Dial Plan Dual Forking: PBX rings to phone & Communicator Multiple Dial Plans: multiple numbers per user Networked PBX: Split Dial Plan Mediation Server located behind the PBX Connect using SIP/PSTN Gateway or Direct SIP Direct to PSTN: Owned Dial Plan Calls are sent/received directly with PSTN SIP Trunking from carrier or circuits to Gateway Deployment Scenarios Overlay PBX dual forking Calls from PSTN & PBX phones are forked by IP-PBX to Office Communications Server Any calls made by Office Communicator are also forked to IP-PBX Provides a blended, single number experience for end-users Each infrastructure element implements equivalent dial plan PBX upgrade required for Direct SIP plus Dual-Forking qualification Overlay PBX multiple dial plans Each user has two numbers – one for Office Communicator & one for PBX No forking Users configure personal call forwarding between systems Configure with SIP/PSTN Gateway or Direct SIP to IP-PBX If using Gateway, trunk expense rises as size of deployment grows Fundamentally a transition over time, assume PBX phone goes away Deployment Scenarios Networked PBX using a SIP/PSTN Gateway Users are moved off the PBX Calls delivered from PSTN to PBX and routed to SIP/PSTN Gateway Wide availability of Gateways for geography & circuit configurations Dialing behavior preserved for calls between all users Fast & inexpensive to deploy for pilot & smaller production Double-trunking through the PBX increases cost with scale Networked PBX Direct SIP Users are moved off the PBX Calls delivered from PSTN to IP-PBX Mediation Server connects directly to SIP interface on supported IP-PBX But 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-PBX As production deployments grow, Direct SIP has OA&M advantages Deployment Scenarios Direct to PSTN IP-IP Gateways Mediation Server sits behind same SIP/PSTN Gateway used by IP-PBX Supported configuration as long as the Gateway is qualified with OCS Gateway routes based on DID or trunk group, may require configuration on the carrier Some Gateways support doing an AD Query for routing determination Increased flexibility and negligible impact to trunking costs when moving users from PBX Direct to PSTN using a SIP/PSTN Gateway Separate PSTN interconnect infrastructure from PBX Number routing change or new numbers provisioned by Carrier Requires zero PBX changes, eventually move trunks from PBX to SIP/PSTN Gateway Internal calls between user groups routed through PSTN Direct to PSTN SIP Trunking Connecting Mediation Server to SIP Trunking Service No on-premise third party products (SBCs, etc.) required Uses nailed up VPN to Service Provider for security Brings telephony trunking into datacenter consolidation strategy Still early days…not all carriers signed on to support modern (SIP Connect 1.1) standards More Options All of these can be deployed in a mixed fashion Scenarios can change as deployment matures Trunking both to IP-PBX and PSTN clouds Use 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 & Remote Great pilot users to validate capability Discontinuous number range Headquarters / Single Site Expect a mix of Communicator & PBX for coverage of all enterprise telephony features Most IW users can use Enterprise Voice exclusively Branch Office Gateways for Least Cost Routing & Local number termination WAN Survivability via Cell Phone & Internet Deployment Factors to Consider Company Size Current Stage Deployment Scenario User Population Deployment Goal Interop 250 No Voice Deployed Overlay PBX Mobile & Remote Employee With Existing IP-PBX SIP/PSTN Gateway Pilot Successful Networked PBX Mixed environment IP-PBX Initial Deployment Successful Direct to PSTN OCS for Voice SIP Trunking Service 1000 5000 10,000 50,000 Headquarters Single Site Branch Office Recommendations no OCS voice yet – heading to pilot Start thinking about scale Architecture: HA, DR, Security, Management Validation & Testing of headsets, devices, etc. Build a solid infrastructure foundation Address any outstanding issues with these elements Anything with AD, DNS or Certificates will surface Telephony integration for rapid success Gateways maximize flexibility SIP Trunking maximizes environmental simplicity Recommendations successful Pilot , heading to production Scale thinking pays off – now take the next step Look at traffic flows on LAN/WAN Managing usage as appropriate Costs of Least Cost Routing vs. PSTN / Carrier Respect the Users Deploy Monitoring End-user training resources Telephony integration for scale IP-PBXs: Direct SIP behind or alongside TDM PBXs: Direct to carrier Moving to OCS Voice "I'm done with my PBX!" Deploy a gateway to the PSTN or SIP Trunking Give everyone OCS Voice Remove desk phones for those who don't need them - give others the choice Users on PBX can do personal call forwarding from Desk phone to OCS phone Turn 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 & Standards Unified Communications Open Interoperability Program Microsoft Office Protocol Documentation White Papers Integrating Telephony with Office Communications Server 2007 and 2007 R2 Microsoft Quality of Experience Documentation VoIP Architecture Configuring Voice Quality of Service Voice Breakout Sessions this week Tues 14:45 – 16:00: UNC309 - OCS 2007 R2 Dial-in Audio Conferencing – ROOM 403B Wed 10:15 – 11:30: UNC302 – Managing Response Group Service of OCS – ROOM 403A Wed 14:45 – 16:00: UNC306 – Archiving, CDR, and QoE Monitoring in OCS – ROOM 403B Thurs 08:30 – 9:45: UNC303 – Deep Dive into the Edge Server in OCS 2007 R2 – ROOM 404 Thurs 14:45-16:00: UNC323 – Troubleshooting OCS 2007 R2 – ROOM 403A Call to Action Learn More! Related Content at TechEd on “Related Content” Slide Attend in-person or consume post-event at TechEd Online Check out online learning/training resources http://technet.microsoft.com/exchange/2010 http://technet.microsoft.com/office/ocs Try It Out! Download the Exchange Server 2010 Beta Evaluation http://www.microsoft.com/exchange/2010/try-it Get a 5-Day Trial of Office Communications Server 2007 R2 https://r2.uctrial.com/ Resources www.microsoft.com/teched www.microsoft.com/learning Sessions On-Demand & Community Microsoft Certification & Training Resources http://microsoft.com/technet http://microsoft.com/msdn Resources for IT Professionals Resources for Developers www.microsoft.com/learning Microsoft Certification and Training Resources Complete an evaluation on CommNet and enter to win! © 2009 Microsoft Corporation. All rights reserved. Microsoft, Windows, Windows Vista and other product names are or may be registered trademarks and/or trademarks in the U.S. and/or other countries. The information herein is for informational purposes only and represents the current view of Microsoft Corporation as of the date of this presentation. Because Microsoft must respond to changing market conditions, it should not be interpreted to be a commitment on the part of Microsoft, and Microsoft cannot guarantee the accuracy of any information provided after the date of this presentation. MICROSOFT MAKES NO WARRANTIES, EXPRESS, IMPLIED OR STATUTORY, AS TO THE INFORMATION IN THIS PRESENTATION.