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.

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Transcript 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!
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