Transcript Jamie Stark Senior Product Manager Microsoft UNC305 Agenda Before your deployment Enterprise Voice Elements Deployment Scenarios Recommendations & Next steps.

Jamie Stark
Senior Product Manager
Microsoft
UNC305
Agenda
Before your deployment
Enterprise Voice Elements
Deployment Scenarios
Recommendations & Next steps
Customer's OCS Voice state
No voice with Office Communications Server
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 level and higher
AD used to store global settings & groups
Single-forest & multiple-forest environments
Exchange Server 2010 or Exchange 2007 SP1
Unified Messaging, Missed Call Notification,
Auto Attendant, Outlook Voice Access
For OCS: Windows Server x64 only
Windows Server 2003 x64 SP2
Windows Server 2008 x64 (RTM and SP2)
SQL Server 2008 SP1 and SQL 2005 SP2 (SP3)
Choosing 2007 R2 Topology
Topology Choice
Users
High
Availability
Comments
2007 R2 Standard
Edition
Up to 5,000
No
Simplest deployment, lacks HA
2007 R2 Virtualized
Up to 40,000
Yes
Media workloads are not
supported virtualized
2007 R2 Enterprise
Edition Consolidated
Up to 100,000*
Yes
Recommended topology, simplified
deployment (from 2007), HA, Scale.
2007 R2 Enterprise
Edition Consolidated
(IM & P only)
Up to 200,000*
Yes
IM only deployment
2007 R2 Enterprise
Edition Expanded
Up to 125,000
Yes
Special custom scenarios:
Examples: Lots of web conf, no A/V
Recommended 2007 R2 Topology
Enterprise Consolidated topology
Primary supported topology
Consolidated Edge Server
Monitoring Server (collocating CDR and QoE)
Dedicated Archiving Server
Motivation for Consolidated topology
Simplify Office Communications Server deployments
Fewer, more powerful systems (64 bit)
Collocate server roles
Requirements for CDR, Archiving, and QoE
Maintain flexibility
Hardware Recommendations
For Enterprise Edition
Front-end server
x64 Dual Quad-Core CPU, or 4 way Dual-Core CPU
8 GB Memory
10K rpm disk drives
Back-end SQL server
x64 Dual Quad-Core CPU, or 4 way Dual-Core CPU
16 GB Memory
10K or 15K rpm disk drives (multiple spindles)
Other servers
Use Front-end or Back-end spec as appropriate
Planning Tool
Enterprise Voice Elements
Mediation Server: Intermediate signaling & 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 Providers
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
Altigen
Huawei
Innovaphone
Mitel
Nortel
Seltatel
Interoute
Jajah
Global Crossing
Sprint
ThinkTel
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
Call Routing & Handling
Location Profiles & Normalization Rules
Phone Usages
Voice Policies
Outbound call routes
Response Groups
Outbound UC Call – SIP URI
SIP URI
UC endpoint
Called Party
Number
SIP INVITE
Translations
Application
Outbound
Routing
Mediation Server
Gateway
UC endpoint/PSTN
[email protected]
sip:[email protected]
Outbound UC Call – E.164 number
E.164 number
UC endpoint
Called Party
Number
+12065551111
SIP INVITE
sip:[email protected]
Translations
Application
Outbound
Routing
RNL maps
E.164 number
to SIP URI
Reverse Number Lookup
Success
Fail
sip:[email protected]
MS
selected
Routing
sip:[email protected]
Mediation Server
Gateway
UC endpoint/PSTN
Mediation
Server
sip:[email protected]
sip:[email protected]
GW
Converted
to local
format
555-1111
US PSTN
Outbound UC Call – Non E.164
Non-E.164 number
UC endpoint
Called Party
Number
51111
sip:[email protected];
Phone-context=Redmond
SIP INVITE
Translations
Application
Location
Profile
Number Normalization
sip:[email protected]
Outbound
Routing
RNL maps
E.164 number
to SIP URI
Converted
to E.164
Reverse Number Lookup
Success
Fail
sip:[email protected]
MS
selected
Routing
sip:[email protected]
Mediation Server
Gateway
UC endpoint/PSTN
Mediation
Server
sip:[email protected]
sip:[email protected]
GW
Converted
to local
format
555-1111
US PSTN
Outbound UC Call – Summary
SIP URI
E.164 number
Non-E.164 number
+12065551111
51111
UC endpoint
Called Party
Number
SIP INVITE
[email protected]
sip:[email protected]
sip:[email protected];
Phone-context=Redmond
sip:[email protected]
Translations
Application
Location
Profile
Number Normalization
sip:[email protected]
Outbound
Routing
RNL maps
E.164 number
to SIP URI
Converted
to E.164
Reverse Number Lookup
Success
Fail
sip:[email protected]
MS
selected
Routing
sip:[email protected]
Mediation Server
Gateway
UC endpoint/PSTN
Mediation
Server
sip:[email protected]
sip:[email protected]
GW
Converted
to local
format
555-1111
US PSTN
Inbound PSTN Call – E.164 number
E.164 number
US PSTN
Gateway
Called Party Number
+142555511111
Mediation Server
SIP INVITE
Translations
Application
sip:[email protected]
RNL maps
E.164 number
to SIP URI
Reverse Number Lookup
Outbound
Routing
Success
sip:[email protected]
UC endpoint
Fail
Inbound PSTN Call – non E.164
Non-E.164 number
US PSTN
Gateway
11111
Called Party Number
Mediation Server
sip:[email protected];
Phone-context=Redmond
SIP INVITE
Number Normalization
Translations
Application
RNL maps
E.164 number
to SIP URI
sip:[email protected]
Reverse Number Lookup
Outbound
Routing
Success
sip:[email protected]
UC endpoint
Fail
Converted
to E.164
Location
Profile
Inbound PSTN Call - Summary
E.164 number
US PSTN
US PSTN
+142555511111
11111
Gateway
Called Party Number
Non-E.164 number
Mediation Server
SIP INVITE
sip:[email protected];
Phone-context=Redmond
sip:[email protected]
Number Normalization
Translations
Application
RNL maps
E.164 number
to SIP URI
sip:[email protected]
Reverse Number Lookup
Outbound
Routing
Success
sip:[email protected]
UC endpoint
Fail
Converted
to E.164
Location
Profile
Call Routing UI
Voice Deployment Goals
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
Voice Deployment Scenarios
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
Voice Deployment Scenarios
Direct to PSTN: Owned dial plan
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
Voice Deployment Scenarios
Networked PBX: Split dial plan
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
For Example: use Gateways for Pilot, Direct SIP to HQ
IP-PBX, keep some users on PBX but move others
Pilot
Deploy
Scale
Considering User Population
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
Company
Size
250
Current Stage
No Voice
Deployed
1000
5000
10,000
50,000
Deployment
Scenario
User
Population
Deployment
Goal
Interop
Networked
PBX
Mobile &
Remote
Employee
With Existing
IP-PBX
SIP/PSTN
Gateway
Mixed
environment
IP-PBX
OCS for Voice
SIP Trunking
Service
Pilot
Successful
Initial
Deployment
Successful
HQ /
Single Site
Direct to
PSTN
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
Recommendations
Moving to OCS Voice
Deploy SIP Trunking or Gateway to PSTN
Roll out OCS Voice to everyone in the company
For end-users who don’t need a PBX phone
Remove it
Port the number if possible or acquire new DIDs
For those with unmet requirements
Keep the PBX phone – migrate them as features become
available.
Personal call forwarding to OC to get the experience
Give everyone else the choice with an opt-in approach
Look at carrier process for new DIDs vs. moving numbers
Slowly phase out the IP-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!
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Related Content at Teched
UNC04-IS - Microsoft Office Communications Server Video Strategy
11/10/2009
13:30 - 14:45
Interactive Theatre 5 - Yellow
UNC303 - Voice Administration and Monitoring
11/10/2009
15:15 - 16:30
Berlin 2 - Hall 7-3a
UNC204 - Ten Ways to Become a Hero with Microsoft Office Communications Server
11/11/2009
09:00 - 10:15
Berlin 2 - Hall 7-3a
UNC305 - Voice Architecture and Planning
11/11/2009
10:45 - 12:00
Berlin 2 - Hall 7-3a
UNC313 - Audio Conferencing Deep Dive
11/12/2009
09:00 - 10:15
Berlin 2 - Hall 7-3a
UNC07-IS - Troubleshooting Microsoft Office Communications Server 2007 R2
11/12/2009
17:00 - 18:15
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
A Practical Approach to Deploying Real-time
Communications
Microsoft Quality of Experience
Documentation
VoIP Architecture
Configuring Voice Quality of Service
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