Software-Powered VoIP

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1 Software-Powered VoIP Ali Rohani Anthony Murphy Scott Stubberfield

2 Unified Communications Architecture Core Scenarios UC endpoints QOE Monitoring Archiving CDR AOL Public IM Clouds Yahoo Remote Users MSN DMZ Data Audio/ Video SIP Access Server Front-End Server(s) Inbound Routing Outbound Routing Voice Mail Routing Backend SQL server Communicator Web Access Federated Businesses Mediation Server Exchange 2007 Server UM Active Directory (SIP-PSTN GW) Voic PSTN PRI PBX

3 Unified Communications Architecture Voice Components UC endpoints QOE Monitoring Archiving CDR AOL Public IM Clouds Yahoo Remote Users MSN DMZ Data Audio/ Video SIP Access Server Front-End Server(s) Inbound Routing Outbound Routing Voice Mail Routing Backend SQL server Communicator Web Access Federated Businesses Mediation Server Exchange 2007 Server UM Active Directory (SIP-PSTN GW) Voic PSTN PRI PBX

4 Unified Communications Architecture Mediation Server Connects OCS and SIP/PSTN Gateway or IP-PBX Front-end of the Microsoft OCS voice world Intermediate signaling and call flow Manage innovative elements of the SIP transaction Transcode RTP flows from G.711 to RTAudio and SIREN Act as an ICE Client for PSTN-originated calls Enables OCS to Provide IP telephony Interconnect with the legacy PSTN

5 Software-Powered Telephony Architecture Deployment Scenarios Build Foundation IM & Presence Web Conferencing Audio/Video Conferencing Build Foundation Option 1: Coexistence (Dual Forking) Add software-powered VoIP with OCS 2007 Keep legacy phone Coexistence Option 2: Standalone (Enterprise Voice) Add software-powered VoIP with OCS 2007 Standalone

6 Software-Powered Telephony Architecture Option 1: Coexistence (Dual Forking) Simultaneous ringing on Office Communicator and legacy phone Provide interoperability with PBX systems Allow a single user to have both, Office Communicator and legacy phone Existing UC Enabled PBX Mediation Server Coexistence OCS 2007 Inbound Routing Outbound Routing Make use of PBX legacy capabilities Receptionist and boss-admin needs Emergency call requirements Analog lines (e.g. fax machines) Native SIP Voice Mail Routing IM, Presence, Audio, Video, Conferencing, IVR

7 Software-Powered Telephony Architecture Option 2: Standalone (Enterprise Voice) Migrate teams or departments to softwarepowered VoIP Typically information or mobile workers Non-VoIP users have IM and presence Supports standard PBX migration procedures Common PBX network interfaces PBX numbering plans No PBX upgrade required Existing PBX SIP/PSTN Gateway Mediation Server Native SIP Standalone OCS 2007 Inbound Routing Outbound Routing Voice Mail Routing IM, Presence, Audio, Video, Conferencing, IVR

8 SIP/PSTN Gateways Basic Media Gateway Standalone appliance Supports TDM features SIP over TCP RFC 3261 compliant SIP Basic Media Gateway Mediation Server Front End or Director Enterprise Voice Client G.711 Works with Mediation Server Hybrid Media Gateway Media Gateway appliance Collocated with Mediation Server Hybrid Media Gateway Front End or Director Enterprise Voice Client

9 Unique Unified Experience UC Peripherals Extending Communicator experience Personal computer, Web and mobile devices Dedicated communications devices Software for innovative IP Devices Implementations design with embedded software Partners deliver production devices

10 Unique Unified Experience Communicator Phone Edition The Communicator phone experience simplifies everyday calling features Scrolling through the status of people on users buddy lists Initiating a call simply by touching a name Setting up impromptu conference calls Merging two calls into one Forwarding calls

11 Communicator Phone Edition

12 Voice Routing Elements

13 Voice Routing Elements Location Profiles Define how phone numbers are translated when dialed from a given location Determines outbound routing Translates a phone number based on the user s location (dial plan) Sent through in-band provisioning of the client Global in scope

14 Voice Routing Elements Normalization Rules Normalization rules convert dialed numbers into a standard format Each Location Profile contains one or more Normalization Rules A normalization rule consists of two parts Match Regular Expression Translation Syntax If a dialed number matches the regular expression, it is converted to standard form using the translation syntax Location Profile Normalization Rule 1 Normalization Rule n

15 Voice Routing Elements Phone Usage Phone Usage Records assign calling permissions to users and routes Phone Usage Records can be given any name, but typically indicate a calling destination; for example, Local Phone Usage Records are global in scope One or more Phone Usage Records are associated with a User Policy Phone Usage Records can also identify a route

16 Voice Routing Elements User Policies

17 Voice Routing Elements Routes Defines how Communications Server handles calls placed by users Phone usage associated with users are used to look up a route in the routing table When a matching route is found, the target phone number is checked against the dialed phone number; if matched, the call can be routed to the Gateway

18 Voice Routing Elements Visual Representation User Initiates call Location profiles applied No E 164 Format Number Number or SIP URI Yes SIP URI Normalization Rule matched No Error Yes Reverse number lookup success Yes - SIP URI INBOUND Routing Apply user preferences Ring PC, Forward, Dual Fork, Ex UM, etc. No Mediation Server and/or Media Gateway OUTBOUND Routing Phone Usage Match Yes Route Match Yes Select Route PSTN No No PBX Error Error

19 Unified Communications Architecture Voice Components UC endpoints QOE Monitoring Archiving CDR AOL Public IM Clouds Yahoo Remote Users MSN DMZ Data Audio/ Video SIP Access Server Front-End Server(s) Inbound Routing Outbound Routing Voice Mail Routing Backend SQL server Communicator Web Access Federated Businesses Mediation Server Exchange 2007 Server UM Active Directory (SIP-PSTN GW) Voic PSTN PRI PBX

20 Voice Configuration

21 Quality Of Experience (QoE)

22 The Challenges of Packet Network Traditional IP telephony not designed for IP networks Transfer of circuit switched concepts Fragile codecs, sensitive to minute network impairments Even a 1% loss can significantly degrade the user experience with G.711, which is considered the standard for toll quality 1 The default G.729 codec requires packet loss far less than 1% to avoid audible errors 2 Network engineering required for traditional IP telephony QoS and CAC work to recreate conditions of switched networks 1 - Intel: Overcoming Barriers to High-Quality Voice over IP Deployments 2 - Cisco: Quality of service for Voice over IP

23 Audio Codecs And Network Service Quality

24 The ripe taste of cheese improves with age. Act on these orders with great speed. Codec Perfect Network With Network Loss G.711 G.729 RT Audio Narrowband

25 Traditional IP Telephony Limitations Cost $ Traditional approach of QoS/CAC is complex and difficult to manage Ubiquity Complexity Admins may not control whole network Users are increasingly mobile Most common source of user dissatisfaction is Voice Quality Many factors affect voice quality

26 Microsoft Quality of Experience Complete and Comprehensive solution that doesn t require QoS Comprehensive, user-focused approach to quality Smart, adaptive end-points Real time metrics of actual experience Media stack optimized for unmanaged IP networks Quality of Experience Monitoring Server

27 Microsoft Real-Time Codecs RTAudio & RTVideo Supports Wideband and Narrowband modes Wideband greatly improves intelligibility and naturalness of speech Constant and Variable bit rate modes Dynamically responds to changes in audio complexity Highly efficient use of Bandwidth More quality at equivalent bandwidth Same quality at lower bandwidth Multi-rate codec Enables real-time adaptation RTAudio (8kHz) RTAudio (16kHz) 28 Kbps 45 Kbps G Kbps G Kbps

28 Quality of Experience At Work Noise free Wide-Band MOS Rating Office Communicator Traditional IP Phone 0 Perfect Network Corporate Network Internet High Congestion Source: Psytechnics 12/06

29 Quality of Experience At Work Office Environment Wide-Band MOS Rating Office Communicator Traditional IP Phone 0 Perfect Network Corporate Network Internet High Congestion Source: Psytechnics 12/06

30 What Does Media Take? Codec Min Bandwidth Max Bandwidth Real-time Audio (RTA) 24 Kbps 45 Kbps Siren 48 Kbps 48 Kbps Real-time Video (VC-1) 50 Kbps 250 Kbps Roundtable Panorama 50 Kbps 350 Kbps These are one-way on the wire numbers Numbers are worst-case Silence suppression saves more bandwidth Packetization dynamically changes based on usage

31 Right Sizing your Network You are adding a service to your network Even with 2:1 adaptability clients need bandwidth Holds true for VoIP telephony or conferencing Simple policies give you control

32 Other Network Considerations Delay Engineer to less than a mean of 150 milliseconds Loss Up to 10% can be handled without significant problems Connectivity The clients can connect through most common networks Call Admission Control Hard limits aren't required to provide quality

33 Managing the Usage Server policy Type of conference that can be setup Who can setup a conference How many users per conference Client policy Bandwidth used per application Specify the ports, limiting the range used

34 QoS Support OCS 2007 does not require DiffServ, RSVP or CAC We do work within a DiffServ environment DSCP marking by the end-points By default, end-points mark all media Audio: Expedited Forwarding Video: Class 3 of Assured Forwarding DSCP marking can also be tuned through policy

35 Federation with Audio & Video Communication with users outside your organization Signaling and data encrypted with certificates Leverages same infrastructure as for external users Offers significant toll reduction OCS 2007 Installation Edge Servers In DMZ Edge Servers In DMZ OCS 2007 Installation

36 Thank You