Software-Powered VoIP

(1)

Unified. Simplified.

Software-Powered VoIP

Ali Rohani

Anthony Murphy Scott Stubberfield

(2)

Unified Communications Architecture

Core Scenarios

QOE Monitoring

Archiving CDR

Public IM Clouds

AOL MSN

Yahoo

Remote Users

DMZ

Federated Businesses

Front-End Server(s)

Inbound Routing Outbound

Routing

PSTN

Backend SQL server

Exchange 2007 Server UM

Voicemail

Active Directory Voice Mail

Routing

PBX

(SIP-PSTN GW)

Access Server

Data Audio/

Video SIP

Mediation Server

PRI

UC endpoints

Communicator Web Access

(3)

Voice Components

QOE Monitoring

Archiving CDR

Public IM Clouds

AOL MSN

Yahoo

Remote Users

DMZ

Front-End Server(s)

Routing

PSTN

Backend SQL server

Voicemail

Routing

PBX

(SIP-PSTN GW)

Access Server

Data Audio/

Video SIP

Mediation Server

PRI

UC endpoints

(4)

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

Mediation Server

(5)

Build Foundation

Coexistence

Standalone

Software-Powered Telephony Architecture

Deployment Scenarios

Build Foundation

IM & Presence Web Conferencing

Audio/Video Conferencing

Option 1: Coexistence (Dual Forking)

Add software-powered VoIP with OCS 2007 Keep legacy phone

Option 2: Standalone (Enterprise Voice)

Add software-powered VoIP with OCS 2007

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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

Make use of PBX legacy capabilities

Receptionist and boss-admin needs Emergency call requirements

Analog lines (e.g. fax machines)

Coexistence

Mediation Server

Native SIP

Existing UC Enabled

PBX

OCS 2007

IM, Presence,

Audio, Video, Conferencing, IVR

Inbound Routing Outbound Routing Voice Mail Routing

Option 1: Coexistence (Dual Forking)

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Standalone

Migrate teams or

departments to software- powered 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

Mediation Server

Native SIP

OCS 2007

IM, Presence,

Audio, Video, Conferencing, IVR SIP/PSTN

Gateway

Option 2: Standalone (Enterprise Voice)

Inbound Routing Outbound Routing Voice Mail Routing

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Basic Media Gateway

Standalone appliance Supports TDM features SIP over TCP

RFC 3261 compliant SIP G.711

Works with Mediation Server

SIP/PSTN Gateways

Hybrid Media Gateway

Media Gateway appliance

Collocated with Mediation Server

Enterprise Voice Client Front End or

Director Mediation

Server Basic Media

Gateway

Enterprise Voice Client Front End or

Director Hybrid Media

Gateway

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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

Unique Unified Experience

UC Peripherals

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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

Unique Unified Experience

Communicator Phone Edition

The Communicator phone experience simplifies everyday calling features

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Communicator Phone Edition

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Voice Routing Elements

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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

Location Profiles

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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

Normalization Rules

Location Profile

Normalization Rule 1 Normalization Rule n

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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

Phone Usage

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User Policies

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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

Routes

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Visual Representation User Initiates call

Number or SIP

URI E 164

Format Location profiles applied

Normalization Rule matched

Reverse number lookup

success

Error

INBOUND Routing Ring PC, Forward,

Dual Fork, Ex UM, etc.

OUTBOUND Routing

Phone Usage Match

Error

Route Match

Error

Select Route

Mediation Server and/or Media Gateway SIP URI

Number

Yes No

No

Yes

Yes - SIP URI

No

Yes

No

Yes

PSTN PBX

Apply user preferences

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Voice Components

QOE Monitoring

Archiving CDR

Public IM Clouds

AOL MSN

Yahoo

Remote Users

DMZ

Front-End Server(s)

Routing

PSTN

Backend SQL server

Voicemail

Routing

PBX

(SIP-PSTN GW)

Access Server

Data Audio/

Video SIP

Mediation Server

PRI

UC endpoints

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Voice Configuration

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Quality Of Experience (QoE)

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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

The Challenges of Packet Network

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Audio Codecs

And Network Service Quality

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“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

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Traditional approach of QoS/CAC is complex and

difficult to manage Admins may not control whole network

Users are increasingly mobile

Most common source of user dissatisfaction is Voice Quality

Many factors affect voice quality

Cost $

Complexity

Ubiquity

Traditional IP Telephony Limitations

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Comprehensive, user-focused approach to quality

Smart, adaptive end-points

Real time metrics of actual experience

Media stack optimized for unmanaged IP networks

Microsoft Quality of Experience

Complete and Comprehensive solution that doesn’t require QoS

Quality of Experience Monitoring Server

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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)

28 Kbps RTAudio

(16kHz)

45 Kbps G.726 48 Kbps G.711 80 Kbps

Microsoft Real-Time Codecs

RTAudio & RTVideo

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0 0.5 1 1.5 2 2.5 3 3.5 4 4.5

Office Communicator Traditional IP Phone

Wide-Band MOS

Rating

Perfect Network

Corporate Network

Internet High Congestion

Source: Psytechnics 12/06

Quality of Experience At Work

Noise free

(29)

0 0.5 1 1.5 2 2.5 3 3.5 4 4.5

Office Communicator Traditional IP Phone

Perfect Network

Corporate Network

Internet High Congestion

Source: Psytechnics 12/06

Wide-Band MOS

Rating

Quality of Experience At Work

Office Environment

(30)

These are one-way “on the wire” numbers Numbers are worst-case

Silence suppression saves more bandwidth

Packetization dynamically changes based on usage

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

What Does Media Take?

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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

Right Sizing your Network

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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

Other Network Considerations

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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

Managing the Usage

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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

QoS Support

(35)

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

OCS 2007 Installation Edge Servers

In DMZ Edge Servers

In DMZ

Federation with Audio & Video

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Thank You