1 © 2007 Avaya Inc. All rights reserved. Understanding SIPs Role in Intelligent Communications Tom...

1© 2007 Avaya Inc. All rights reserved.

Understanding SIP’s RoleUnderstanding SIP’s Roleinin

Intelligent CommunicationsIntelligent Communications

Understanding SIP’s RoleUnderstanding SIP’s Roleinin

Intelligent CommunicationsIntelligent Communications

Tom DoriaDirector – Avaya P2P Technical Business DevelopmentChair – Avaya SIP Virtual Team


AgendaAgendaIntelligent Communications & SIPIntelligent Communications & SIP

– The Changing World & Evolving Business NeedsThe Changing World & Evolving Business Needs– Understanding Avaya’s Vision for Intelligent CommunicationsUnderstanding Avaya’s Vision for Intelligent Communications– The Building Blocks to Intelligent CommunicationThe Building Blocks to Intelligent Communication

Defining SIPDefining SIP– IETF’s Vision for SIPIETF’s Vision for SIP– Key Concepts (SIP, SIPing, SIMPLE)Key Concepts (SIP, SIPing, SIMPLE)

Why SIPWhy SIP– Multi-Vendor InteroperabilityMulti-Vendor Interoperability– Presence and PreferencePresence and Preference– Native Support for MultimediaNative Support for Multimedia– Single User IdentitySingle User Identity– Why SIP TrunkingWhy SIP Trunking

Building a SIP Enabled EnterpriseBuilding a SIP Enabled Enterprise– SIP Communication Components DefinedSIP Communication Components Defined– Mapping Industry Names to Avaya SolutionsMapping Industry Names to Avaya Solutions– Peer-to-Peer (P2P) SIPPeer-to-Peer (P2P) SIP

Vision Executed - SIP Service ExamplesVision Executed - SIP Service Examples– Example #1 - Avaya Quick Edition with SIP TrunkingExample #1 - Avaya Quick Edition with SIP Trunking– Example #2 – Avaya Integration with Microsoft Office CommunicatorExample #2 – Avaya Integration with Microsoft Office Communicator


The World Is Changing The World Is Changing

fromSeparate services: Local, long distance, mobile, video, Internet to

Bundled services: All distance voice, voice/data/video packages

fromSeparate platforms: Phone for voice, computer for Internet to

Multi-use devices: Blackberries; mobile phones with messaging, e-mail, video; computer for Internet phone and IM

fromSeparate providers: Cable companies for video, phone companies for voice

toMulti-product providers: Cable, mobile, and wireline companies offering voice/video/data


State of Convergence Today! End user view

We Need Converged Communication Applications that

Improve Productivity and Allow Service

Integration


Seamlessly and openly integrate communication applications and business applications

Intelligently connects Employees, Customers and Processes to the right people at the right time through the right medium

Deliver business agility with speed, responsiveness and control, increasing global competitiveness

Avaya’s Vision for a Avaya’s Vision for a New Era of Intelligent CommunicationsNew Era of Intelligent Communications

Intelligent Communication Solutions should:Intelligent Communication Solutions should:


What does Intelligent Communications Look Like?

Email

Instant Messaging

Voice Messaging

Unified Communication

Telephony

Contact Centers

Mobility & Softphone

Conferencing

Collaboration


Session Initiation Protocol Session Initiation Protocol (SIP)(SIP)

Next Generation Carrier ServicesNext Generation Carrier Services

Service Oriented Architecture (SOA)Service Oriented Architecture (SOA)

Avaya Intelligent CommunicationsAvaya Intelligent CommunicationsThe Building Blocks to Next Generation Intelligent Communication SolutionsThe Building Blocks to Next Generation Intelligent Communication Solutions


Intelligent CommunicationsIntelligent CommunicationsRedefining the way People, Processes, and Information ConnectRedefining the way People, Processes, and Information Connect

SIPSIP

SOASOA

INTELLIGENCE Between Applications & UserINTELLIGENCE Between Applications & User

Middleware Connects Silos of Application & Communication ServicesMiddleware Connects Silos of Application & Communication Services

Event Driven Services Initiate Communication with Users or Other ApplicationsEvent Driven Services Initiate Communication with Users or Other Applications

Application Protocol FabricApplication Protocol Fabric

Intelligent Routing of Communication & ServicesIntelligent Routing of Communication & Services

Supports Open Integration of Devices and Application ServicesSupports Open Integration of Devices and Application Services

Native Support for Multimedia & Multimodal CommunicationsNative Support for Multimedia & Multimodal Communications

Carrier

Carrier

Services

Services

NEXT GEN Carrier ServicesNEXT GEN Carrier Services

Mobile and Fixed Services IntegratedMobile and Fixed Services Integrated

Supports Rich PresenceSupports Rich Presence

SIP Enriched Multimedia Hosted AppsSIP Enriched Multimedia Hosted Apps.

• IP & SIP TrunkIP & SIP Trunk• IMSIMS• 3G & 4G Wireless3G & 4G Wireless









Defining Key ConceptsSIP, SIPPING, SIMPLE

and other words that begin with “S”


SSession ession IInitiation nitiation PProtocolrotocol

– SIP was designed to embrace the IETF concepts of KISS

– IETF standard for communications convergence (RFC 3261)

– Media agnostic – voice, video, text, etc.

– Enables applications to be integrated into communication sessions

Communication sessions based on “presencepresence”

– I.e. the publication of your willingness and ability to be communicated with

– Note that Presence publication is selective by user “preferencepreference”

IETF Site: http://www.ietf.org/html.charters/sip-charter.html

Shorthand Definition

What is SIP?What is SIP?

http://www.ietf.org/html.charters/sip-charter.html


Session Initiation Protocol Project INvestiGation– IETF working group

• Chartered to document the use of SIP for several applications related to telephony and multimedia

SIPPING-19 refers to SIP Services Examples draft– draft-ietf-sipping-service-examples-07– 19 example telephony features implemented in SIP– Purpose is to ensure that basic features interoperate

Other SIPPING items– SIP Basic Call Flow Examples (RFC 3665)– Message Waiting Indication (RFC 3842)

IETF Site: http://www.ietf.org/html.charters/sipping-charter.html

What is SIPPING-16, SIPPING-19, etc?What is SIPPING-16, SIPPING-19, etc?

http://www.ietf.org/html.charters/sipping-charter.html


SIP Services ExamplesSIP Services Examplesa.k.a. SIPPING-19a.k.a. SIPPING-19

Call Hold Consultation Hold Music on Hold Transfer – Unattended Transfer – Attended Transfer – Instant Messaging Call Forwarding –

Unconditional Call Forwarding – Busy Call Forwarding – No Answer 3-way Conference – 3rd Party

Added

3-way Conference – 3rd Party Joins Single Line Extension Find-Me Incoming Call Screening Outgoing Call Screening Call Park Call Pickup Automatic Redial Click to Dial Message Waiting Indication


What is SIMPLE?What is SIMPLE?

SSIP for IP for IInstant nstant MMessaging and essaging and PPresence resence LLeveraging everaging EExtensionsxtensions– IETF working group

Introduces “Presence” into communications stateIntroduces “Presence” into communications state– Builds on RFC 3265– Now a standard: RFC 3856

TerminologyTerminology– Presentity - The entity whose presence information is tracked (also called “buddy”) – Presence Agent – a program that servers presence subscription for a resource

(also called “Notifier”)

– Watcher - An endpoint (UA) that subscribes to presence changes (also called “subscriber”)

IETF Site:IETF Site: http://www.ietf.org/html.charters/simple-charter.html

http://www.ietf.org/html.charters/sipping-charter.html


Shorthand Definition

What is a SIP Trunk?What is a SIP Trunk?

A SIP Trunk is a single conduit pipeline for multimedia elements (voice, video and data)

A SIP Trunk is primarily a concurrent call that is routed over the IP backbone of a carrier using VoIP technology

SIP Trunks are commonly used in conjunction with an IP-PBX and are thought of as replacements for traditional circuits such PRI, T1, or analog circuits









Why SIP?Why SIP?


SIP Is a Key Enabler of Intelligent CommunicationsSIP Is a Key Enabler of Intelligent Communications

Presence and PreferencePresence and Preference

Native Support for MultimediaNative Support for Multimedia

Single User IdentitySingle User Identity

Multi-Vendor InteroperabilityMulti-Vendor Interoperability

The Right Person, in the Right Place, at the Right Time, the Right Way


SIP Promotes Interoperability– IETF SIP,SIPPING, and SIMPLE specifications provide the

foundation for interoperability– Compliant Devices can easily operate in the same enterprise– Options expand to a wide range of “best-of-breed” devices– SIP/Telephony Feature servers extend capabilities

(example Avaya Communication Manager)

Example: RIM Blackberry 7270– All-in-one mobile device combining SIP phone, intranet, email,

corporate data & application access– Works over any 802.11b/g WLAN

Example: Nokia Dual-Mode Phones– Support for cellular and 802.11 wireless VoIP– Embedded SIP client– Graphical user controls of desktop on cellular phone

RIM Blackberry 7270

Nokia Dual-Mode

Multi-Vendor InteroperabilityMulti-Vendor InteroperabilityMulti-Vendor InteroperabilityMulti-Vendor Interoperability

20© 2007 Avaya Inc. All rights reserved. 20© 2006 Avaya Inc. All rights reserved.Avaya – Proprietary & Confidential. For Limited Internal Distribution. The information contained in this document may not be distributed or reproduced, in whole or in part.

SIP Supports Intelligent Communication ChoicesSIP Supports Intelligent Communication Choices– Use Presence to determine availability (Avoid mail jail )

– Use Presence to determine correct Modality (Voice, IM, Video)

– Use Presence enabled applications to trigger event based communications (SOA)

SIP lets Users Control their WorldSIP lets Users Control their World– Selectively advertise your Presence (buddy lists)

– Monitor key individuals availability

– Select your preferred mode of communication

Presence and PreferencePresence and Preference


Intelligent Communications & Customer Service Scenario

4. Expert available online but busy on phone

5. Rep IM’s the expert - begins to get answers

IMIM

IMIM

1. Customer calls Customer Service

6. Rep notices expert has ended conversationAnd has expert join conference with customer

7. Expert answers questions and continues IM chats with the Rep in the background

IMIM

IMIM

2. Rep needs technical answer from expert

3. Rep uses presence to “peek over the cubicle” to see if expert available

IP Agent with SIP/SIMPLE IM ServicesIP Agent with SIP/SIMPLE IM Services


SIP Natively Supports Multimedia Sessions– Single network for voice & video– Unified global dial plan for voice & video

Phone features extended to video – Hold, mute, transfer, forward…

Voice and video meetings – Presence-based soft phone – Or MSFT client that is as easy to use as a phone

SIP Gateways support multi-point video integration – ISDN, H.320, H.323, and SIP video conferences

SIP Promotes Interoperability between multimediaSolutions from Different Vendors

Native Support for MultimediaNative Support for MultimediaNative Support for MultimediaNative Support for Multimedia


Connect people with people, not with devicesSingle user identity: SIP Address Of Record (AOR) is mapped across multiple devices (e.g. sip:[email protected] or sip:[email protected])

SIP Presence and Preference controls can be used to automatically route communications to the preferred device that is being “used”

SIP WiFi Phone

PDA

Softphone

Traditional phone

SIP Phone

Instant Messaging

Single User IdentitySingle User IdentitySingle User IdentitySingle User Identity


Limitation & Challenges of Private IP (H.323) Trunking

IP WANIP WAN

PSTNLocal &

Long Distance

PSTNLocal &

Long Distance

IP Data &Telephony

• Private IP (H.323) trunks are limited to VoIP communications between internal systems/sites• Separate TDM interfaces are required for external communication (partners/suppliers/customers)• Extra cost, extra hardware, extra complexity

Customers/ Partners/ Suppliers

LAN LAN

TDM in

terfa

ces TDM

interfaces

Why SIP Trunking?

IP Phone IP Phone IP Phone IP Phone

Exter

nal C

omm

unicat

ions

External Comm

unications

Internal Communications


SIP TrunkingA Single Pipe to the Cloud

Service Provider’s

WAN

Service Provider’s

WAN

PSTNLocal &

Long Distance

PSTNLocal &

Long Distance

H.323

SIPSIP

Customers/ Partners/ Suppliers

LAN LAN

IP Phone IP Phone IP Phone IP Phone

• Single IP link for voice/Multimedia/Data• Optimize use of WAN access by consolidating voice and data services• Eliminate PSTN interfaces for long-distance and local access (carrier provides the gateways)• Assign local telephone numbers to any ‘virtual location,’ independent of physical location• Save on toll charges• Prepares for future SIP solutions









SIP Infrastructure ComponentsSIP Infrastructure Components

• SIP User AgentSIP User Agent - - Any network endpoint that can originate or terminate a SIP session; this might include a SIP-enabled telephone, a SIP PC client (known as a "softphone"), or a SIP-enabled gateway

The building blocks of an Intelligent SIP NetworkThe building blocks of an Intelligent SIP Network:

•Presence Server - Presence Server - Accepts, stores, and distributes presence information. The Accepts, stores, and distributes presence information. The presence server has two distinct sets of clients:presence server has two distinct sets of clients:

- - PresentitiesPresentities (producers of information) provide presence information to the server to be stored and distributed

- - WatchersWatchers (consumers of information) receive presence information from the server•SIP proxy serverSIP proxy server - - A call-control device that provides many services such as routing of SIP messages between SIP user agents

• SIP redirect serverSIP redirect server - - A call-control device that provides routing information to user agents when requested, giving the user agent an alternate uniform resource identifier (URI) or destination user-agent server (UAS)

• SIP registrar serverSIP registrar server - - A device that stores the logical location of user agents within that domain or subdomain; a SIP registrar server stores the location of user agents and dynamically updates its data via REGISTER messages


• SIP location servicesSIP location services - - Additional functionality that can be used by proxy, redirect, and registrar servers to find the identity (with a unique URI) and "logical" location of user agents within the network (e.g. LDAP Directory Structures)

SIP Components (Continued)

• Back-to-back user agentBack-to-back user agent (B2BUA)(B2BUA) - - A call-control device that provides routing similar to a proxy server, but allows centralized control of the network call flows; this device allows SIP networks to replicate certain traditional telephony services that require centralized knowledge of device state, such as call park and pickup; this component is always dialog "stateful” (e.g. Avaya Communication Manager)

• Feature ServerFeature Server - - A application server that extends additional capabilities to a SIP UA in compliment to IETF defined services (e.g. Avaya Communication Manager)

• SIP-aware network devicesSIP-aware network devices - - Devices that have knowledge of the SIP protocol and allow the network to function more efficiently; this type of device might be a firewall or Network Address Translation (NAT) device that can allow SIP traffic to traverse network borders, or a load-balancing switch that allows requests to SIP servers to be more efficiently handled

• ENUM servicesENUM services – – Electronic Number Mapping, or ENUM, provides a method to encode telephone numbers (formally known as E.164 numbers) into the Domain Name System (DNS). ENUM is also used to map phone numbers to URI’s.


How Avaya Products Map to Industry SIP ComponentsSES = Proxy, Presence, Registrar, and Location serverSES = Proxy, Presence, Registrar, and Location serverIn addition provides SIP/SIMPLE IM ServicesIn addition provides SIP/SIMPLE IM Services & LDAP Directory Plug-Ins& LDAP Directory Plug-Ins for extendfor extendlocation/address translation serviceslocation/address translation services

Avaya IP Softphone & Avaya IP Agent = Hybrid soft client UA Uses H.323/H.248 for call signaling and SIP/SIMPLE for IM Services

Avaya 4600 SIP Phones = Wire Connected User Agents (UA)Wire Connected User Agents (UA)

BlackBerry/RIM 7270 = Wireless User Agent (UA) Example of SIP Stack on Handheld Devices with 802.11 connection

CM = B2BUA CM = B2BUA (back-to-back User Agent)(back-to-back User Agent) & Telephony Feature Server & Telephony Feature Server

Avaya SIP Softphone = Pure SIP Soft Client UAUses SIP for call signaling, SIP/SIMPLE for IM, can operate in Peer-to-Peer mode

SBC’s & Firewalls = SIP Aware Network Devices Provides –SIP/NAT, SIP QoS, Security, and ENUM Services (not avail. in all devices)

Session Border Controllers

Quick Edition = Peer-to-Peer (P2P) SIP serverless UAC/UAS withQuick Edition = Peer-to-Peer (P2P) SIP serverless UAC/UAS withOn-board feature/application services and SIP TrunkingOn-board feature/application services and SIP Trunking


Overview - Avaya’s Current SIP Proxy Based ArchitectureUser Access

Meeting ExchangeEnterprise

SES 4.0

SIPSoftphone

IPSoftphone

9600 Series

Third-Party

SIP-Enabled Applications

Expanded Meet Me Conferencing

SIP

SIP

SIP

Communication Manager

SIP

IPAgent

Third-Party SIP Application

Session BorderController

User Control

SIP/SIMPLEH.3

23

HTTPS

Web Browser

Handle-BasedDialing LDAP

Personal ProfileManager

SIPServices

Service ProviderSIP Network

SIPTrunking

SIP PersonalInformation Mgr

SIP/SIMPLEH.3

23

4600 Series

Mobile Devices


Peer-to-Peer (P2P) SIPPeer-to-Peer (P2P) SIP


Peer-to-Peer (P2P) SIP: The Future of “Collective” Communications Intelligence?

“Resistance is Futile …Prepare to be Assimilated”(Star Trek Next Generation “Borg” Collective Intelligence)


What is Peer-to-Peer (P2P) SIP Telephony?

Peer-to-Peer (P2P) technology overview:Dramatically different approach to communications from traditional client/server-based architecturesA “collective” (group) of intelligent nodes (peers), collaborate with one another to provide the services traditionally provided by a central server (or group of servers)Hive Mentality - Processing of services is dynamically allocated across all the intelligent nodes (Peers)Information about Peering Group automatically distributed as new nodes are “assimilated” (added) or dropped from the collectiveDependency on central servers is greatly reducedTypical platforms:

– “Smart IP hard phones” (e.g., Avaya Quick Edition)– P2P soft clients (e.g., Skype)

SIP P2PSIP P2PWork groupWork groupB2BUAB2BUA

PSTN


Example: Traditional SIP/IP PBX Architecture

IP/SIP TrunkIP/SIP TrunkIP/SIP TrunkIP/SIP Trunk IP/SIP TrunkIP/SIP Trunk

IP/SIP TrunkIP/SIP Trunk

IP or SIP TrunksIP or SIP TrunksWAN or LAN WAN or LAN

Network ConnectsNetwork Connects

“Unintelligent”SIP Endpoints




Border ControllerBorder Controller

Central SIP Proxy Central SIP Proxy &&

Applications ServicesApplications Services

Examples:

CM & SES

Attributes Centralized SIP Proxy / Registrar / Location Services

Centralized Call / Application Services

“Unintelligent” SIP endpoints (UACs) register with central SIP Proxy and derive application services from servers


P2P SIP Distributed IP Telephony Architecture(example)


Attributes Intelligent SIP endpoints (peer nodes)

Call services & applications distributed between peer nodes

Central SIP Proxy/Registrar replaced with distributed Proxy/Registrar services

P2P SIP is used to discover peer nodes within SIP domain and configure services

SIP trunks connect sites

SIP identities for all peer nodes are distributed to all peers within the SIP domain

“Super Nodes” may be used to support application load for “lesser” peers/nodes

Peer nodes join via ad hoc connectivity (no predefined client/server connections)

P2P SIP Distributed IP Telephony Architecture(example)


Why Peer-to-Peer (P2P) SIP Telephony?

Advantages:Distributed architecture minimizes traditional client/server central points of failure

Auto-discovery/auto-configuration capabilities simplify deployment & moves-adds-changes (MACs)

Serverless architecture reduces maintenance and operation costs

Considerations:Hard-phone processor and memory limitations can limit both application capabilities and the size of the peering group

Bandwidth and processing resources to support traffic to/from “Super Nodes” must be carefully planned for

Proper certificate-based security measures needed to prevent “rogue peers” from illegitimately joining peering groups

SIP P2PSIP P2PWork groupWork group

B2BUAB2BUA

PSTN


Combining Traditional SIP/IP PBX Architecture with P2P-SIP

Hybrid SolutionsHybrid Solutions– Hybrid solutions combine P2P to address small/medium-size

sites networked with enterprise or hosted SIP Proxy/Registrars and application services

AdvantagesAdvantages– Leverages Central Application Services where needed

– Improves Scalability

– Distributes Risk - Improves Survivability

– Supports Trunk Aggregation Strategies


B2BUAB2BUA

PSTN


P2P SIP Hybrid Hosted Services Architecture(example)

““Intelligent Edge”Intelligent Edge”P2PP2P

BranchBranchLocationsLocations

IP/SIP TrunkIP/SIP TrunkIP/SIP TrunkIP/SIP Trunk IP/SIP TrunkIP/SIP Trunk

IP/SIP TrunkIP/SIP Trunk

IP or SIP TrunksIP or SIP Trunks

Border ControllerBorder Controller

Hosted or Central SiteHosted or Central SitePSTN

Local & Long Distance Access

Other SIPOther SIPDomainsDomains

SIP Trunk(s)SIP Trunk(s)SIP Proxy/Registrar SIP Proxy/Registrar

&&Application ServersApplication Servers

oror


B2BUAB2BUA

PSTN


B2BUAB2BUA

PSTN


B2BUAB2BUA

PSTN


B2BUAB2BUA

PSTN









Vision Executed Vision Executed SIP Service ExamplesSIP Service Examples

(Live Demonstrations)(Live Demonstrations)


Branch LocationsBranch Locations

SIP,IP, Wireless, Digital & Analog SIP,IP, Wireless, Digital & Analog EndpointsEndpoints

SESSES

CommunicationCommunicationManagerManager

CommunicationCommunicationManagerManager

SIP Trunks

Avaya Hub Environment Avaya Hub Environment Coppell, TXCoppell, TX

SIP Routing SIP Routing between between

SitesSites

ISPISPInternet (VPN)Internet (VPN)

Example #1 - Avaya Quick Edition with SIP TrunkingExample #1 - Avaya Quick Edition with SIP Trunking

GatewayGatewayGatewayGateway

Juniper Netscreen – 5GTSSG appliance

PSTN

IP/VPNIP/VPNSIP TrunkSIP Trunk

PSTN PSTN PSTN

B2BUAB2BUAB2BUAB2BUAB2BUAB2BUA B2BUAB2BUA

B2BUAB2BUA

T.38FAX

Laptop with QE Multisite Provisioning Tool &

Avaya SIP Softphone

Acme PacketAcme PacketSession Border ControllerSession Border Controller

Service Service ProviderProvider

SIP NetworkSIP Network

SIP TrunkSIP Trunk

PSTNLocal & Long Distance

Cell Network Access




B2BUAB2BUA

PSTNLocal & Long Distance

Cell Network Access

JuniperJuniperJ2300-2J2300-2

Series RouterSeries Router


IP/SIP Trunk VPN WAN Connection

to DemoAvayaLAN Cat5 or better cabling

G10 or G11G10 or G11Four port PSTN GatewayFour port PSTN Gateway (Analog Loop Start Lines)

Laptop PC Demo PlatformLaptop PC Demo Platform

NetGear FS108P PoE SwitchNetGear FS108P PoE Switch

• LAN Backbone for Demo Environment• (4) IEEE 802.3af Power over Ethernet Ports • (4) Non-Powered 10/100 Ethernet Ports

Quick Edition Quick Edition SIP Trunk Branch EnvironmentSIP Trunk Branch Environment

Avaya Quick EditionAvaya Quick Editionon 4610SW and/or 4621SW IP telephoneson 4610SW and/or 4621SW IP telephones

Ext. 200 Ext. 201 Ext. 202

Powered PortsNon-powered Ports

• SIP Trunk Created to DemoAvaya.com (10.147.12.7) • SIP Identity register at DemoAvaya.com (2475)• “Customer Service” Workgroup linked to SIP Identity (ext. 204)• “Global” outbound dialing over SIP trunk enabled

“Trusted” NetworkConnection

Juniper NetScreen 5GTJuniper NetScreen 5GTIPSec VPN and FirewallIPSec VPN and Firewall

“Untrusted” NetworkConnection

• Provides DHCP Service to Trusted Network Devices (10.10.200.x range)• Creates Secure VPN tunnel to DemoAvaya.com Site • Takes IP Address on Untrusted Port from DHCP service on Network Router

ROUTER, Cable Modem,ADSL Modem, or

Integrated Access Device (IAD)

• one-X Desktop Edition (SIP Softphone client) serves as a CM extension on Demo Avaya CM Switch• Multisite Provisioning Tool with local & DemoAvaya QE Branch registration• QE Web Admin for Site Administration

PSTN


LAN Cat5 or better cabling

G10 or G11G10 or G11Four port PSTN GatewayFour port PSTN Gateway (Analog Loop Start Lines)

Laptop PC Demo PlatformLaptop PC Demo Platform

NetGear FS108P PoE SwitchNetGear FS108P PoE Switch

• LAN Backbone for Demo Environment• (4) IEEE 802.3af Power over Ethernet Ports • (4) Non-Powered 10/100 Ethernet Ports

Avaya Quick EditionAvaya Quick Editionon 4610SW and/or 4621SW IP telephoneson 4610SW and/or 4621SW IP telephones

Ext. 200 Ext. 201 Ext. 202

Powered PortsNon-powered Ports

• SIP Trunk Created to DemoAvaya.com (10.147.12.7) • SIP Identity register at DemoAvaya.com (2475)• “Customer Service” Workgroup linked to SIP Identity (ext. 204)• “Global” outbound dialing over SIP trunk enabled

“Trusted” NetworkConnection “Untrusted” Network

Connection

Juniper NetScreen 5GTJuniper NetScreen 5GTIPSec VPN and FirewallIPSec VPN and Firewall

• Provides DHCP Service to Trusted Network Devices (10.10.210.x range)• Creates Secure VPN tunnel to DemoAvaya.com Site • Takes IP Address on Untrusted Port from DHCP service on Router

• one-X Desktop Edition (SIP Softphone client) serves as a CM extension on Demo Avaya CM Switch• Multisite Provisioning Tool with local & DemoAvaya QE Branch registration• QE Web Admin for Site Administration

PSTNIP/SIP Trunk

VPN WAN Connectionto DemoAvaya

Quick Edition Quick Edition SIP Trunk Branch EnvironmentSIP Trunk Branch Environment

(Mobile Broadband Router Option)(Mobile Broadband Router Option)

EV

DO

Top Global MB6800Top Global MB6800Mobile Broadband RouterMobile Broadband Router

Verizon Wireless Broadband

Service

• Provides EVDO Broadband Service• Provides IPSEC/VPN pass-through• Provides DHCP to Untrusted Port


Group “Customer Service”Local Ext. 475 (stations 200,201,202)Linked to SIP Identity = 2475

SIP Enablement Services (SES)SIP Proxy/Registrar

demoavaya.com SIP Domain

Demo Avaya

Communication Manager

Coppell, Texas

Demo Kit Site QE Branch, CA

PSTN

200 201

202CM Ext. 1022

On Demo Avaya CM Switch

SIP Trunks

Demo Avaya

QE Branch, Texas

CM ext. 2311

Cell Phone(972)457-2011in Coppell, Texas

SIP Trunk connections to SES from Quick Edition branch sites can be used to create communication session between Quick Edition Branch locations using the registered SIP Identities to make calls - EXAMPLE:EXAMPLE:

With SIP trunks in place and SIP Identities defined, calls can be placed from CM to Quick Edition Auto Attendant, Station Groups, or individual QE extensions using abbreviated dialing schemes EXAMPLE:EXAMPLE:

IP Trunk

PSTN

Quick Edition Branch in California calls Demo Avaya Quick Edition branch’s Auto Attendant (ext. 500) linked to SIP Identity 2474 by Dialing “82474”Dialing “82474”

24

74

24

75

Registered SIPIdentities

SIP Trunk connections to SES from Quick Edition branch sites can also be used to create communication session between Quick Edition Branch extensions and CM extensions - EXAMPLE:EXAMPLE:

500 AA

QE Auto Attendant Local Ext. 500Linked to SIP Identity = 2474

Quick Edition ext. 201 in California calls Demo Avaya Communication Manager extension 2311 over the SIP Trunk by Dialing “82311”Dialing “82311”

Call Path ExamplesCall Path Examples

CM Extension 2311 calls the “Customer Service” QE station group (extensions 200, 201, and 202) using the SIP Identity linked with that group to make the call by Dialing “2475”Dialing “2475”


200 201

202


200 201

202500 AA

QE Auto Attendant Local Ext. 500Linked to SIP Identity = 2474

CM ext. 2311

Quick Edition branch sites can also use PSTN trunks connected to Communication Manager to make calls to PSTN stations using the SIP Trunk for long distance toll-bypass - EXAMPLE:EXAMPLE:

Quick Edition ext. 201 in California calls Cell Phone in Texas over the SIP Trunk by Dialing “894572011”Dialing “894572011”



CM ext. 2311


Example #2 – Avaya Integration with Example #2 – Avaya Integration with Microsoft Office CommunicatorMicrosoft Office Communicator


Voice Telephony IntegrationVoice Telephony Integration

Feb 2007

IP Softphone 6.0

on local PC

LCS (OCS) /MOC integration with Avaya CM –

Click to call / call control of Avaya end points from MOC and SmartTags in Office apps.

Adds Avaya functions to MOC tabs.

LCS 2005 and OCS 2007Communication Manager

2007SESMOC as SIP Softphone

Avaya provides Enterprise Gateway for MOC

Feb 2007

Application Enablement Services 4.0

nothing required on PC

LCS (OCS) /MOC integration with Avaya CM –

Click to call / call control of Avaya end points from MOC and SmartTags in Office apps

Network SIP/CSTA gateway

Additional Avaya functions on MOC tabs some months later

Simple peer-to-peer conversations

Not enterprise voice

No coverage for voice mail

No PBX rules (class of service)

Decades of experience delivering enterprise class telephony

48© 2007 Avaya Inc. All rights reserved. 48© 2006 Avaya Inc. All rights reserved.Avaya – Proprietary & Confidential. For Limited Internal Distribution. The information contained in this document may not be distributed or reproduced, in whole or in part.

Communicator will be on PC

– Corporate IM is driving

Avaya Provides Telephony Connection

Click to Call in MOC, SmartTags

Escalate IM to call, Conference

Phone & MOC stay in synch

– Presence is shared

Call Control - Hold,Transfer,etc.

– 17 functions (CSTA)

Avaya Brings Enterprise Telephony to CommunicatorAvaya Brings Enterprise Telephony to Communicator

Additional Avaya Functions provided as extensions to Office Communicator on tabbed UI


SIP

/SIM

PLE

uaC

ST

A

(CS

TA

ove

r S

IP)

Microsoft LCS 2005 - or

Microsoft OCS 2007

Microsoft Reference Model

Implements SIP/CSTA (TR/87) gateway on AE Services 4.0 Server

Desktop Call Control

– MOC can control phone

– Use phone and MOC is updated

Server Solution = no local software required. Can use MOC alone

Optionally, can add IP Softphone on PC

– Shared Control

– Telecommuter

– RoadWarriorCell Phones(EC500)

TDM / Analog

H.323

Avaya CM

uaCSTA (CSTA over SIP)

Avaya CTI

Application Enablement ServicesApplication Enablement Services

AE Services 4.0

Avaya protocol


SIP

/SIM

PL

E

uaC

ST

A

uaC

ST

A

(CS

TA

ove

r S

IP)

(CS

TA

ove

r S

IP)

Microsoft LCS 2005 - or

Microsoft OCS 2007

Cell Cell PhonesPhones(EC500)(EC500)

TDM / TDM / AnalogAnalog

H.323H.323

AvayaAvaya CM CM

IP Softphone - LCS integrationIP Softphone - LCS integration

uaC

ST

A

uaC

ST

A

(CS

TA

ove

r S

IP

(CS

TA

ove

r S

IP)

Avaya protocol

Avaya protocol

Part of upcoming release of IP Softphone 6.0

– SIP/CSTA gateway on local Windows client

– Softphone UI hidden if using MOC

– Requires small server applet (can be on LCS server)

Client side solution

Desktop Call Control

– MOC can control phone

– Use phone and MOC is updated

Added Avaya features on MOC tabs – beyond TR/87

– EC500, History, Video


Greater than the sum of the parts Greater than the sum of the parts

Telephony & desktop presence synch

Emails for missed calls

MOC interface controlling your home / mobile phone

Enterprise IM

– Active Directory Identities

– Multi-party IMs

– Archived, searchable

Microsoft desktop

–User interface

– SmartTags

Enterprise Telephony

– Mature, five 9’s reliable

Video & Conferencing

– Richer, more mature

Mobility

– Follow Me / EC500


Q & A

53© 2007 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. Under NDA


Backup SlidesBackup Slides


Why SIP – Top 10 Reasons:Why SIP – Top 10 Reasons:

Presence Based Communications:Presence Based Communications:– SIP adds intelligence to communications by allowing users, as well as applications, to intelligently connect

parties based on their Presence (registered availability) in the enterprise. This concept is best exemplified through SIP's ability to support "intelligent forking which is the ability to route communications to the right persons, in the right medium (voice, video, IM), on the right device, and at the right time.

Preference Based Communications:Preference Based Communications:– Like SIP Presence, SIP adds intelligence to communications through allowing users to control the parameters

by which they can be communicated with (e.g. time-of-day, preferred medium, preferred users, etc). This concept is best exemplified through SIP's ability to support "buddy list" based communications.

SIP is an open standard:SIP is an open standard:– The SIP standard is defined in RFC 3261 by the Internet Engineering Task Force (IETF). The IETF is a large

open international community of network designers, operators, vendors, and researchers concerned with the evolution of the Internet architecture and the smooth operation of the Internet.

Interoperability:Interoperability:– Several working groups, including SIPIT, SIP Foundry, and SIP Connect, arrange events where companies with

SIP-based hardware and software products can best interoperability with other SIP-based products. This helps to promote smoother integration of SIP-based products in enterprise networks.

Uniform Addressing:Uniform Addressing:– SIP URI (Uniform Resource Identifier) addressing provides a unifying identifier that can be used for routing all

communication to a user. This eliminates the need for tracking users’ multiple phone numbers, email addresses, and IM contact names. Simply put, SIP URI allows for a single user identity to be mapped across multiple devices which facilitates the ability for people to connect with people, without needing to know which devices they have and are presently using.


Why SIP – Top 10 Reasons Why SIP – Top 10 Reasons (continued):(continued):

Operation Cost Savings / SIP TrunkingOperation Cost Savings / SIP Trunking– SIP provides a low cost trunking alternative to standard PSTN transport. SIP trunks can be used to facilitate

communication models that leverage the possibilities expressed in the previous points. SIP trunks support the concepts of converged communications thus allowing for true mutli-media communication streams to exist on a common carrier circuit. Finally, SIP trunks reduce operational costs by allowing the user to eliminate hardware, software, and recurrent network charges associated with using traditional PSTN trunks for voice communications.

Simplified Communication Architecture:Simplified Communication Architecture:– At the foundation of SIP's philosophy is the concept that intelligence in the communication enterprise should reside in

the endpoint. This concept is manifested in SIP's ability to support peer-to-peer communication architectures. Peer-to-Peer communications environments do not rely on communications servers, switches, or other intermediate devices to support communications between users. Because peer-to-peer is, in essence, "switchless" by its nature, simple easy-to-configure communication environments can be created which use only intelligence endpoints as the mechanism for establishing a communications enterprise. This architecture is best exemplified by the technology represented by Avaya's NIMCAT based solutions.

Creation of New Services:Creation of New Services:– SIP is a structured, text-based protocol that is modeled after HTTP, or HyperText Transport Protocol, the language that

powers the World Wide Web. Because SIP is text-based and similar to HTTP, application developers and system engineers will have an easier time developing applications and integrating applications into complex communications environments.

Ease of Support and Implementation:Ease of Support and Implementation:– Since SIP is text-based and modeled after HTTP and XML it is much easier to learn and troubleshoot/support. From

analyzing network packets to application code, SIP’s structured language will stand out and be easily understood and interpreted.

Native Mobility:Native Mobility:– SIP’s awareness of a user’s communication capabilities will aid international travelers who must use different (or multi-

modal) cell phones and other messaging devices and protocols in different countries. A caller who is trying to locate such a traveler need not know the traveler’s availability or location: SIP by nature will know how a person can be reached, and facilitate the connection without the calling party’s need to know where the traveler is or how he (or she) can be contacted.


What is SIP? What is SIP? (Longhand explanation)(Longhand explanation) http://www.ietf.org/html.charters/sip-charter.htmlhttp://www.ietf.org/html.charters/sip-charter.html

“SIP is an IETF application layer-protocol that can establish, modify, and terminate multimedia sessions” (summary definition - RFC 3261)

Media agnosticMedia agnostic– Voice, video, instant messaging, etc.

Media negotiationMedia negotiation– Offer-Answer model (Invite & Acknowledge)

Similar to HTTPSimilar to HTTP– Request-Response model– Text message-based protocol

• Easy to debug

Reuses other IETF protocolsReuses other IETF protocols– UDP, TCP, TLS, DHCP, DNS, SDP, RTP, MIME, etc.

Communication sessions based on “presence”Communication sessions based on “presence”

– I.e. the publication of your willingness and ability to be communicated with

– Note that Presence publication is selective by user “preference”

SIP is a very simple protocol with very few client/server style messages


SES 3.0

HTTPS

Personal ProfileManager (PPM)

SIPServices

SIP PersonalInformation Mgr

When the user logs into the phone, the latest parameters are loaded in

automatically

SOAP/HTTPSSOAP/HTTPS

SIP Personal Information Manager allows the user to access and

modify their profiles, access control lists, contact lists and device

parameters through a secure web browser

PPM receives/stores/distributes:• Contact and group list management• Access Control Lists for user

presence• Device Parameters (i.e. dial plan,

speed dial list, feature button mappings, etc)

What is SOAP?

SOAP (Simple Object Access Protocol) is a way for a program running in one kind SOAP (Simple Object Access Protocol) is a way for a program running in one kind of operating system (such as Windows 2000) to communicate with a program in of operating system (such as Windows 2000) to communicate with a program in the same or another kind of an operating system (such as Linux) by using the the same or another kind of an operating system (such as Linux) by using the World Wide Web's Hypertext Transfer Protocol (HTTP) and its Extensible Markup World Wide Web's Hypertext Transfer Protocol (HTTP) and its Extensible Markup Language (XML) as the mechanisms for information exchange. Language (XML) as the mechanisms for information exchange.

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