Understanding SIP’s Role in Intelligent Communications

Understanding SIP’s Role in Intelligent Communications
Tom Doria Director – Avaya P2P Technical Business Development Chair – Avaya SIP Virtual Team

Agenda Intelligent Communications & SIP Defining SIP Why SIP
The Changing World & Evolving Business Needs Understanding Avaya’s Vision for Intelligent Communications The Building Blocks to Intelligent Communication Defining SIP IETF’s Vision for SIP Key Concepts (SIP, SIPing, SIMPLE) Why SIP Multi-Vendor Interoperability Presence and Preference Native Support for Multimedia Single User Identity Why SIP Trunking Building a SIP Enabled Enterprise SIP Communication Components Defined Mapping Industry Names to Avaya Solutions Peer-to-Peer (P2P) SIP Vision Executed - SIP Service Examples Example #1 - Avaya Quick Edition with SIP Trunking Example #2 – Avaya Integration with Microsoft Office Communicator

from to from to from to The World Is Changing
Separate services: Local, long distance, mobile, video, Internet to Bundled services: All distance voice, voice/data/video packages Multi-use devices: Blackberries; mobile phones with messaging, , video; computer for Internet phone and IM There’s no doubt that the telecom market is rapidly changing. In fact, we are experiencing a fundamental shift in the way customers select their communications products and services, and from whom. The idea of traditional companies offering traditional services is falling by the wayside. Today’s customer doesn’t want the hassle of having to call one provider for phone service, another for their video, another for Internet. He or she wants one provider for all communications and entertainment services. At the same time, we are witnessing a technology evolution in which communications services are shifting from separate networks to integrated offerings, virtually seamlessly, no matter what device is used. To today’s customer, it’s all about convenience, ease of use, and mobility. And he or she doesn’t really care if services are coming from “a phone company, “ a cable provider, or something in-between. from Separate platforms: Phone for voice, computer for Internet to Separate providers: Cable companies for video, phone companies for voice Multi-product providers: Cable, mobile, and wireline companies offering voice/video/data from to

State of Convergence Today! End user view
We Need Converged Communication Applications that Improve Productivity and Allow Service Integration

Avaya’s Vision for a New Era of Intelligent Communications
Intelligent Communication Solutions should: 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

What does Intelligent Communications Look Like?
Contact Centers Collaboration Unified Communication Conferencing Telephony Mobility & Softphone Voice Messaging Instant Messaging

Avaya Intelligent Communications The Building Blocks to Next Generation Intelligent Communication Solutions Session Initiation Protocol (SIP) Next Generation Carrier Services Service Oriented Architecture (SOA)

Intelligent Communications Redefining the way People, Processes, and Information Connect
Application Protocol Fabric Intelligent Routing of Communication & Services Supports Open Integration of Devices and Application Services Native Support for Multimedia & Multimodal Communications NEXT GEN Carrier Services Mobile and Fixed Services Integrated Supports Rich Presence SIP Enriched Multimedia Hosted Apps. IP & SIP Trunk IMS 3G & 4G Wireless SIP SOA Carrier Services INTELLIGENCE Between Applications & User Middleware Connects Silos of Application & Communication Services Event Driven Services Initiate Communication with Users or Other Applications

Defining Key Concepts SIP, SIPPING, SIMPLE and other words that begin with “S”

What is SIP? Session Initiation Protocol
Shorthand Definition Session Initiation Protocol 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 “presence” I.e. the publication of your willingness and ability to be communicated with Note that Presence publication is selective by user “preference” IETF Site:

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

SIP Services Examples a.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? Introduces “Presence” into communications state
SIP for Instant Messaging and Presence Leveraging Extensions IETF working group Introduces “Presence” into communications state Builds on RFC 3265 Now a standard: RFC 3856 Terminology 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:

What is a SIP Trunk? Shorthand Definition 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?

SIP Is a Key Enabler of Intelligent Communications
The Right Person, in the Right Place, at the Right Time, the Right Way Multi-Vendor Interoperability Presence and Preference Native Support for Multimedia Just as SIP is driving technology for our product lines, it is also a key enabler of Intelligent Communications. Intelligent Communications often involves multiple communication system components, sometimes even from different vendors. SIP is a standard that enables multi-vendor interoperability. Reducing or eliminating human latency from business processes requires real-time awareness of user’s presence and orchestration of real-time events. SIP has mechanisms for that. Unified Communications encompasses Instant Messaging, video, conferencing, and voice. SIP supports all media. Part of the reason CTI exists is to coordinate data with communications. Because SIP is modeled off of HTTP, data can now be delivered within the SIP signaling. Single User Identity

Multi-Vendor Interoperability
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, , corporate data & application access Works over any b/g WLAN Example: Nokia Dual-Mode Phones Support for cellular and wireless VoIP Embedded SIP client Graphical user controls of desktop on cellular phone RIM Blackberry 7270 Nokia Dual-Mode Avaya supports and will troubleshoot SES with select third party SIP clients to the extent they have been tested by Avaya’s SITL, Solution and Interoperability Test Lab, and documented in Application Notes. However, Avaya does not provide installation, implementation, configuration, or maintenance of any third party SIP end points. All of these below have been tested officially (***) or at least played with: See for the latest Cisco 7940 / 7960 *** Cisco SIP Proxy Server *** Cisco AS5400 *** RIM Blackberry 7270 *** Counterpath (Xten) Eyebeam *** Snom SIP Telephones *** Grandstream SIP Telephone Nuance Voice Platform *** Ingate SIParator *** Kagoor VoiceFlow *** Acme Packet Session Director *** NexTone SBC *** Jasomi PeerPoint *** Meru Networks Access Point w/ Hitachi WIP-5000

Presence and Preference
SIP 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 World Selectively advertise your Presence (buddy lists) Monitor key individuals availability Select your preferred mode of communication © 2006 Avaya Inc. All rights reserved. 20 Avaya – Proprietary & Confidential. For Limited Internal Distribution. The information contained in this document may not be distributed or reproduced, in whole or in part.

Intelligent Communications & Customer Service Scenario
IP Agent with SIP/SIMPLE IM Services Intelligent Communications & Customer Service Scenario 1. Customer calls Customer Service 2. Rep needs technical answer from expert 3. Rep uses presence to “peek over the cubicle” to see if expert available 6. Rep notices expert has ended conversation And has expert join conference with customer 7. Expert answers questions and continues IM chats with the Rep in the background IM 4. Expert available online but busy on phone 5. Rep IM’s the expert - begins to get answers IM

Native Support for Multimedia
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 multimedia Solutions from Different Vendors

Connect people with people, not with devices
Single User Identity Connect people with people, not with devices Single user identity: SIP Address Of Record (AOR) is mapped across multiple devices (e.g. or SIP Presence and Preference controls can be used to automatically route communications to the preferred device that is being “used” Instant Messaging Softphone Traditional phone SIP Phone PDA (AOR) SIP WiFi Phone

Limitation & Challenges of Private IP (H.323) Trunking
Why SIP Trunking? Limitation & Challenges of Private IP (H.323) Trunking 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 PSTN Local & Long Distance Customers/ Partners/ Suppliers TDM interfaces External Communications TDM interfaces IP WAN External Communications IP Data &Telephony LAN LAN Internal Communications IP Phone IP Phone IP Phone IP Phone

SIP Trunking A Single Pipe to the Cloud
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 PSTN Local & Long Distance Customers/ Partners/ Suppliers SIP SIP Service Provider’s WAN H.323 LAN LAN IP Phone IP Phone IP Phone IP Phone

SIP Infrastructure Components
The building blocks of an Intelligent SIP Network: SIP 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 Presence Server - Accepts, stores, and distributes presence information. The presence server has two distinct sets of clients: - Presentities (producers of information) provide presence information to the server to be stored and distributed - Watchers (consumers of information) receive presence information from the server SIP proxy server - A call-control device that provides many services such as routing of SIP messages between SIP user agents SIP 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 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 Components (Continued)
SIP 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) Back-to-back user agent (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 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 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 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 Components
SES = Proxy, Presence, Registrar, and Location server In addition provides SIP/SIMPLE IM Services & LDAP Directory Plug-Ins for extend location/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) BlackBerry/RIM 7270 = Wireless User Agent (UA) Example of SIP Stack on Handheld Devices with connection CM = B2BUA (back-to-back User Agent) & Telephony Feature Server Avaya SIP Softphone = Pure SIP Soft Client UA Uses 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 with On-board feature/application services and SIP Trunking

Overview - Avaya’s Current SIP Proxy Based Architecture
User Access User Control 9600 Series 4600 Series Mobile Devices Third-Party SIP Softphone Web Browser IP Agent IP Softphone SIP/SIMPLEH.323 SIP/SIMPLEH.323 SES 4.0 SIP Service Provider SIP Network SIP Trunking SIP Personal Information Mgr Personal Profile Manager HTTPS SIP Services Handle-Based Dialing Session Border Controller LDAP SIP SIP-Enabled Applications SIP SIP Expanded Meet Me Conferencing Meeting Exchange Enterprise Third-Party SIP Application Communication Manager

Peer-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 architectures A “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 collective Dependency on central servers is greatly reduced Typical platforms: “Smart IP hard phones” (e.g., Avaya Quick Edition) P2P soft clients (e.g., Skype) SIP P2P Work group B2BUA PSTN

Example: Traditional SIP/IP PBX Architecture
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 IP/SIP Trunk IP or SIP Trunks WAN or LAN Network Connects “Unintelligent” SIP Endpoints Border Controller Central SIP Proxy & Applications Services Examples: CM & SES

P2P SIP Distributed IP Telephony Architecture (example)

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)

Why Peer-to-Peer (P2P) SIP Telephony?
SIP P2P Work group B2BUA PSTN 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

Combining Traditional SIP/IP PBX Architecture with P2P-SIP
Hybrid Solutions Hybrid solutions combine P2P to address small/medium-size sites networked with enterprise or hosted SIP Proxy/Registrars and application services Advantages Leverages Central Application Services where needed Improves Scalability Distributes Risk - Improves Survivability Supports Trunk Aggregation Strategies SIP P2P Work group B2BUA PSTN

P2P SIP Hybrid Hosted Services Architecture (example)
“Intelligent Edge” P2P Branch Locations IP/SIP Trunk IP or SIP Trunks Border Controller Hosted or Central Site PSTN Local & Long Distance Access Other SIP Domains SIP Trunk(s) SIP Proxy/Registrar & Application Servers or SIP P2P Work group B2BUA

(Live Demonstrations)
Vision Executed SIP Service Examples (Live Demonstrations)

Example #1 - Avaya Quick Edition with SIP Trunking
Avaya Hub Environment Coppell, TX Communication Manager SIP,IP, Wireless, Digital & Analog Endpoints Service Provider SIP Network B2BUA B2BUA SIP Trunks Acme Packet Session Border Controller SES T.38 FAX SIP Trunk SIP Routing between Sites Juniper J2300-2 Series Router Gateway PSTN Local & Long Distance Cell Network Access PSTN Local & Long Distance Cell Network Access ISP Internet (VPN) Juniper Netscreen – 5GT SSG appliance IP/VPN SIP Trunk IP/VPN SIP Trunk IP/VPN SIP Trunk IP/VPN SIP Trunk B2BUA B2BUA B2BUA B2BUA Laptop with QE Multisite Provisioning Tool & Avaya SIP Softphone Branch Locations PSTN PSTN PSTN PSTN

Quick Edition SIP Trunk Branch Environment PSTN
Laptop PC Demo Platform 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 Avaya Quick Edition on 4610SW and/or 4621SW IP telephones PSTN SIP Trunk Created to DemoAvaya.com ( ) SIP Identity register at DemoAvaya.com (2475) “Customer Service” Workgroup linked to SIP Identity (ext. 204) “Global” outbound dialing over SIP trunk enabled G10 or G11 Four port PSTN Gateway (Analog Loop Start Lines) Ext. 200 Ext. 201 Ext. 202 IP/SIP Trunk VPN WAN Connection to DemoAvaya LAN Cat5 or better cabling “Trusted” Network Connection “Untrusted” Network Connection Non-powered Ports Powered Ports ROUTER, Cable Modem, ADSL Modem, or Integrated Access Device (IAD) NetGear FS108P PoE Switch LAN Backbone for Demo Environment (4) IEEE 802.3af Power over Ethernet Ports (4) Non-Powered 10/100 Ethernet Ports Juniper NetScreen 5GT IPSec VPN and Firewall Provides DHCP Service to Trusted Network Devices ( x range) Creates Secure VPN tunnel to DemoAvaya.com Site Takes IP Address on Untrusted Port from DHCP service on Network Router

Quick Edition SIP Trunk Branch Environment PSTN
(Mobile Broadband Router Option) Laptop PC Demo Platform 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 Avaya Quick Edition on 4610SW and/or 4621SW IP telephones PSTN IP/SIP Trunk VPN WAN Connection to DemoAvaya SIP Trunk Created to DemoAvaya.com ( ) SIP Identity register at DemoAvaya.com (2475) “Customer Service” Workgroup linked to SIP Identity (ext. 204) “Global” outbound dialing over SIP trunk enabled G10 or G11 Four port PSTN Gateway (Analog Loop Start Lines) EVDO Ext. 200 Ext. 201 Ext. 202 Verizon Wireless Broadband Service LAN Cat5 or better cabling “Trusted” Network Connection “Untrusted” Network Connection Non-powered Ports Powered Ports Top Global MB6800 Mobile Broadband Router NetGear FS108P PoE Switch LAN Backbone for Demo Environment (4) IEEE 802.3af Power over Ethernet Ports (4) Non-Powered 10/100 Ethernet Ports Juniper NetScreen 5GT IPSec VPN and Firewall Provides EVDO Broadband Service Provides IPSEC/VPN pass-through Provides DHCP to Untrusted Port Provides DHCP Service to Trusted Network Devices ( x range) Creates Secure VPN tunnel to DemoAvaya.com Site Takes IP Address on Untrusted Port from DHCP service on Router

Demo Kit Site QE Branch, CA
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: Call Path Examples PSTN Demo Avaya Communication Manager Coppell, Texas Cell Phone (972) in Coppell, Texas Cell Phone (972) in Coppell, Texas 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” Cell Phone (972) in Coppell, Texas CM ext. 2311 CM ext. 2311 Group “Customer Service” Local Ext. 475 (stations 200,201,202) Linked to SIP Identity = 2475 200 201 202 Group “Customer Service” Local Ext. 475 (stations 200,201,202) Linked to SIP Identity = 2475 200 201 202 CM ext. 2311 IP Trunk 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: SIP Enablement Services (SES) SIP Proxy/Registrar demoavaya.com SIP Domain Quick Edition Branch in California calls Demo Avaya Quick Edition branch’s Auto Attendant (ext. 500) linked to SIP Identity 2474 by Dialing “82474” 500 AA QE Auto Attendant Local Ext. 500 Linked to SIP Identity = 2474 SIP Trunks 2474 Registered SIP Identities 2475 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: Demo Avaya QE Branch, Texas Demo Kit Site QE Branch, CA 200 201 202 CM Ext. 1022 On Demo Avaya CM Switch Quick Edition ext. 201 in California calls Demo Avaya Communication Manager extension 2311 over the SIP Trunk by Dialing “82311” PSTN 500 AA QE Auto Attendant Local Ext. 500 Linked to SIP Identity = 2474 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: Group “Customer Service” Local Ext. 475 (stations 200,201,202) Linked to SIP Identity = 2475 Quick Edition ext. 201 in California calls Cell Phone in Texas over the SIP Trunk by Dialing “ ”

Example #2 – Avaya Integration with Microsoft Office Communicator
Avaya’s Unified Communications solutions integrate business communications applications, systems and services in a reliable and secure fashion. The result is a superior, seamless user experience across all Avaya solutions regardless of location, network, or device. 46

Voice Telephony Integration
Communication Manager LCS 2005 and OCS 2007 Decades of experience delivering enterprise class telephony Simple peer-to-peer conversations Not enterprise voice No coverage for voice mail No PBX rules (class of service) 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 Application Enablement Services 4.0 nothing required on PC Feb 2007 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. IP Softphone 6.0 on local PC Feb 2007 MOC as SIP Softphone Avaya provides Enterprise Gateway for MOC SES 2007

Avaya Brings Enterprise Telephony to Communicator
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) Additional Avaya Functions provided as extensions to Office Communicator on tabbed UI © 2006 Avaya Inc. All rights reserved. 48 Avaya – Proprietary & Confidential. For Limited Internal Distribution. The information contained in this document may not be distributed or reproduced, in whole or in part.

Application Enablement Services
uaCSTA (CSTA over SIP) Avaya CTI Application Enablement Services AE Services 4.0 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 RoadWarrior Microsoft LCS or Microsoft OCS 2007 SIP/SIMPLE uaCSTA (CSTA over SIP) Avaya protocol Avaya CM Cell Phones (EC500) TDM / Analog H.323

IP Softphone - LCS integration
uaCSTA (CSTA over SIP) Client side solution 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) Microsoft LCS or Microsoft OCS 2007 SIP/SIMPLE uaCSTA (CSTA over SIP) 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 Avaya protocol Avaya CM Cell Phones (EC500) TDM / Analog H.323

Greater than the sum of the parts
Enterprise Telephony Mature, five 9’s reliable Video & Conferencing Richer, more mature Mobility Follow Me / EC500 Enterprise IM Active Directory Identities Multi-party IMs Archived, searchable Microsoft desktop User interface SmartTags Telephony & desktop presence synch s for missed calls MOC interface controlling your home / mobile phone

Backup Slides

Why SIP – Top 10 Reasons: 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: 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: 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: 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: 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, 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 (continued):
Operation 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: 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: 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: 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: 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. (Longhand explanation) http://www. ietf. org/html
“SIP is an IETF application layer-protocol that can establish, modify, and terminate multimedia sessions” (summary definition - RFC 3261) Media agnostic Voice, video, instant messaging, etc. Media negotiation Offer-Answer model (Invite & Acknowledge) Similar to HTTP Request-Response model Text message-based protocol Easy to debug Reuses other IETF protocols UDP, TCP, TLS, DHCP, DNS, SDP, RTP, MIME, etc. 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

What is SOAP? 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 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 Language (XML) as the mechanisms for information exchange. 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 When the user logs into the phone, the latest parameters are loaded in automatically SOAP/HTTPS HTTPS SES 3.0 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) SIP Personal Information Mgr Personal Profile Manager (PPM) SIP Services

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