Ubiquity Software CorporationUbiquity Software Corporation

~ SIP ~Simple Protocol - Profound Implications

Working Agenda

Introduction to Ubiquity Software CorporationAn overview of the Session Initiation Protocol (SIP)SIP in the marketplaceImplications for QwestWorldwide service provider SIP initiativesHow can Ubiquity help?Going forwardQuestion & answer session

Introduction to Ubiquity Six years of experience developing advanced telephony applications for

service providers Offices in US; UK and Canada Management team / directors include recognized authorities of SIP

Technology: Michael Doyle – CTO Professor Henning Schulzrinne - Columbia University (Board Member) Martin De Prycker – CTO, Alcatel (Board Member)

Raised US$42 million in venture capital - August 2000 CapVest Equity Partners Fund, L.P; Celtic House International; JK&B Capital; Alcatel

Recognized authorities in signaling and programming languages SIP; JAVA

Active in many associated standards bodies and working groups IETF; SIP; SOAP; JAIN SIP LITE

Founders of the SIP Center www.sipcenter.com Co-authors of SIPstone (SIP server performance benchmarking) First to enable SIP click-to-dial from within Microsoft applications

Current Relationship With Qwest

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Henning Schulzrinne Associate Professor, Columbia University

Department of Computer Science and Electrical Engineering

Ubiquity Software, Corp. Board MemberSince March 2001

Acknowledged as the architect of SIP Co-Authored RFC2543 with aid of student and

colleaguesOther related experience Includes:

Internet telephony; Internet multimedia; quality-of-service; mobility; security

Other co-authored RFC’s Include:RTSP & RTP

A Brief History Of SIP

Feb. 1996: earliest Internet draftsFeb. 1999: Proposed StandardMarch 1999: RFC 2543April 1999: first SIP bake-offNovember 2000: SIP accepted as 3GPP signaling

protocolDecember 2001: 6th bake-off, 200+ participantsMarch 2001: 7th bake-off, first time outside U.S.

VoIP Signaling Architectures

MGCP, Megaco = master / slaveH.323 = (Mostly) single administrative domainSIP = Peer-to-peer, cross domain

VoIP ArchitecturesFeature SIP H.323 Megaco/MGCP

Multiple Domains X ? -

Third-Party Control X - Single-domain

Multimedia X Fixed Set Unlikely

End System Control X X -

Extensible X ? Limited

Generic Events X - -

CGI Scripting X - -

Servlets X - -

CPL X X -

SIP Inheritance URLs:

General references to any Internet service (“forward to email”) Recursive embedding

HTTP: Basic request/response format, status codes, authentication, … Proxies (but no caching) CGI programming interface; servlets

Email/SMTP: Addressing (user@domain) MX SRV records for load balancing and redundancy Header / body separation, MIME

SIP Design Choices

Transport protocol neutrality:Run over reliable (TCP, SCTP) and unreliable (UDP)

channels, with minimal assumptions

Request routing:Direct (performance) or proxy-routed (control)

Separation signaling vs. media description: Can add new applications or media types, SDP SDPng

Extensibility: Indicate and require proxy and UA capabilities

What is SIP?A Session Initiation Protocol

Ratified as RFC2543Being refined in RFC2543bis

A signaling protocolCall-control mechanism

Setup – modification – teardown

Resolves call endpointsDomain name to IP addresses

Describes the sessionTypically SDP (Session Description Protocol)

SDP’s Role in SIP

Session Description Protocol - RFC2327 Describes session information to potential session participants

Carried within the SIP message bodyDefines call attributes

Structured language to describe session characteristics Indicates transport protocol and parameters

Typically, RTP & payload formatEstablishes port numbers on which media should be sent

Typically, UDP ports 1024 to 65535Negotiates / exchanges available media capabilities

Audio, video, shared apps, chat,… including encoding methods

SIP AttributesLight & simple but flexible

Few transactions Scalable and extensible

Uses ‘Internet’ formats & components Text-based messages - HTTP/1.1 message syntax Internationalized: ISO 10646 char. set, UTF-8 encoding

Re-uses common ratified standards SDP; MIME; DNS; URL; HTTP authentication

Enables non-standard call set-up information ‘Useful’ information may be carried within payload Allows devices to make intelligent call-handling decisions Invokes various high-level services

URLs as identifiers Easy to re-direct to web resources (web push/pull)

Multicast ready For scaling and announcements (mostly future use)

Basic SIP Call Flow

sip:jim@zaffire.net (192.1.10.100) sip:jane@ubiquity.net (204.1.64.200)

UA UA

A B

SIP Proxy

Root DNS

Cache

Local

Secondary DNS

Registry

Primary DNS

media transportnetwork

SIP Proxy

Registry

qw

est.

net

192.

1.10

.1u

biq

uity.n

et204.1.64.200

(3)

(3)

(3)

(1)

(1) (1)

(1)

(4)

(4)

(5)

(6)

(2)

(7)(7)

(7)

(7)

“CALL JANE”

“CALL JANE”

SIP signalingnetwork

1. Register2. Initiate call request (sip:jane@ubiquity.net)3. DNS – resolve IP Address (ubiquity.net)4. Forward call request to remote proxy 5. Locate user in registry (jane)

6. Forward call request to end-user7. Accept call request8. Establish media connection

Standardization SIP and SIPPING working group are some of the most active

in IETF About 120 active internet drafts related to SIP Typically, 400 attend WG meetings at IETF 80-20% – 20% of the technical work takes 80% of the time!

Participation in SIP Bake-Offs (SIPit)

From RFC Release to Present Day

0

10

20

30

40

50

60

Org

aniz

atio

ns

Par

tici

pat

ing

04-99 08-99 12-99 04-00 08-00 12-00 08-01Source: SiPiT

5757

45

36

26

1516

0

Date

Technology Adoption

Columbia CS Phone System

SIP Proxy UnifiedMessaging

Server

UnifiedMessaging

Server

RTSPMediaServer

RTSPMediaServer

ConferencingServer(MCU)

ConferencingServer(MCU)

CAS/PCM

SIP/RTP

Video ConferencingMobile PDA

Sun SolarisPC Linux/FreeBSD/NT

Proxy/Redirect Server

SIP Phone

Black Phone sipc

T1PSTNNortel

MeridianPBX

NortelMeridian

PBX

SIP Phone

SIP Phone

Data base

MySQL User Database

Data base

LDAP Server

sipum

RTSP

sipconf rtspd

SIP/RTP

H.323/RTP802.11bWireless

“Plug ‘n SIPPOTS

Converter

Cisco 2600

sipd

What Problems Does It Solve? Integration of telephony with other mediaTelephony becomes another element of the IP / Internet

mixLowers the barrier for application development --

making it easier to be innovative Minimal clients and feature programming H323 and IN were/are not easy

Industry-standard platforms, web servers and IP infrastructures enable new services Most of these platforms already exist in the network SIP helps tie them together

New signaling and services architecture that is widely adopted By service providers and vendors

Impact on Service Providers Shift of telephony value add to the edge Facilities-less network service provider separation of bit

transport and services AOL, Yahoo, MSN...

It destroys the centralized business model of telephony Reduces the time to create new value-add services Easier to add vertical-market applications (integration with IT

infrastructure) Application-creation by non-specialists, similar to web services More personalized service model where the user has a greater

level of control

Market Dynamics

Converged PBX (CBX) Packet-based PBX; 4.1% of worldwide PBX sales in 2000; 19% in 2004

PC CBXSmall system for small business (CPE/CLE)

IP CBXLarger systems (carrier network based)

VoIP PBX/CBX Trends

Network-Based Applications Services

High-Level Sip OpportunitiesPresence managementPersonal & session mobilityUser profilingWeb call centersDesktop call managementVoice-enabled e-commerceMobile (3GPP) adoptionLocation servicesUnified messagingInstant messaging

Mobility, Presence & Profiles

SIP SignalingNetwork

ApplicationServices Broker

User profile Database

Voicemail Server

ASB

Data Base

VM Server

Services associated with a user not a device User may have multiple associations

Presence management for single ‘number’ reachability Selective call forwarding based on profile

E.g., unknown caller transferred to voicemail

BOSS

MOM

Long Distance Slammer

REGISTER

REGISTER

PROFILE

CALL OK

CALL OKUNKNOWN – SEND TO VM

Voice-Enabled Help DeskCall Center Application

Voice-Enabled e-Commerce

• Customer clicks-to-dial from a web page – pertinent details popped

• Customer browses website then navigates through an IVR

• Representative shares media (web push) with customer (e.g., technical documentation)

• Video conferencing initiated – negotiation, “show me”

Name: Bert BlogsOccup: MarketingModel: DishwasherPurchased: 11/23/96Last Contact: 1/9/99Last Service: 9/3/98

• Customer is connected to the appropriate representative

SIP SignalingNetwork

Application Services Broker

Integrated Voice Response Server Web Server

VoiceXML VoiceXML

ASB

IVR Server

3rd Generation Partnership Project

Calling Party

Application services broker – services and applications

environments

Home Network # 1

P- CSCF

Data base

S- CSCF

ASB

Service Control

Authorize QoSResources

Resource Reservation

CSCF = Call State Control Function – All SIP-based signalling platformsP = Proxy – 1st. point-of-contact. emergency service break-out and triggers local services (e.g., directory, QoS reservations) S = Serving – Determines what operator a subscriber belongs too. Provides subscriber services (call forward, VPN, etc.)I= Interrogating – Well-known entry point to different operator – ;oad Balancer for HSSHSS = Home Subscriber Server = Current location information (superset of GSM HLR (Home Location Register))

Called Party

Home Network # 2

P- CSCF

S- CSCF

I- CSCF

HSSWell-Known Entry Point

3GPP Release 5 - sample call between different service

providers

Server

Diameter

Serving GPRS Support Node (SGSN)

Gateway GPRS Support Node (GGSN)

GPRS = General Packet Radio Service

Radio Access Network

(3)

Location-Based Mobile Services

1. Taxi service requests user location from HSS

2. Location information used to retrieve list of cab companies in the area

3. User selects taxi service – call established to cab company

4. Cab company simultaneously updated with general location – closest cab cispatched

Dial-a-Cab

Dial-a-Cab

Application Services Broker

Home Subscriber Service

Web Server

ASB

HSS

SIP SignalingNetwork

Cab 1Cab 2

Cab 1 Cab 2 (1)

(1)

(2)

(3)

(2)

(4)Server

Impact to QwestDoes Qwest need to invest in disruptive technology?

Have the CLEC threats diminished? Will box/software providers playing in the edge be able to

sell CLASS features?Should Qwest fall back on traditional revenue

streams? New services Adding value to popular services Reducing costs

Should Qwest embrace or slow down technology adoption process? Big enough to through a large spanner in the works

Is SIP an opportunity or a threat for / to Qwest?

Service Provider Initiatives

Level 3WorldComAT&TBritish TelecomTeliaMicrosoft

Level 3Very active in the SIP arena

Integral part of their softswitch strategyActive in standards bodies and working groupsAnnounced industries first SIP-based IP voice networkInteroperability certification program

(3)Works voice certification program Designed stateless core proxy in-house Working closely with companies like Ubiquity on edge

strategyAggressive plans to expand capabilities and offeringsShunning traditional telephony applications

Less vertically integrated than WorldCom, for exampleNot attempting to reinvent the PSTN

WorldcomVery active in the SIP arenaEmploys major SIP advocate and promoter

Henry Sinnreich - Distinguished Member of EngineeringDesigned proxy in-house

Opened up to public for interoperability testinghttp://sipaccount.wcom.com/sipregistration.html

Recently announced a fully SIP-based Service“IP Communications” serviceRetail offering of hosted business communications applicationsIP Centrex (PBX replacement) Plus ……. Targets midsize to large customer base

Using a broker architecture to layer servicesDesigned in-house or from 3rd party vendors

Plans to offer SIP phonesCan be seen as a major play to undermine Class 5 services

AT&T

Taking the usual “early majority” stanceEmbracing SIP for future VoIP support

Currently using H.323 until SIP is broadly acceptedExpected to fully adopt SIP and replace H.323 in 12 to 18 months

Focusing on enterprise VPNs and managed servicesManaged Internet Service (MIS) – IPManaged Router Service (MRS) – Frame

British Telecom

Publicly Evaluating Ubiquity Products for Advanced ServicesWorking with the Ubiquity product portfolio to

create advanced, new, servicesFocus on both residential & business verticalsInitial services to roll-out shortly

TeliaEarly adopter of SIP-based applications

and services‘Second-line’ residential services targeted at

teenagersPresence; call profile; web push; IM

Focus on specific vertical markets Market-specific applicationsNetwork-based / hostedCall profiles; presence; IM

Employing an applications service broker architecture

MicrosoftMaking a huge play for ubiquitous support of SIP

at all levelsUnder the “.NET” architecture umbrellaWindows XP (GA)

SIP-enabled version of messengerSIP user agent / client

Windows XP Server (July 2002)Extensible SIP proxy server

Windows CE (July 2002)SIP user agent / client

Windows Embedded - OS for Appliances (July 2002)SIP user agent / client

SIP phone (Q1 2002)“Stinger”

Xbox gaming platform (Nov. 2001)“Hoot ‘n holler” – voice with networked games

Other Carriers Active in SIP

Verizon (US)Genuity (US)Broadwing (US)Telecom Italia (Italy)FranceTelecom (France)Deutsche Telekom (Germany)KPN Telecom (Netherlands)Elisa (Helsinki Telephone - Sweden)Telenor (Norway)Orange (UK Mobile)

Primary focus is advanced applications and services - not pure backbone infrastructure - US carriers Typically want to augment NB IP VPN services

Ubiquity Market PresenceExtend leadership position as provider of carrier grade,

end-to-end, SIP infrastructure solutionsDevelop joint solution platforms with partners that they can

sell to their customers: Ubiquity + Carrier Enterprise Ubiquity + NEV Carrier Ubiquity + NEV Enterprise

Create ‘pull’ demand in the carrier space for NEV / infrastructure solutions

Eventually create ‘pull’ demand directly from enterprisesPartner with best of breed application providers (e.g., media

servers) to enable advanced bundled solutions on top of the Ubiquity platform

Offer telco-class applications designed in-house

Product PortfolioProxy Server

SIP Network Server

Applications Services Broker

Design Deck

Element Manager

SIP Proxy

Net Server

ASB

NMS

DD

Signaling Network EvolutionS

lower

Statefu

ll

Fas

tS

tate

less

ServiceAware

Non-ServiceAware

SIP Proxy

Net Server

ASB

Edge Provisioning Optimized for service

delivery

Core RoutingOptimized for speed

SIP Network Server

Load Balancing ManagerLoad Balancing Manager

TransactionStatefulProxy

SIP EngineRegistrarModule

Location

Service

Mod

ule

Au

then

tication

Mod

ule

SIP

SIP RADIUSDNSENUM

RoutingModule

RoutingModule

DatabaseInterfaceModule

DatabaseInterfaceModule

Management Server

SNMPMIB

Event &Config Log

Database

SNMP

JDBC

RedirectServer

Ubiquity in the Converged Network

TransmissionSwitching

Call Control(Signaling)

ApplicationsASB “Gear”

Network Services O/S

Services

The ASB Drives service creation by

mediating and smoothly integrating the applications and

signaling layers.

Thus, the ASB aids in the deployment of new,

disparate, multi-vendor services and

easies feature interaction issues

Application Service Broker (ASB)

Service DirectorService Director

Service Host

TransactionStateful Proxy

RegistrarModule

Med

ia P

ush

/Pu

ll

UserAgent

Module

3rd P

arty Call

Con

trol

CP

L

En

gine

Presen

ce

Location

Service

Mod

ule

Au

then

tication

Mod

ule

SIP

RADIUSDNSENUM

RoutingModule

RoutingModule

DatabaseInterfaceModule

DatabaseInterfaceModule

Management Server

SNMPMIB

Event &Config Log

External Database

SNMP

JDBC

HTTP

SOAP

SIP

Server

External Resources

Service

Aggregation

Service

Con

figuration

Service

Policy

Service

Su

bscrip

tionsS

ER

VIC

EL

OG

ICS

ER

VIC

EE

NG

INE

S

Distributed Service Architecture

ASB

WEB Server

Data base

Net ServerNet

ServerNet Server

SIP Endpoint ‘A’ Enhanced

Services

Source-Routing

SIP SignalingNetwork

IP Transport Network

SIP Endpoint ‘B’ No Services

SIP Proxy

Transport

NETWORK EDGE NETWORK CORE

ServicesNetwork

Signaling

Applications

Media Stream (i.e. RTP/IP)

HTTP

SDP/SIP

Enhanced, Brokered, Data ServicesApplication

Services Broker

Altavista’s Babelfish

Translation Server

Short Message

Service (SMS) Gateway

1. Send an instant message2. Forward message to a translation server3. Translated message forwarded to SMS gateway4. Message delivered to mobile phone

SOAPAltavista Babelfish

Helloeverybody!

Bonjour, tout le monde!

FIXED USER

MOBILEUSER

SIP SignalingNetwork

Mobile Network

ASB

SMS Gateway

MBSMESSAGE sip:simon@ubiquity.net;translate=en_frSIP/2.0

URI: “urn:xmethodsBabelFish”call.setMethodName: "BabelFish"translationmode: "en_fr"Sourcedata: "Hello everybody!"

SMS

Bonjour, tout le monde!

Hello, everybody!

English to French

Design DeckA set of APIs that when ported into any IDE allow a

Service Designer to create applications that can access Resources on the Application Services Broker (ASB) JavaBeans to Interface with ASB Modules License to Develop and Upload CPL Scripts onto the ASB JavaDocs Detailing the APIs Extensive Documentation and Sample Code

Service modules in the ASB are building blocks whose functionality is accessed via the DesignDeck API

Enables IP telephony call-control elements to be manipulated in combinations with user agents and web servers

Includes the follows Java Beans and Associated Java Docs Presence management; Instant messaging; Third-party call

control; CPL storage; Forwarding; Call logging

Sample Design Deck Application

call when available service

CALL PROFILE

WEB Server

Data base

ASB

LDAP JDBC

“Automatically Establish Call When SIP Endpoint ‘B’ Becomes Available”

Data base

1. Set Call Profile Via Web Interface using Java Server Pages (JSP)

SIP Endpoint ‘A’

SIP Endpoint ‘B’

2. Endpoint ‘B’ Notifies Availability via SIP REGISTER – Presence Status Updated

3. PM Element Detects Change-of-Sate and Triggers 3PCC Element, INVITEing the

Two Third Parties

DD

Use Design Deck to Generate Java Code

With Beans

SIP SignalingNetwork

Java Server Pages

Register

Register

HT

TP

HTTP

IP Transport Network

Media

SERVICE

MODULES

Execute Service Logic

Update Presence (NOTIFY)

Call Logic (Subscribe)

Call Set-Up Message (UDP)

Invite

Invite(s)

Invite

“B” OFFLINE

“B”ONLINE

Going Forward

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Question & Answer Session

OPEN FORUM