Dredgie 3rd Review Draft Qwest Presentation
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Ubiquity Software Corporation
~ SIP ~
Simple Protocol - Profound Implications
Working Agenda
Introduction to Ubiquity Software Corporation
An overview of the Session Initiation Protocol (SIP)
SIP in the marketplace
Implications for Qwest
Worldwide service provider SIP initiatives
How can Ubiquity help?
Going forward
Question & 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
NEED THIS INFO – SALES?
Henning Schulzrinne
Associate Professor, Columbia University
Department of Computer Science and Electrical
Engineering
Ubiquity Software, Corp. Board Member
Since March 2001
Acknowledged as the architect of SIP
Co-Authored RFC2543 with aid of student and
colleagues
Other related experience Includes:
Internet telephony; Internet multimedia; quality-ofservice; mobility; security
Other co-authored RFC’s Include:
RTSP & RTP
A Brief History Of SIP
Feb. 1996: earliest Internet drafts
Feb. 1999: Proposed Standard
March 1999: RFC 2543
April 1999: first SIP bake-off
November 2000: SIP accepted as 3GPP signaling
protocol
December 2001: 6th bake-off, 200+ participants
March 2001: 7th bake-off, first time outside U.S.
VoIP Signaling Architectures
MGCP, Megaco = master / slave
H.323 = (Mostly) single administrative domain
SIP = Peer-to-peer, cross domain
VoIP Architectures
Feature
SIP
H.323
Megaco/MGCP
Multiple Domains
Third-Party Control
Multimedia
X
X
X
?
Fixed Set
Single-domain
Unlikely
End System Control
Extensible
Generic Events
X
X
X
X
?
-
Limited
-
CGI Scripting
Servlets
CPL
X
X
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 RFC2543
Being refined in RFC2543bis
A signaling protocol
Call-control mechanism
Setup – modification – teardown
Resolves call endpoints
Domain name to IP addresses
Describes the session
Typically 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 body
Defines call attributes
Structured language to describe session characteristics
Indicates transport protocol and parameters
Typically, RTP & payload format
Establishes port numbers on which media should be sent
Typically, UDP ports 1024 to 65535
Negotiates / exchanges available media capabilities
Audio, video, shared apps, chat,… including encoding methods
SIP Attributes
Light & 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)
Secondary
DNS
Primary
DNS
Basic SIP Call Flow
1. Register
2. Initiate call request (sip:[email protected])
3. DNS – resolve IP Address (ubiquity.net)
4. Forward call request to remote proxy
(3)
Root DNS
(4)
(3)
SIP signaling
network
(7)
5. Locate user in registry (jane)
6. Forward call request to end-user
7. Accept call request
8. Establish media connection
(7)
(4)
(3)
SIP
Proxy
SIP
Proxy
Cache
(5)
(1)
Registry
Registry
(1)
ubiquity.net
204.1.64.200
qwest.net
192.1.10.1
Local
(1)
(1)
(2)
(7)
A
UA
sip:[email protected] (192.1.10.100)
(7)
“CALL
JANE”
(6)
media
transport
network
B
UA
sip:[email protected] (204.1.64.200)
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!
57
Participation in SIP
Bake-Offs (SIPit)
From RFC Release
to Present Day
Organizations Participating
60
57
45
50
36
40
26
30
16
15
04-99
08-99
20
10
0
0
12-99
04-00
08-00
Date
12-00
08-01
Source: SiPiT
Technology Adoption
Columbia CS Phone System
MySQL User
Database
SIP
Phone
Data
base
sipconf
LDAP
Server
Conferencing
Server
(MCU)
Data
base
rtspd
RTSP
Media
Server
RTSP
SIP
Phone
SIP Proxy
CAS/PCM
PSTN
Nortel
Meridian
PBX
sipd
T1
SIP/RTP
Cisco 2600
POTS
“Plug ‘n SIP
sipc
sipum
Proxy/Redirect Server
Sun Solaris
PC Linux/FreeBSD/NT
Black
Phone
Unified
Messaging
Server
SIP/RTP
802.11b
Wireless
Mobile PDA
SIP
Phone
H.323/RTP
Converter
Video
Conferencing
What Problems Does It Solve?
Integration of telephony with other media
Telephony becomes another element of the IP / Internet
mix
Lowers 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
VoIP PBX/CBX Trends
Converged PBX (CBX)
Packet-based PBX; 4.1%
of worldwide PBX sales in
2000; 19% in 2004
PC CBX
Small system for small
business (CPE/CLE)
IP CBX
Larger systems (carrier
network based)
Network-Based Applications Services
High-Level Sip Opportunities
Presence management
Personal & session mobility
User profiling
Web call centers
Desktop call management
Voice-enabled e-commerce
Mobile (3GPP) adoption
Location services
Unified messaging
Instant messaging
Mobility, Presence & Profiles
User profile
Database
Application
Services Broker
Data
Base
Voicemail
Server
VM
Server
Long Distance
Slammer
ASB
SIP Signaling
Network
MOM
BOSS
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
Voice-Enabled Help Desk
Name: Bert Blogs
Occup: Marketing
Model: Dishwasher
Purchased: 11/23/96
Last Contact: 1/9/99
Last Service: 9/3/98
Call Center Application
Voice-Enabled e-Commerce
Integrated Voice
Response Server
Application
Services Broker
VoiceXML
Web Server
VoiceXML
Server
IVR
ASB
SIP Signaling
Network
•
Customer clicks-to-dial from a web page – pertinent details popped
•
Customer browses website then navigates through an IVR
•
Customer is connected to the appropriate representative
•
Representative shares media (web push) with customer (e.g., technical documentation)
•
Video conferencing initiated – negotiation, “show me”
3rd Generation Partnership Project
Application services broker
– services and applications
environments
Data
base
Authorize QoS
Resources
ASB
Server
3GPP Release 5 - sample call
between different service
providers
Service
Control
PCSCF
SCSCF
SCSCF
Home Network # 1
Calling
Party Resource
Reservation
Radio Access Network
ICSCF
Well-Known
Entry Point
HSS
PCSCF
Home Network # 2
Diameter
Gateway GPRS Support Node (GGSN)
Called
Party
Serving GPRS Support Node (SGSN)
GPRS = General Packet Radio Service
CSCF = Call State Control Function – All SIP-based signalling platforms
P = 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 HSS
HSS = Home Subscriber Server = Current location information (superset of GSM HLR (Home Location Register))
Location-Based Mobile Services
Home Subscriber
Service
(1)
Application
Services Broker
Server
HSS
Cab 1
Dial-a-Cab
Cab 2
Dial-aCab 1
Cab
Cab2
Web Server
(2)
(4)
ASB
(1)
(2)
(3)
SIP Signaling
Network
(3)
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
Impact to Qwest
Does 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 3
WorldCom
AT&T
British Telecom
Telia
Microsoft
Level 3
Very active in the SIP arena
Integral part of their softswitch strategy
Active in standards bodies and working groups
Announced industries first SIP-based IP voice network
Interoperability certification program
(3)Works voice certification program
Designed stateless core proxy in-house
Working closely with companies like Ubiquity on edge
strategy
Aggressive plans to expand capabilities and offerings
Shunning traditional telephony applications
Less vertically integrated than WorldCom, for example
Not attempting to reinvent the PSTN
Worldcom
Very active in the SIP arena
Employs major SIP advocate and promoter
Henry Sinnreich - Distinguished Member of Engineering
Designed proxy in-house
Opened up to public for interoperability testing
http://sipaccount.wcom.com/sipregistration.html
Recently announced a fully SIP-based Service
“IP Communications” service
Retail offering of hosted business communications applications
IP Centrex (PBX replacement) Plus …….
Targets midsize to large customer base
Using a broker architecture to layer services
Designed in-house or from 3rd party vendors
Plans to offer SIP phones
Can be seen as a major play to undermine Class 5 services
AT&T
Taking the usual “early majority” stance
Embracing SIP for future VoIP support
Currently using H.323 until SIP is broadly accepted
Expected to fully adopt SIP and replace H.323 in 12 to 18 months
Focusing on enterprise VPNs and managed
services
Managed Internet Service (MIS) – IP
Managed Router Service (MRS) – Frame
British Telecom
Publicly Evaluating Ubiquity Products for
Advanced Services
Working with the Ubiquity product portfolio to
create advanced, new, services
Focus on both residential & business verticals
Initial services to roll-out shortly
Telia
Early adopter of SIP-based applications and
services
‘Second-line’ residential services targeted at
teenagers
Presence; call profile; web push; IM
Focus on specific vertical markets
Market-specific applications
Network-based / hosted
Call profiles; presence; IM
Employing an applications service broker
architecture
Microsoft
Making a huge play for ubiquitous support of SIP
at all levels
Under the “.NET” architecture umbrella
Windows XP (GA)
SIP-enabled version of messenger
SIP 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
Primary focus is advanced applications and services not pure backbone infrastructure - US carriers
Typically want to augment NB IP VPN services
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)
Ubiquity Market Presence
Extend leadership position as provider of carrier grade,
end-to-end, SIP infrastructure solutions
Develop 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 enterprises
Partner 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 Portfolio
Proxy Server
SIP Network Server
Applications Services Broker
Design Deck
Element Manager
SIP
Proxy
Net
Server
ASB
DD
NMS
Signaling Network Evolution
Edge Provisioning
Optimized for service
delivery
Service
Aware
ASB
Optimized for speed
Fast
Stateless
Net
Server
SIP
Proxy
Non-Service
Aware
Slower
Statefull
Core Routing
SIP Network Server
SIP
Load Balancing Manager
SIP Engine
SIP
Authentication
Module
Redirect
Server
Location Service
Module
Transaction
Stateful
Proxy
Registrar
Module
ENUM DNS RADIUS
Routing
Module
Database
Interface
Module
JDBC
Management Server
SNMP
MIB
Event &
Config Log
SNMP
Database
Ubiquity in the Converged Network
Applications
Services
Call Control
(Signaling)
Switching
ASB “Gear”
Network
Services O/S
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, multivendor services and
easies feature
interaction issues
Transmission
Application Service Broker (ASB)
External
Resources
SIP
HTTP
Routing
Module
Service Director
SOAP
Server
SIP
SERVICE
LOGIC
Service
Policy
Service
Subscriptions
3rd Party
Call Control
CPL
Engine
Presence
Authentication
Module
Service
Configuration
Media
Push/Pull
User
Agent
Module
Registrar
Module
Location Service
Module
Service
Aggregation
Transaction
Stateful Proxy
SERVICE
ENGINES
Service Host
ENUM DNS RADIUS
Database
Interface
Module
JDBC
Management Server
SNMP
MIB
Event &
Config Log
SNMP
External
Database
Distributed Service Architecture
Applications
HTTP
WEB
Server
Data
base
ASB
Services
Network
Signaling
Net
Net
Server
Net
Server
Server
SIP Endpoint
‘A’ Enhanced
Services
SDP/SIP
SIP
Proxy
SIP Signaling
Network
SourceRouting
Transport
SIP Endpoint
‘B’ No Services
Media Stream (i.e. RTP/IP)
NETWORK EDGE
IP Transport
Network
NETWORK CORE
Enhanced, Brokered, Data Services
Altavista’s
Babelfish
Translation
Server
Application
Services
Broker
Altavista
Babelfish
Hello,
everybody!
SOAP
SMS
URI: “urn:xmethodsBabelFish”
call.setMethodName: "BabelFish"
translationmode: "en_fr"
Sourcedata: "Hello everybody!"
Short
Message
Service (SMS)
Gateway
SMS
Gateway
ASB
Hello
everybody!
SIP Signaling
Network
FIXED USER
Mobile
Network
MBS
MESSAGE sip:[email protected];translate=en_frSIP/2.0
Bonjour, tout
le monde!
English to French
1. Send an instant message
2. Forward message to a translation server
3. Translated message forwarded to SMS gateway
4. Message delivered to mobile phone
Bonjour, tout
le monde!
MOBILE
USER
Design Deck
A 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
DD
Use Design Deck to
Generate Java Code
With Beans
Java Server
Pages
Data
base
Data
base
LDAP
WEB
Server
Call Logic (Subscribe)
Update Presence (NOTIFY)
Call Set-Up Message (UDP)
3. PM Element Detects Change-of-Sate and
Triggers 3PCC Element, INVITEing the
Two Third Parties
IP Transport
Network
“B”
ONLINE
OFFLINE
S
E
R
V
I
C
E
M
O
D
U
L
E
S
SIP Signaling
Network
Invite
“Automatically Establish Call When
SIP Endpoint ‘B’ Becomes Available”
call when
available
service
Invite(s)
HTTP
2. Endpoint ‘B’ Notifies Availability via SIP
REGISTER – Presence Status Updated
ASB
Register
1. Set Call Profile Via Web Interface using
Java Server Pages (JSP)
JDBC
Execute
Service Logic
CALL
PROFILE
SIP Endpoint
‘A’
Media
SIP Endpoint
‘B’
Going Forward
NEED THIS INFO – SALES?
Question & Answer Session
OPEN FORUM