Dredgie 3rd Review Draft Qwest Presentation

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Transcript Dredgie 3rd Review Draft Qwest Presentation

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