1. Session Initiation Protocol (SIP) Awareness
May 11, 2005
Rod Averill
Global Solutions Manager, Avaya Federal Solutions

2. State of the Industry Two technology shifts in progress
TDM → IP
H.323 → SIP
Most companies will select their IP Communications vendor by end of 2005
Interest in SIP is steadily increasing
Currently viewed as “nice-to-have”
Forecasting 50% of IP RTUs are SIP by 2008
We’re here with SIP

3. History of SIP 1995: Work Begins Feb 1999: Published as RFC 2543
April 2000: SIP/SIMPLE selected by 3gpp
Adoption was initially very slow
H.323 vs. SIP debate
Accelerated with the support of Cisco, Microsoft, Nokia, etc.
Summer 2001: MS announces SIP as core of Windows XP
Spring 2005: MS announces LCS
Today
3 major IETF SIP working groups
40+ SIP RFCs
100+ SIP-related Internet Drafts
SIP products from nearly every major telecom vendor

4. What is Session Initiation Protocol (SIP). http://www. ietf. org/html
“SIP is an IETF application layer-protocol that can establish, modify, and terminate multimedia sessions”
RFC 3261
Media agnostic
Voice, video, instant messaging, etc.
Media negotiation
Offer-Answer model
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.

5. What is H.323? H.323 encodes messages in a compact binary format.
ITU-T created H.323 as a “Packet-based multimedia communications systems” The ITU-T started work on defining VoIP signaling protocols in May In December 1996, Study Group 16 passed the H.323 v.1
H.323 encodes messages in a compact binary format.
Allows peer to peer media
Media agnostic
Support for audio is mandatory, while data and video are optional.
Bulky Protocol
Currently more feature rich than SIP, but also more time consuming to implement.
Reuses other protocols
RTP, RTCP, H.225, H.245, H.450, H.460, T.120, T.38 fax, etc.

6. H.323 versus SIP
Both define a general framework for call control features. The framework defines a standardization process and rules for the implementation of new features.
Features:
H.323 defines support for many more features than SIP, but SIP is slowly catching up.
Avaya’s H.323 supports almost all of the TDM features (700+) but does so in a proprietary fashion. Similarly, CISCO uses proprietary “skinny” protocol for H.323 feature support. Third party H.323 phones must do considerable extra work to access Avaya’s H.323 feature set.
Avaya’s SIP support includes all SIPPING features plus many others through extending our Off Premise Station (OPS) feature. Any third party SIP phone can access these features without additional software development.

7. What is SIMPLE? http://www.ietf.org/html.charters/sipping-charter.html
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
Selected as basis for 3gpp networks & applications

8. What is SIPPING-16, SIPPING-19, etc. http://www. ietf. org/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)

9. 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

10. The SIP Address of Record (AOR)
SIP provides a logical identity, the “public address”, for users
e.g.
Mapped to any number of arbitrary devices
Independent of physical location
Hoteling and User Mobility is native to SIP
Hand held IP Device
Desktop Application (e.g. softphone)
SIP Phone
Desk phone
Instant Messaging Client
Mobile Phone

11. Components of SIP User Agent Registrar Location Service Proxy Server
User Agent Client
Generates and sends SIP requests and receives responses
User Agent Server
Receives SIP requests and generates SIP responses
Registrar
Provides mapping of logical SIP addresses to physical SIP addresses
Location Service
Used by SIP Proxy or Redirect server to obtain the mapping from logical SIP addresses to physical SIP addresses
Proxy Server
Forwards SIP requests downstream and responses upstream
Redirect Server
Generates 3xx responses directing clients to contact an alternate set of URIs
Presence Server
Acts as a Presence Agent or proxy server for SUBSCRIBE requests

12. SIP Messages Requests (Methods) REGISTER INVITE, ACK, CANCEL BYE
Register contact information
INVITE, ACK, CANCEL
Setting up sessions
BYE
Terminating sessions
OPTIONS
Querying servers about their capabilities
SUBSCRIBE, NOTIFY (RFC 3265)
Event notification framework
MESSAGE (RFC 3428)
Instant messages
Responses
1xx: Provisional
request received, continuing to process the request
2xx: Success
the action was successfully received, understood, and accepted
3xx: Redirection
further action needs to be take in order to complete the request
4xx: Client Error
the request contains bad syntax or cannot be fulfilled at this server
5xx: Server Error
the server failed to fulfill an apparently valid request
6xx: Global Failure
the request cannot be fulfilled at any server

13. Example Call Flow Alice Proxy Bob INVITE sip:bob@example.com
407 Proxy Authentication Required
ACK
INVITE
INVITE
100 Trying
Bob
answers
180 Ringing
180 Ringing
200 Ok
200 Ok
ACK
ACK
Bob
hangs up
RTP
BYE
BYE
200 Ok
200 Ok

14. Avaya and SIP Avaya has a leadership role in IETF SIP working groups
Extensive SIP-based product line
Communication Manager 2.2
Converged Communications Server 2.1
IP Softphone R5.2
4602 / 4602SW SIP IP Phone
Avaya has a leadership role in IETF SIP working groups
Technical advisor to core SIP working group
Contributes to emerging SIP standards, including presence
Avaya participates in SIPit (bi-annual SIP interop convention)
Vendors tested with:
Alcatel, AT&T, AudioCodes, AudioTest, Broadcom, Cisco, Compaq, Digaco, dynamicsoft, HCL, Hughes, Indigo, Mediatrix, Mitel, NIST, Nokia, Nuera, Pingtel, Polycom, Radcom, Radvision, Siemens, SNOM, Sylantro, Telogy, TonesTest, Trillium, Vovida, Webley, Wipro, Worldcom

15. Avaya’s SIP Solution Value Proposition Moving to an open user-centric communications architecture
Investment Protection and Managed Evolution
Building on existing infrastructure, making old and new work together
New Converged Communications Server (CCS)
SIP proxy, registrar, location, and presence server
Software foundation for integrating Avaya & 3rd party, applications & endpoints
New 4602 / 4602SW SIP IP Telephone firmware
Same hardware, firmware upgradeable at no additional cost
Multi-Vendor Standards-Based Interoperability
Committed to supporting IETF standards for basic interoperability
Support for 3rd party vendor SIP endpoints:
Cisco 7940/7960
Polycom SoundPoint IP 600
More to come
Provides Business-Class Telephony Features to SIP Endpoints
Going beyond basic SIP: hold, transfer, 3-way conference, and Message Waiting Indication
Extending Communication Manager features through Avaya SIP infrastructure
Enabling migrations to SIP telephony without sacrificing your favorite features and functionality
Reduces OPEX via Lower Cost Trunking to Service Providers
New SIP trunking through CCS provides alternative to PRI, H.323
Available now or in near future from all major carriers in US and abroad
Increases Productivity and Enhances Collaboration
Introduces new Presence capabilities within IP Softphone 5.1
Secure Enterprise Instant Messaging with persistent logging

16. Avaya Converged Communications Evolutionary path, integrating old and new, Avaya, and 3rd party
3rd Party SIP Servers & Applications
PSTN /
Mobile
Service Provider
SIP Trunks
Converged
Communications
Server 2.1
Communication
Manager 2.1.1
“Feature Server”
sip:example.com
3rd Party SIP
Servers &
Applications
SEAMLESS
Communications
CM Features
IP Softphone SIP/SIMPLE
Presence & IM
IP, Wireless, Digital and Analog Endpoints
3rd Party SIP Endpoints
Avaya SIP Endpoints

17. Avaya Converged Communications Server
Provides standards-based SIP architecture for telephony, Instant Messaging, video, etc.
Utilizes Avaya Communication Manager as telephony feature server
Deployed on S8500 (IBM x305) hardware platform
Three flavors
Home CCS
SIP proxy, location, registrar, and presence server
Edge CCS
SIP proxy server
ingress/egress, inter-Home routing
Combo CCS
Home + Edge for single server deployments

18. Supported IETF Standards & Drafts
RFC 1889 RTP: Real-Time Transport Protocol
RFC 2246 The TLS Protocol
RFC 2327 SDP: Session Description Protocol
RFC 2396 URI generic syntax
RFC 2617 Digest Authentication
RFC 2782 A DNS RR for specifying the location of services (DNS SRV)
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 3261 SIP: Session Initiation Protocol
RFC 3262 Reliability of Provisional Responses in the Session Initiation Protocol (SIP)
RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers
RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP)
RFC 3265 SIP-Specific Event Notification: Message Summary and Message Waiting Indication Event Package
RFC 3420 Internet Media Type message/sipfrag
RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging
RFC 3515 REFER method
RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)
RFC 3856 A Presence Event Package for the Session Initiation Protocol (SIP)
RFC 3891 The Session Initiation Protocol (SIP) "Replaces" Header
draft-ietf-impp-cpim-pidf-05
draft-ietf-simple-winfo-package-03
draft-ietf-sip-session-timer-15
draft-ietf-sipping-cc-conferencing-04

19. Avaya Embraces Interoperability
Avaya Solution and Interoperability Test Lab
Phones Tested:
Cisco, Polycom, Pingtel, etc.
Avaya Services will support tested third-party phones that conform to our documented configuration
Servers Tested:
Cisco SIP Proxy Server w/ CallManager
SIP DevConnect Partner Program
Ingate
Kagoor

20. SIP Telephony Features Available to Any SIP Telephone
Active appearance select
Automatic call back
Automatic call back cancel
Call forwarding – Unconditional
Call forwarding – Busy
Call forwarding – No answer
Call forwarding deactivation
Call hold
Call park
Call park answer back
Call pick-up
Conference on answer
Calling party number block
Calling party number unblock
Consultation hold
Directed call pick-up
Drop Last Added Party
Exclusion (Toggle on/off)
Find-me
Held appearance select
Incoming call screening
Idle appearance select
Last number dialed
Malicious call trace activation
Malicious call trace deactivation
Manual signaling
Message waiting indication
Music on hold
Outgoing call screening
Priority call
Send all calls enable
Send all calls disable
3-way conference – 3rd party added
Transfer – Unattended
Transfer – Attended
Transfer on hang up
Transfer to voice mail

21. SIP Trunking to SIP VoIP Service Providers Clear ROI
Several Service Providers actively developing offers
Origination-only, termination-only, or both
Toll-reduction, toll-elimination
Signaling trunks between SP and enterprise SIP proxies
Reminiscent of H.323 signaling group within IP Trunks, ISDN or QSIG signaling within PRI
Easier to implement in SIP vs H.323
Avaya creating certification program for SIP VoIP SPs
Provides guaranteed quality and reduces customer implementation cycle
List certified providers posted to avaya.com

22. Presence-based Communications Deliver Business results
Better customer service, faster decisions
By increasing accessibility between co-workers
Enabling always-on communications
Speeding access to the right people for the right customer
Increased productivity
By increasing user control over preferred modes of contact and rules of communications
Supporting multi-modal user interfaces (e.g. IM and telephony) and easy transitions (e.g. click-to-talk)
Cost savings
Investment protection through evolutionary migration
Leveraging existing applications, systems and phones (IP and digital)
Enabling 3rd party endpoints with extended functionality
Lowering IT TCO with security and manageability

23. Example Collaborative Scenario
1. Customer calls Manufacturer’s Rep
2. Rep needs technical answer from expert
3. Rep uses presence to “peek over the cubicle” to see if expert available
5. Rep IM’s the expert - begins to get answers
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
Customer Scenario:
Customer calls Manufacturer’s Rep
Rep needs technical answer from product expert
Rep uses presence to “peek over the cubicle” to see if expert is available
The expert is available online, but is also on the phone
Rep IM’s the expert and begins to get answers
Rep notices that expert has ended her conversation
Expert agrees to join conference with customer
Expert answers questions in the conference call, as she continues to chat with the Rep in the background
With all of his questions satisfied, the customer wants to place an order
The expert drops out of the call to allow the Rep to complete the sale.

24. User-centric Communications Architecture Application Enablement with secure SIP-based services
Architecture focus shifts to user
Users not devices, enabling user mobility
User identity, presence, rules, routing
Unified Access
Enhanced softphones, phones, PDA apps
Increased endpoint intelligence
Integrate with speech, web, wireless (UCC)
Application integration
SIP is a key “glue” between business and communications applications
Along with XML and Web Services
Communication Services
Secure SIP-based integration layer to be leveraged across Avaya portfolio
Natural evolution from current products
Open standards-based APIs
Aggressive third-party support (DevConnect)
SIP - an enabling protocol
Leverages Internet-based standards
Simple – lowers cost, increases scalability
Enables presence and instant messaging
Promotes interoperability of applications, networks, vendors
Logical identity across communication modes
More intelligent endpoints and applications

25. Hospital Presence and Notify Scenario
7. Presence server status allows call routing to on-call physician and reserves resources on a Pre-surgery Video Conference.
1. System Administrator sets up notification rules in Notification Application
2. Physician enters his notification and presence preferences into self-service portal
8. On-call physician accesses test results and immediately is put into a video and IM conference with Available experts from around the nation.
3. Patient undergoes a series of medical tests
9. Notify App system maintains a record of the notification and response and conference. All encrypted
4. Lab tech enters test results into patient records management system
5a. Records management system sends event to Notification App
6. Based on the Physician’s calendar, he is
Not Available …a little too busy!
5b. Or, alternatively, App polls records management system for updates
Medical
Information
System
Avaya CCS

26. The Bottom Line
H.323 has better security story … SIP catching up quickly.
H.323 has better capacity and performance story …
H.323 is more fully featured… SIP making good progress.
SIP is better for distributed network architecture.
SIP is implemented in a more standards-compliant way … better for customers who truly want plug and play components.
AVAYA supports Both H.323 and SIP.
Just a firmware change in 4602, 4610, and 4620 phones (H.323 or SIP)
I predict that SIP will win the race because more vendors will provide products for the SIP protocol.

27. Some current articles
Advanced SIP interoperability is slow in the making
Network Computing Well-Connected nomination for best in category
SIP PBXes stake a claim

28. Official Convergence Communication Provider for the 2002 and 2006 FIFA World Cup™ FIFA Women’s World Cup USA 2003
Marvellous