1 / 25

Session Initiation Protocol (SIP) Awareness

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

aimee-cobb
Download Presentation

Session Initiation Protocol (SIP) Awareness

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  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.charters/sip-charter.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? ITU-T created H.323 as a “Packet-based multimedia communications systems” The ITU-T started work on defining VoIP signaling protocols in May 1995. 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.charters/sipping-charter.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. 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 SIP Services Examples a.k.a. SIPPING-19

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

  11. Components of SIP • User Agent • 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. Requests (Methods) REGISTER 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 SIP Messages

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

  14. Avaya and SIP • 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 PropositionMoving 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 CommunicationsEvolutionary 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 SEAMLESS 3rd Party SIP Servers & Applications Communications CM Features IP SoftphoneSIP/SIMPLE Presence & IM IP, Wireless, Digitaland 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. 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 SIP Telephony FeaturesAvailable to Any SIP Telephone

  21. SIP Trunking to SIP VoIP Service ProvidersClear 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 DeliverBusiness 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. 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.

  24. Some current articles • Advanced SIP interoperability is slow in the making • http://www.networkworld.com/research/2005/050205-ilabs-sip.html • Network Computing Well-Connected nomination for best in category • http://www.networkcomputing.com/showitem.jhtml?docid=1608wcasec8 • SIP PBXes stake a claim • http://www.infoworld.com/article/05/05/02/18TCpbx_2.html

  25. Official Convergence Communication Providerfor the 2002 and 2006 FIFA World Cup™FIFA Women’s World Cup USA 2003

More Related