Presentation on theme: "Session Initiation Protocol (SIP) Awareness"— Presentation transcript:
1Session Initiation Protocol (SIP) Awareness May 11, 2005Rod AverillGlobal Solutions Manager, Avaya Federal Solutions
2State of the Industry Two technology shifts in progress TDM → IPH.323 → SIPMost companies will select their IP Communications vendor by end of 2005Interest in SIP is steadily increasingCurrently viewed as “nice-to-have”Forecasting 50% of IP RTUs are SIP by 2008We’re here with SIP
3History of SIP 1995: Work Begins Feb 1999: Published as RFC 2543 April 2000: SIP/SIMPLE selected by 3gppAdoption was initially very slowH.323 vs. SIP debateAccelerated with the support of Cisco, Microsoft, Nokia, etc.Summer 2001: MS announces SIP as core of Windows XPSpring 2005: MS announces LCSToday3 major IETF SIP working groups40+ SIP RFCs100+ SIP-related Internet DraftsSIP products from nearly every major telecom vendor
4What 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 3261Media agnosticVoice, video, instant messaging, etc.Media negotiationOffer-Answer modelSimilar to HTTPRequest-Response modelText message-based protocolEasy to debugReuses other IETF protocolsUDP, TCP, TLS, DHCP, DNS, SDP, RTP, MIME, etc.
5What 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.1H.323 encodes messages in a compact binary format.Allows peer to peer mediaMedia agnosticSupport for audio is mandatory, while data and video are optional.Bulky ProtocolCurrently more feature rich than SIP, but also more time consuming to implement.Reuses other protocolsRTP, RTCP, H.225, H.245, H.450, H.460, T.120, T.38 fax, etc.
6H.323 versus SIPBoth 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.
7What is SIMPLE? http://www.ietf.org/html.charters/sipping-charter.html SIP for Instant Messaging and Presence Leveraging ExtensionsIETF working groupIntroduces “Presence” into communications stateBuilds on RFC 3265Now a standard: RFC 3856Selected as basis for 3gpp networks & applications
8What is SIPPING-16, SIPPING-19, etc. http://www. ietf. org/html Session Initiation Protocol Project INvestiGationIETF working groupChartered to document the use of SIP for several applications related to telephony and multimediaSIPPING-19 refers to SIP Services Examples draftdraft-ietf-sipping-service-examples-0719 example telephony features implemented in SIPPurpose is to ensure that basic features interoperateOther SIPPING itemsSIP Basic Call Flow Examples (RFC 3665)Message Waiting Indication (RFC 3842)
9SIP Services Examples a.k.a. SIPPING-19 Call HoldConsultation HoldMusic on HoldTransfer – UnattendedTransfer – AttendedTransfer – Instant MessagingCall Forwarding – UnconditionalCall Forwarding – BusyCall Forwarding – No Answer3-way Conference – 3rd Party Added3-way Conference – 3rd Party JoinsSingle Line ExtensionFind-MeIncoming Call ScreeningOutgoing Call ScreeningCall ParkCall PickupAutomatic RedialClick to DialMessage Waiting Indication
10The SIP Address of Record (AOR) SIP provides a logical identity, the “public address”, for userse.g.Mapped to any number of arbitrary devicesIndependent of physical locationHoteling and User Mobility is native to SIPHand held IP DeviceDesktop Application (e.g. softphone)SIP PhoneDesk phoneInstant Messaging ClientMobile Phone
11Components of SIP User Agent Registrar Location Service Proxy Server User Agent ClientGenerates and sends SIP requests and receives responsesUser Agent ServerReceives SIP requests and generates SIP responsesRegistrarProvides mapping of logical SIP addresses to physical SIP addressesLocation ServiceUsed by SIP Proxy or Redirect server to obtain the mapping from logical SIP addresses to physical SIP addressesProxy ServerForwards SIP requests downstream and responses upstreamRedirect ServerGenerates 3xx responses directing clients to contact an alternate set of URIsPresence ServerActs as a Presence Agent or proxy server for SUBSCRIBE requests
12SIP Messages Requests (Methods) REGISTER INVITE, ACK, CANCEL BYE Register contact informationINVITE, ACK, CANCELSetting up sessionsBYETerminating sessionsOPTIONSQuerying servers about their capabilitiesSUBSCRIBE, NOTIFY (RFC 3265)Event notification frameworkMESSAGE (RFC 3428)Instant messagesResponses1xx: Provisionalrequest received, continuing to process the request2xx: Successthe action was successfully received, understood, and accepted3xx: Redirectionfurther action needs to be take in order to complete the request4xx: Client Errorthe request contains bad syntax or cannot be fulfilled at this server5xx: Server Errorthe server failed to fulfill an apparently valid request6xx: Global Failurethe request cannot be fulfilled at any server
13Example Call Flow Alice Proxy Bob INVITE sip:firstname.lastname@example.org 407 Proxy Authentication RequiredACKINVITEINVITE100 TryingBobanswers180 Ringing180 Ringing200 Ok200 OkACKACKBobhangs upRTPBYEBYE200 Ok200 Ok
14Avaya and SIP Avaya has a leadership role in IETF SIP working groups Extensive SIP-based product lineCommunication Manager 2.2Converged Communications Server 2.1IP Softphone R5.24602 / 4602SW SIP IP PhoneAvaya has a leadership role in IETF SIP working groupsTechnical advisor to core SIP working groupContributes to emerging SIP standards, including presenceAvaya 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
15Avaya’s SIP Solution Value Proposition Moving to an open user-centric communications architecture Investment Protection and Managed EvolutionBuilding on existing infrastructure, making old and new work togetherNew Converged Communications Server (CCS)SIP proxy, registrar, location, and presence serverSoftware foundation for integrating Avaya & 3rd party, applications & endpointsNew 4602 / 4602SW SIP IP Telephone firmwareSame hardware, firmware upgradeable at no additional costMulti-Vendor Standards-Based InteroperabilityCommitted to supporting IETF standards for basic interoperabilitySupport for 3rd party vendor SIP endpoints:Cisco 7940/7960Polycom SoundPoint IP 600More to comeProvides Business-Class Telephony Features to SIP EndpointsGoing beyond basic SIP: hold, transfer, 3-way conference, and Message Waiting IndicationExtending Communication Manager features through Avaya SIP infrastructureEnabling migrations to SIP telephony without sacrificing your favorite features and functionalityReduces OPEX via Lower Cost Trunking to Service ProvidersNew SIP trunking through CCS provides alternative to PRI, H.323Available now or in near future from all major carriers in US and abroadIncreases Productivity and Enhances CollaborationIntroduces new Presence capabilities within IP Softphone 5.1Secure Enterprise Instant Messaging with persistent logging
16Avaya Converged Communications Evolutionary path, integrating old and new, Avaya, and 3rd party 3rd Party SIP Servers & ApplicationsPSTN /MobileService ProviderSIP TrunksConvergedCommunicationsServer 2.1CommunicationManager 2.1.1“Feature Server”sip:example.com3rd Party SIPServers &ApplicationsSEAMLESSCommunicationsCM FeaturesIP Softphone SIP/SIMPLEPresence & IMIP, Wireless, Digital and Analog Endpoints3rd Party SIP EndpointsAvaya SIP Endpoints
17Avaya Converged Communications Server Provides standards-based SIP architecture for telephony, Instant Messaging, video, etc.Utilizes Avaya Communication Manager as telephony feature serverDeployed on S8500 (IBM x305) hardware platformThree flavorsHome CCSSIP proxy, location, registrar, and presence serverEdge CCSSIP proxy serveringress/egress, inter-Home routingCombo CCSHome + Edge for single server deployments
18Supported IETF Standards & Drafts RFC 1889 RTP: Real-Time Transport ProtocolRFC 2246 The TLS ProtocolRFC 2327 SDP: Session Description ProtocolRFC 2396 URI generic syntaxRFC 2617 Digest AuthenticationRFC 2782 A DNS RR for specifying the location of services (DNS SRV)RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony SignalsRFC 3261 SIP: Session Initiation ProtocolRFC 3262 Reliability of Provisional Responses in the Session Initiation Protocol (SIP)RFC 3263 Session Initiation Protocol (SIP): Locating SIP ServersRFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP)RFC 3265 SIP-Specific Event Notification: Message Summary and Message Waiting Indication Event PackageRFC 3420 Internet Media Type message/sipfragRFC 3428 Session Initiation Protocol (SIP) Extension for Instant MessagingRFC 3515 REFER methodRFC 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" Headerdraft-ietf-impp-cpim-pidf-05draft-ietf-simple-winfo-package-03draft-ietf-sip-session-timer-15draft-ietf-sipping-cc-conferencing-04
19Avaya Embraces Interoperability Avaya Solution and Interoperability Test LabPhones Tested:Cisco, Polycom, Pingtel, etc.Avaya Services will support tested third-party phones that conform to our documented configurationServers Tested:Cisco SIP Proxy Server w/ CallManagerSIP DevConnect Partner ProgramIngateKagoor
20SIP Telephony Features Available to Any SIP Telephone Active appearance selectAutomatic call backAutomatic call back cancelCall forwarding – UnconditionalCall forwarding – BusyCall forwarding – No answerCall forwarding deactivationCall holdCall parkCall park answer backCall pick-upConference on answerCalling party number blockCalling party number unblockConsultation holdDirected call pick-upDrop Last Added PartyExclusion (Toggle on/off)Find-meHeld appearance selectIncoming call screeningIdle appearance selectLast number dialedMalicious call trace activationMalicious call trace deactivationManual signalingMessage waiting indicationMusic on holdOutgoing call screeningPriority callSend all calls enableSend all calls disable3-way conference – 3rd party addedTransfer – UnattendedTransfer – AttendedTransfer on hang upTransfer to voice mail
21SIP Trunking to SIP VoIP Service Providers Clear ROI Several Service Providers actively developing offersOrigination-only, termination-only, or bothToll-reduction, toll-eliminationSignaling trunks between SP and enterprise SIP proxiesReminiscent of H.323 signaling group within IP Trunks, ISDN or QSIG signaling within PRIEasier to implement in SIP vs H.323Avaya creating certification program for SIP VoIP SPsProvides guaranteed quality and reduces customer implementation cycleList certified providers posted to avaya.com
22Presence-based Communications Deliver Business results Better customer service, faster decisionsBy increasing accessibility between co-workersEnabling always-on communicationsSpeeding access to the right people for the right customerIncreased productivityBy increasing user control over preferred modes of contact and rules of communicationsSupporting multi-modal user interfaces (e.g. IM and telephony) and easy transitions (e.g. click-to-talk)Cost savingsInvestment protection through evolutionary migrationLeveraging existing applications, systems and phones (IP and digital)Enabling 3rd party endpoints with extended functionalityLowering IT TCO with security and manageability
23Example Collaborative Scenario 1. Customer calls Manufacturer’s Rep2. Rep needs technical answer from expert3. Rep uses presence to “peek over the cubicle” to see if expert available5. Rep IM’s the expert - begins to get answers6. Rep notices expert has ended conversationAnd has expert join conference with customer7. Expert answers questions and continues IM chats with the Rep in the backgroundIM4. Expert available online but busy on phoneCustomer Scenario:Customer calls Manufacturer’s RepRep needs technical answer from product expertRep uses presence to “peek over the cubicle” to see if expert is availableThe expert is available online, but is also on the phoneRep IM’s the expert and begins to get answersRep notices that expert has ended her conversationExpert agrees to join conference with customerExpert answers questions in the conference call, as she continues to chat with the Rep in the backgroundWith all of his questions satisfied, the customer wants to place an orderThe expert drops out of the call to allow the Rep to complete the sale.
24User-centric Communications Architecture Application Enablement with secure SIP-based services Architecture focus shifts to userUsers not devices, enabling user mobilityUser identity, presence, rules, routingUnified AccessEnhanced softphones, phones, PDA appsIncreased endpoint intelligenceIntegrate with speech, web, wireless (UCC)Application integrationSIP is a key “glue” between business and communications applicationsAlong with XML and Web ServicesCommunication ServicesSecure SIP-based integration layer to be leveraged across Avaya portfolioNatural evolution from current productsOpen standards-based APIsAggressive third-party support (DevConnect)SIP - an enabling protocolLeverages Internet-based standardsSimple – lowers cost, increases scalabilityEnables presence and instant messagingPromotes interoperability of applications, networks, vendorsLogical identity across communication modesMore intelligent endpoints and applications
25Hospital 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 Application2. Physician enters his notification and presence preferences into self-service portal8. 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 tests9. Notify App system maintains a record of the notification and response and conference. All encrypted4. Lab tech enters test results into patient records management system5a. Records management system sends event to Notification App6. Based on the Physician’s calendar, he isNot Available …a little too busy!5b. Or, alternatively, App polls records management system for updatesMedicalInformationSystemAvaya CCS
26The Bottom LineH.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.
27Some current articlesAdvanced SIP interoperability is slow in the makingNetwork Computing Well-Connected nomination for best in categorySIP PBXes stake a claim
28Official Convergence Communication Provider for the 2002 and 2006 FIFA World Cup™ FIFA Women’s World Cup USA 2003Marvellous