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- Deck Set 2 0 5 – Simple Presentation Creator Download
- Deck Set 2 0 5 – Simple Presentation Creator Download Free
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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: Ä Michael Doyle – CTO Ä Professor Henning Schulzrinne - Columbia University (Board Member) Ä Martin De Prycker – CTO, Alcatel (Board Member) Raised US$42 million in venture capital - August 2000 Ä Ä Cap. Vest Equity Partners Fund, L. P; Celtic House International; JK&B Capital; Alcatel Recognized authorities in signaling and programming languages Ä SIP; JAVA Active in many associated standards bodies and working groups Ä 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 ÄDepartment of Computer Science and Electrical Engineering Ubiquity Software, Corp. Board Member ÄSince March 2001 Acknowledged as the architect of SIP ÄCo-Authored RFC 2543 with aid of student and colleagues Other related experience Includes: ÄInternet telephony; Internet multimedia; quality-ofservice; mobility; security Other co-authored RFC's Include: ÄRTSP & RTP
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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 3 GPP signaling protocol December 2001: 6 th bake-off, 200+ participants March 2001: 7 th bake-off, first time outside U. S.
Vo. IP Signaling Architectures MGCP, Megaco = master / slave H. 323 = (Mostly) single administrative domain SIP = Peer-to-peer, cross domain
Vo. IP Architectures Feature SIP H. 323 Megaco/MGCP Multiple Domains Third-Party Control X X ? - Single-domain Multimedia End System Control Extensible Generic Events CGI Scripting Servlets CPL X X X X Fixed Set X ? X Unlikely Limited -
SIP Inheritance URLs: Ä General references to any Internet service ('forward to email') Ä Recursive embedding HTTP: Ä Basic request/response format, status codes, authentication, … Ä Proxies (but no caching) Ä CGI programming interface; servlets Email/SMTP: Ä Addressing ([email protected]) Ä MX SRV records for load balancing and redundancy Ä Header / body separation, MIME
SIP Design Choices Transport protocol neutrality: ÄRun over reliable (TCP, SCTP) and unreliable (UDP) channels, with minimal assumptions Request routing: ÄDirect (performance) or proxy-routed (control) Separation signaling vs. media description: ÄCan add new applications or media types, SDPng Extensibility: ÄIndicate and require proxy and UA capabilities
What is SIP? A Session Initiation Protocol ÄRatified as RFC 2543 ÄBeing refined in RFC 2543 bis A signaling protocol ÄCall-control mechanism ÄSetup – modification – teardown ÄResolves call endpoints ÄDomain name to IP addresses ÄDescribes the session ÄTypically SDP (Session Description Protocol)
SDP's Role in SIP Session Description Protocol - RFC 2327 Ä Describes session information to potential session participants Carried within the SIP message body Defines call attributes Ä Structured language to describe session characteristics Indicates transport protocol and parameters Ä Typically, RTP & payload format Establishes port numbers on which media should be sent Ä Typically, UDP ports 1024 to 65535 Negotiates / exchanges available media capabilities Ä Audio, video, shared apps, chat, … including encoding methods
SIP Attributes Light & simple but flexible Ä Few transactions Ä Scalable and extensible Uses ‘Internet' formats & components Ä Text-based messages - HTTP/1. 1 message syntax Ä Internationalized: ISO 10646 char. set, UTF-8 encoding Re-uses common ratified standards Ä SDP; MIME; DNS; URL; HTTP authentication Enables non-standard call set-up information Ä ‘Useful' information may be carried within payload Ä Allows devices to make intelligent call-handling decisions Ä Invokes various high-level services URLs as identifiers Ä Easy to re-direct to web resources (web push/pull) Multicast ready Ä For scaling and announcements (mostly future use)
Secondary DNS Primary DNS Basic SIP Call Flow 1. Register 2. Initiate call request (sip: [email protected] net) 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 Cache (1) SIP Proxy (5) Registry (1) ubiquity. net 204. 1. 64. 200 qwest. net 192. 1. 10. 1 Local (1) (2) (7) A UA sip: [email protected] net (192. 1. 100) (7) 'CALL JANE' (6) media transport network B UA sip: [email protected] net (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! Participation in SIP Bake-Offs (SIPit) From RFC Release to Present Day Organizations Participating 60 57 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: Si. Pi. T
Columbia CS Phone System My. SQL 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 T 1 Black Phone sipd SIP/RTP Cisco 2600 POTS sipum Proxy/Redirect Server Sun Solaris PC Linux/Free. BSD/NT 'Plug ‘n SIP sipc Unified Messaging Server SIP/RTP 802. 11 b 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 Ä Minimal clients and feature programming Ä H 323 and IN were/are not easy Industry-standard platforms, web servers and IP infrastructures enable new services Ä Most of these platforms already exist in the network Ä SIP helps tie them together New signaling and services architecture that is widely adopted Ä 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 Ä 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 Vo. IP PBX/CBX Trends Converged PBX (CBX) Ä Packet-based PBX; 4. 1% of worldwide PBX sales in 2000; 19% in 2004 PC CBX ÄSmall system for small business (CPE/CLE) IP CBX ÄLarger systems (carrier network based)
High-Level Sip Opportunities Presence management Personal & session mobility User profiling Web call centers Desktop call management Voice-enabled e-commerce Mobile (3 GPP) adoption Location services Unified messaging Instant messaging
Current Relationship With Qwest NEED THIS INFO – SALES?
Henning Schulzrinne Associate Professor, Columbia University ÄDepartment of Computer Science and Electrical Engineering Ubiquity Software, Corp. Board Member ÄSince March 2001 Acknowledged as the architect of SIP ÄCo-Authored RFC 2543 with aid of student and colleagues Other related experience Includes: ÄInternet telephony; Internet multimedia; quality-ofservice; mobility; security Other co-authored RFC's Include: ÄRTSP & RTP
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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 3 GPP signaling protocol December 2001: 6 th bake-off, 200+ participants March 2001: 7 th bake-off, first time outside U. S.
Vo. IP Signaling Architectures MGCP, Megaco = master / slave H. 323 = (Mostly) single administrative domain SIP = Peer-to-peer, cross domain
Vo. IP Architectures Feature SIP H. 323 Megaco/MGCP Multiple Domains Third-Party Control X X ? - Single-domain Multimedia End System Control Extensible Generic Events CGI Scripting Servlets CPL X X X X Fixed Set X ? X Unlikely Limited -
SIP Inheritance URLs: Ä General references to any Internet service ('forward to email') Ä Recursive embedding HTTP: Ä Basic request/response format, status codes, authentication, … Ä Proxies (but no caching) Ä CGI programming interface; servlets Email/SMTP: Ä Addressing ([email protected]) Ä MX SRV records for load balancing and redundancy Ä Header / body separation, MIME
SIP Design Choices Transport protocol neutrality: ÄRun over reliable (TCP, SCTP) and unreliable (UDP) channels, with minimal assumptions Request routing: ÄDirect (performance) or proxy-routed (control) Separation signaling vs. media description: ÄCan add new applications or media types, SDPng Extensibility: ÄIndicate and require proxy and UA capabilities
What is SIP? A Session Initiation Protocol ÄRatified as RFC 2543 ÄBeing refined in RFC 2543 bis A signaling protocol ÄCall-control mechanism ÄSetup – modification – teardown ÄResolves call endpoints ÄDomain name to IP addresses ÄDescribes the session ÄTypically SDP (Session Description Protocol)
SDP's Role in SIP Session Description Protocol - RFC 2327 Ä Describes session information to potential session participants Carried within the SIP message body Defines call attributes Ä Structured language to describe session characteristics Indicates transport protocol and parameters Ä Typically, RTP & payload format Establishes port numbers on which media should be sent Ä Typically, UDP ports 1024 to 65535 Negotiates / exchanges available media capabilities Ä Audio, video, shared apps, chat, … including encoding methods
SIP Attributes Light & simple but flexible Ä Few transactions Ä Scalable and extensible Uses ‘Internet' formats & components Ä Text-based messages - HTTP/1. 1 message syntax Ä Internationalized: ISO 10646 char. set, UTF-8 encoding Re-uses common ratified standards Ä SDP; MIME; DNS; URL; HTTP authentication Enables non-standard call set-up information Ä ‘Useful' information may be carried within payload Ä Allows devices to make intelligent call-handling decisions Ä Invokes various high-level services URLs as identifiers Ä Easy to re-direct to web resources (web push/pull) Multicast ready Ä For scaling and announcements (mostly future use)
Secondary DNS Primary DNS Basic SIP Call Flow 1. Register 2. Initiate call request (sip: [email protected] net) 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 Cache (1) SIP Proxy (5) Registry (1) ubiquity. net 204. 1. 64. 200 qwest. net 192. 1. 10. 1 Local (1) (2) (7) A UA sip: [email protected] net (192. 1. 100) (7) 'CALL JANE' (6) media transport network B UA sip: [email protected] net (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! Participation in SIP Bake-Offs (SIPit) From RFC Release to Present Day Organizations Participating 60 57 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: Si. Pi. T
Columbia CS Phone System My. SQL 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 T 1 Black Phone sipd SIP/RTP Cisco 2600 POTS sipum Proxy/Redirect Server Sun Solaris PC Linux/Free. BSD/NT 'Plug ‘n SIP sipc Unified Messaging Server SIP/RTP 802. 11 b 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 Ä Minimal clients and feature programming Ä H 323 and IN were/are not easy Industry-standard platforms, web servers and IP infrastructures enable new services Ä Most of these platforms already exist in the network Ä SIP helps tie them together New signaling and services architecture that is widely adopted Ä 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 Ä 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 Vo. IP PBX/CBX Trends Converged PBX (CBX) Ä Packet-based PBX; 4. 1% of worldwide PBX sales in 2000; 19% in 2004 PC CBX ÄSmall system for small business (CPE/CLE) IP CBX ÄLarger systems (carrier network based)
High-Level Sip Opportunities Presence management Personal & session mobility User profiling Web call centers Desktop call management Voice-enabled e-commerce Mobile (3 GPP) adoption Location services Unified messaging Instant messaging
Mobility, Presence & Profiles User profile Database PROF ILE Application Services Broker Data Base RE GI ST ER Voicemail Server VM Server ASB T S O U N NK O CALL Long Distance Slammer K MOM CALL OK R EG IS TE R SIP Signaling Network – WN D EN VM O BOSS Services associated with a user not a device Ä User may have multiple associations Presence management for single ‘number' reachability Selective call forwarding based on profile Ä 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 Voice. XML Web Server Voice. XML 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'
3 rd Generation Partnership Project Application services broker – services and applications environments Data base Authorize Qo. S Resources ASB Server Service Control PCSCF SCSCF Home Network # 1 Calling Party Resource Reservation 3 GPP Release 5 - sample call between different service providers 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 – 1 st. point-of-contact. emergency service break-out and triggers local services (e. g. , directory, Qo. S 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 = Home Subscriber Server = Current location information (superset of GSM HLR (Home Location Register))
Location-Based Mobile Services Home Subscriber Service (1) Application Services Broker Dial-a. Cab 1 Cab 2 Cab Web Server HSS Cab 1 Dial-a-Cab 2 (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 Adobe zii patcher 4 3 5.
Impact to Qwest Does Qwest need to invest in disruptive technology? Ä Have the CLEC threats diminished? Ä Will box/software providers playing in the edge be able to sell CLASS features? Should Qwest fall back on traditional revenue streams? Ä New services Ä Adding value to popular services Ä Reducing costs Should Qwest embrace or slow down technology adoption process? Ä 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 World. Com AT&T British Telecom Telia Microsoft
Level 3 Very active in the SIP arena ÄIntegral part of their softswitch strategy ÄActive in standards bodies and working groups ÄAnnounced industries first SIP-based IP voice network ÄInteroperability certification program Ä(3)Works voice certification program ÄDesigned stateless core proxy in-house ÄWorking closely with companies like Ubiquity on edge strategy ÄAggressive plans to expand capabilities and offerings ÄShunning traditional telephony applications ÄLess vertically integrated than World. Com, for example ÄNot attempting to reinvent the PSTN
Basingstoke 1 0 – stealth and arcade action game. Worldcom Very active in the SIP arena ÄEmploys major SIP advocate and promoter ÄHenry Sinnreich - Distinguished Member of Engineering ÄDesigned proxy in-house ÄOpened up to public for interoperability testing Ähttp: //sipaccount. wcom. com/sipregistration. html ÄRecently announced a fully SIP-based Service Ä'IP Communications' service ÄRetail offering of hosted business communications applications ÄIP Centrex (PBX replacement) Plus ……. ÄTargets midsize to large customer base ÄUsing a broker architecture to layer services ÄDesigned in-house or from 3 rd party vendors ÄPlans to offer SIP phones ÄCan be seen as a major play to undermine Class 5 services
AT&T Taking the usual 'early majority' stance ÄEmbracing SIP for future Vo. IP support ÄCurrently using H. 323 until SIP is broadly accepted ÄExpected to fully adopt SIP and replace H. 323 in 12 to 18 months ÄFocusing on enterprise VPNs and managed services ÄManaged Internet Service (MIS) – IP ÄManaged Router Service (MRS) – Frame
British Telecom Publicly Evaluating Ubiquity Products for Advanced Services ÄWorking with the Ubiquity product portfolio to create advanced, new, services ÄFocus on both residential & business verticals ÄInitial services to roll-out shortly
Telia Early adopter of SIP-based applications and services Ä‘Second-line' residential services targeted at teenagers ÄPresence; call profile; web push; IM ÄFocus on specific vertical markets ÄMarket-specific applications ÄNetwork-based / hosted ÄCall profiles; presence; IM ÄEmploying an applications service broker architecture
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Microsoft Making a huge play for ubiquitous support of SIP at all levels ÄUnder the '. NET' architecture umbrella ÄWindows XP (GA) ÄSIP-enabled version of messenger ÄSIP user agent / client ÄWindows XP Server (July 2002) ÄExtensible SIP proxy server ÄWindows CE (July 2002) ÄSIP user agent / client ÄWindows Embedded - OS for Appliances (July 2002) ÄSIP user agent / client ÄSIP phone (Q 1 2002) Ä'Stinger' ÄXbox gaming platform (Nov. 2001) Ä'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) France. Telecom (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: Ä Ubiquity + Carrier Enterprise Ä Ubiquity + NEV Carrier Ä 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 ÄOptimized for service delivery Service Aware ASB ÄOptimized for speed Fast Stateless Net Server SIP Proxy Non-Service Aware Slower Statefull Core Routing
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SIP Network Server SIP Load Balancing Manager SIP Engine SIP ENUM DNS Authentication Module Redirect Server Location Service Module Transaction Stateful Proxy Registrar Module 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 Presence User Agent Module Authentication Module Transaction Stateful Proxy Registrar Module Location Service Module Service Subscriptions 3 rd Party Call Control CPL Engine Service Configuration Media Push/Pull Service Policy Service Aggregation 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 Server SIP Endpoint ‘A' Enhanced Services SDP/SIP Proxy SIP Signaling Network Source-Routing 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 Hello, everybody! Application Services Broker Altavista Babelfish SOAP SMS URI: 'urn: xmethods. Babel. Fish' call. set. Method. Name: 'Babel. Fish' translationmode: 'en_fr' Sourcedata: 'Hello everybody!' Hello everybody! SMS Gateway Short Message Service (SMS) Gateway ASB SIP Signaling Network FIXED USER Mobile Network MBS MESSAGE sip: [email protected] net; translate=en_fr. SIP/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
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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) Ä Java. Beans to Interface with ASB Modules Ä License to Develop and Upload CPL Scripts onto the ASB Ä Java. Docs Detailing the APIs Ä Extensive Documentation and Sample Code Service modules in the ASB are building blocks whose functionality is accessed via the Design. Deck 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 Ä 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 LDAP WEB Server Call Logic (Subscribe) Update Presence (NOTIFY) Call Set-Up Message (UDP) 3. PM Element Detects Change-of-Sate and Triggers 3 PCC Element, INVITEing the Two Third Parties 'B' ONLINE OFFLINE IP Transport Network ter HT SIP Signaling Network Regis TP ite Inv call when available service Invite 'Automatically Establish Call When SIP Endpoint ‘B' Becomes Available' ASB Invite(s) HTTP 2. Endpoint ‘B' Notifies Availability via SIP REGISTER – Presence Status Updated M O D U L E S JDBC Register 1. Set Call Profile Via Web Interface using Java Server Pages (JSP) S E R V I C E Data base Execute Service Logic CALL PROFILE SIP Endpoint ‘A' Media SIP Endpoint ‘B'
Question & Answer Session OPEN FORUM
