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January 23-26, 2007 Ft. Lauderdale, Florida SIP Trunking 101 Presenters: Pete Sandstrom, CTO BandTel Janne Magnusson, Director Operations Ingate
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January 23-26, 2007 Ft. Lauderdale, Florida SIP Trunking 101 Session Overview 1. Business Opportunity 2. What is Signaling 3. The SIP & SIP Trunking, Value Prop 4. The Role of the ITSP
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January 23-26, 2007 Ft. Lauderdale, Florida 1. Business Opportunity is Primed IP-PBX Market on track to exceed $10 billion (Synergy Research) By 2008, 55% of all Enterprise Communications Systems will be IP-enabled (Frost and Sullivan) which means the majority of PBXs will be SIP- compatible/ready VoIP Service Providers gaining enormous traction in marketplace
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January 23-26, 2007 Ft. Lauderdale, Florida 2. What is Signaling? Since SIP/SIP Trunking is a standard for signaling, let’s understand what signaling is and the many forms of Voice Over Internet Protocol (VoIP). The 4 major standards- based signaling systems are: 1)H.248 Media Gateway and Control (MEGACO)– an International Telecommunications Union (ITU) (www.itu.it) standard. H.248 originated from the Public Switched Telephone Network (PSTN) using Signaling System 7 (SS7) with dumb telephones.
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January 23-26, 2007 Ft. Lauderdale, Florida 2. What is Signaling? 2) H.323 is an ITU standard and the logical progression from circuit-switched to packet- switched telephone call processing including video services. H.323 networks consist of Call Processing servers, Media Gateways (MG) and GateKeepers.
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January 23-26, 2007 Ft. Lauderdale, Florida 2. What is Signaling? 3) Media Gateway Controller Protocol (MGCP) is a device control protocol developed by Internet Engineering Task Force (IETF) designed to control devices such as Media Gateways (MG) and Integrated Access Devices (IAD). The difference between MGCP and other multimedia control protocol systems is that MGCP allows the endpoints in the network to control the communication session.
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January 23-26, 2007 Ft. Lauderdale, Florida 2. What is Signaling? 4) Session Initiation Protocol (SIP) - an Internet Engineering Task Force (IETF) signaling protocol for internet conferencing, telephony, presence, events notification (emergency calling) and instant messaging. For complete details on SIP, go to www.ietf.org RFC-3261 or http://www.ietf.org/rfc/rfc3261.txt
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January 23-26, 2007 Ft. Lauderdale, Florida A. Bell – The Beginning of Signaling Dumb phones & Dumb network Signaling via Tip & Ring – Loop Start – Ground Start
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January 23-26, 2007 Ft. Lauderdale, Florida Dumb phones & Smart network Signaling via inband (T-1) or out-of-band (ISDN) Ma Bell – Signaling System 7 (SS7) Designed in the 1980s, deployed in the 1990s
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January 23-26, 2007 Ft. Lauderdale, Florida Smart phones & Smart applications SIP network Signaling is via SIP messages sent via UDP/IP No Bell – Internet Adds SIP
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January 23-26, 2007 Ft. Lauderdale, Florida 3. What is Session Initiation Protocol (SIP)? SIP is a telephony signaling protocol that is used to establish a "communications session or connection" such as a telephone call, conference call or other type of communications on an Internet Protocol (IP) network. SIP is a request-response protocol that operates like a "communications browser" protocol such as Hypertext Transfer Protocol (HTTP). SIP is the communications equivalent to such internet protocols such as HTTP and Simple Mail Transfer Protocol (SMTP).
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January 23-26, 2007 Ft. Lauderdale, Florida SIP “Invites” Callers Together
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January 23-26, 2007 Ft. Lauderdale, Florida SIP is a Fully-Featured Protocol
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January 23-26, 2007 Ft. Lauderdale, Florida What is SIP Trunking? SIP Trunk is a network interface device that recognizes SIP signals (signaling) and can process these signals to other SIP devices. In other words, a SIP Trunk is an expandable or “virtual” digital data transmission link with SIP protocol signaling interfaces that recognizes SIP signaling and SIP media using Real-Time Protocol (RTP) for media transmission (voice, rich media).
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January 23-26, 2007 Ft. Lauderdale, Florida SIP Trunking Applications –Local and inbound calling –G.711 and G.729a Codecs –Re-Invite support across multiple locations –Inbound load balancing options across multiple locations –Disaster recovery options –Directory Assistance –Enhanced 911 services access –411 –Directory Listing –Operator assisted dialing –Local Number Porting Management
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January 23-26, 2007 Ft. Lauderdale, Florida SIP Trunking Value Proposition SIP has become “the” defacto standard for IP communications: SIP offers a set of formal rules that provides the basic signals used to initiate, manage, and terminate IP communications sessions. SIP goes beyond voice to support true multimedia communications/convergence – i.e., voice, data, video SIP embraces and supports enhanced communications capabilities, such as presence and mobility applications Reduces toll charges from SIP origination/termination services to the PSTN for external long-distance and local access calls Optimizes Utilization of Enterprise IP Pipes
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January 23-26, 2007 Ft. Lauderdale, Florida SIP Supports Legacy Infrastructures SIP supports for both Legacy and New IP-based Enterprise Communications Systems SIP offers a migration path while retaining investment in legacy equipment while bringing the unique benefits of VoIP SIP offers: –Reduced Transport Costs –Enhanced Services: DID numbers, call transfer between headquarters/branch offices, real-time access to billing and usage
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January 23-26, 2007 Ft. Lauderdale, Florida IP-PBXs Migrate PBXs – ITSPs Emerge ITSP PTSN IP PBX SIP Services GW SAFW SIP-Aware FireWall (SAFW)
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January 23-26, 2007 Ft. Lauderdale, Florida ITSP Do More Public vs. Private IP Pipe ITSP should be able to work in both domains - Public Internet - Private – Virtual Private Network (VPN) - Both can be completely secure, but VPN costs more
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January 23-26, 2007 Ft. Lauderdale, Florida SIP Over Public Internet or VPN
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January 23-26, 2007 Ft. Lauderdale, Florida SIP Over Public Internet or VPN
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January 23-26, 2007 Ft. Lauderdale, Florida SIP Over Public Internet or VPN
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January 23-26, 2007 Ft. Lauderdale, Florida Security and Reliability Need to see that the CPE LAN is correctly designed for VoIP CPE protection- a SIP Aware Firewall (SAFW) is needed allowing Layer 5 (L5) security (i.e. no L2 pinholes) ITSP Message Digest 5 (MD5-digital encryption coding) is minimal requirement IP authentication is needed ITSP should split media and signaling to different redundant locations ITSP must have secure POPs that can fend off all outside attacks, including source-route bridging spoofing detection.
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January 23-26, 2007 Ft. Lauderdale, Florida 4. The Role of the ITSP SIP User Agent is Plug-and-Play with ITSP (Interoperability Program) ITSP Architecture Black Box to SIP User Agent Core ITSP infrastructure is redundant and resilient Rich diversity of PSTN carriers and routes to global destinations
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January 23-26, 2007 Ft. Lauderdale, Florida What Does an ITSP Look Like
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January 23-26, 2007 Ft. Lauderdale, Florida ITSPs “Peer” For Customer Performance
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January 23-26, 2007 Ft. Lauderdale, Florida ITSP “Peering” For Total SIP Solution ISP IP PBX
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January 23-26, 2007 Ft. Lauderdale, Florida SIP Trunking 101 Summary SIP Trunking brings “plug-and-play” interoperability for existing PBX systems and hosted VoIP systems SIP Trunking simplifies deployment and support chores for distributors and channel partners SIP Trunking brings offers new features/services in the form of “open” applications development SIP Trunking provides geographical unification of the enterprise SIP Trunking brings enormous cost savings to customers
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January 23-26, 2007 Ft. Lauderdale, Florida
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Important to have a reliable and well dimensioned network –Consider delay and QoS The firewall is the demarcation point for the enterprise network Secure privacy and prevent eavesdropping Communication on the LAN
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January 23-26, 2007 Ft. Lauderdale, Florida Important to have a reliable and high quality Internet connection –Consider delay to ITSP –Of your connection QoS (voice should have priority) Voice travels over public Internet (as e-mail) Possible to secure privacy and prevent eavesdropping Communication outside the LAN
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January 23-26, 2007 Ft. Lauderdale, Florida
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Questions?
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January 23-26, 2007 Ft. Lauderdale, Florida About Ingate Formed 2001 –Firewall technology from Cendio Systems Appliance firewalls since 1994 –Capital and SIP technology from Intertex Data AB Began SIP development in 1998 Released the worlds first SIP capable Firewall in 2001 Located in Stockholm and Linköping, Sweden with a subsidiary, Ingate Systems Inc., based in Hollis, NH. Confirmed IP-PBX interoperability: 3Com, Asterisk, Avaya, Broadsoft, Cisco Call Manager, Ericsson MX-One, Mitel, Pingtel, SER, Shoretel, Sphere, Swyx, Zultys Confirmed carrier interoperability: Bandtel, Broadband.com, Cbeyond, Global Crossing, IP-Only, O1, RNKTel, Tele2, VoEx
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January 23-26, 2007 Ft. Lauderdale, Florida About BandTel Headquartered in Newport Beach, California, BandTel is a leading worldwide provider of SIP Trunking services. BandTel is dedicated to ensuring its customers and partners alike have access to the most reliable, end-to-end VoIP service available on the market today. BandTel’s N-Plus™ network architecture is designed to solve the throughput and redundancy problems on high-capacity SIP-based networks and eliminate any single point of failure. BandTel continues to develop strong partnerships with leading carriers and telecommunications companies, including Global Crossing, XO Communications, Level 3, Qwest Communications, Verizon Business, and Primus.
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