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SIP and VoIP State of the Nation Internet2 Sip.EDU Workshop Minneapolis, Minnesota Feb 2007

SIP and VoIP State of the Nation Internet2 Sip.EDU Workshop Minneapolis, Minnesota Feb 2007. Walt Magnussen, Ph.D Texas A&M University Director TAMU ITEC. Agenda. Standards Platforms Services. Standards. The following slides are provided by Dr. Henning Schulzrinne, Columbia University.

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SIP and VoIP State of the Nation Internet2 Sip.EDU Workshop Minneapolis, Minnesota Feb 2007

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  1. SIP and VoIPState of the NationInternet2 Sip.EDU WorkshopMinneapolis, Minnesota Feb 2007 Walt Magnussen, Ph.D Texas A&M University Director TAMU ITEC

  2. Agenda • Standards • Platforms • Services

  3. Standards • The following slides are provided by Dr. Henning Schulzrinne, Columbia University

  4. Evolution of VoIP “how can I make it stop ringing?” long-distance calling, ca. 1930 “does it do call transfer?” going beyond the black phone “amazing – the phone rings” catching up with the digital PBX 1996-2000 2000-2003 2004-

  5. SIP is PBX/Centrex ready boss/adminfeatures centrex-style features attendantfeatures from Rohan Mahy’s VON Fall 2003 talk

  6. IETF VoIP efforts ECRIT (emergency calling) ENUM (E.164 translation) SIMPLE (presence) uses SPEERMINT (peering) GEOPRIV (geo + privacy) uses may use uses XCON (conf. control) SIP (protocol) SIPPING (usage, requirements) uses provides IPTEL (tel URL) SPEECHSC (speech services) usually used with MMUSIC (SDP, RTSP, ICE) AVT (RTP, SRTP, media) SIGTRAN (signaling transport) IETF RAI area

  7. A constellation of SIP RFCs Non-adjacent (3327) Symmetric resp. (3581) Service route (3608) User agent caps (3840) Caller prefs (3841) Requestrouting Resource mgt. (3312) Reliable prov. (3262) INFO (2976) UPDATE (3311) Reason (3326) SIP (3261) DNS for SIP (3263) Events (3265) REFER (3515) ISUP (3204) sipfrag (3240) Mostly PSTN Content types Core Digest AKA (3310) Privacy (3323) P-Asserted (3325) Agreement (3329) Media auth. (3313) AES (3853) DHCP (3361) DHCPv6 (3319) Configuration Security & privacy

  8. SIP, SIPPING & SIMPLE –00 drafts includes draft-ietf-*-00 and draft-personal-*-00

  9. RFC publication

  10. ~ 44 SIP-related RFCs published Activities: hitchhiker’s guide infrastructure: GRUUs (random identifiers) URI lists XCAP configuration SIP MIB services: rejecting anonymous requests consent framework location conveyance session policy security: end-to-middle security certificates SAML sips clarification NAT: connection re-use SIP outbound IETF WG: SIP see http://tools.ietf.org/wg/sip’/

  11. 31 RFCs published Policy media policy SBC functions Services service examples call transfer configuration framework spam and spit text-over-IP transcoding Testing and operations IPv6 transition race condition examples IPv6 torture tests SIP offer-answer examples overload requirements IETF WG: SIPPING

  12. Conclusion • Core standards for media and signaling are finished • can build PBX-equivalent devices and services on a large scale • see BT, FiOS, Vonage • Lots of decent server implementations (various vendors; SER, openSER, Asterisk) • but lack of good soft clients for major OS platforms • Ossification of Internet requires application complexity • kludge around NATs, lack of QoS • lack of credential infrastructure • Intersection with policy and business models • NGN, 3G: maintain voice as high-value monopoly service • Not a protocol engineering effort, systems engineering

  13. Platforms • Enterprise VoIP • Carrier Platforms • Open Source

  14. Enterprise Platforms • Cisco Call Manager • Still supporting Skinny with robust SIP support • Trunk side (4.0 and greater) • Line Side (5.0 and greater) • Nortel CS-1000 BCM and MCS 5100 • Unistim for full feature functionality • Trunk side SIP only at this time (4.0 and greater) • Avaya • Sip support for line and trunk • 3COM • Built on SIP

  15. Carrier Platforms • Exclusively SIP • Major vendor support • Broadsoft (Verizon and many TISP providers) • Cedar Point • Sonus

  16. Open Source • Being implemented in many startup solutions (i.e. DetD) – SIP only as well • ASTERISK • IPTel SER • Open SER • Sip Foundry SIPX • Campus wide solutions • Penn State • UNC • Sam Houston State (Texas)

  17. Services • Trunking • Peering • Hosted Centrex • E-911 • Peer to Peer

  18. Trunking • IP trunk access to PSTN avialable from several service providers today (Level3, Paetec etc.) • Inbound and outbound LD • Local access under LNP • 800 Services • Directory Assistance • Advantages • Allows converged access • Lower cost • Eliminates Local PRI costs • Allows easy diversity (when coupled with LCR)

  19. Peering • Commonly used by VoIP service providers • Uses ENUM for scalability • Various service providers • Voice Peering Forum http://www.thevpf.com/ • Verisign NRD service • Hardware solutions can augment service • Nextone does MOS calculations and dynamic rerouting when necessary.

  20. Hosted Centrex • Talked about a lot by Industry • Currently offered by Verizon HIPC • Under evaluation at TAMU (more to follow)

  21. E-911 • Location of devices not required on most campuses (but highly recommended) • Fixed locations supported by: • Telemanagement systems (i.e. Pinnacle) • ILEC services (Verizon and AT&T) • Hardware solutions (i.e. Cisco) • TAMU solution • Lock down fixed telephones per port • ID mobile devices (softphones) in database • More elequent solutions to appear under NG911

  22. Peer to Peer • Not supported by most campuses but rampant (i.e. Skype) • Blocked by many international service providers • UC Santa Barbra approach is to block http://www.oit.ucsb.edu/connect/skype.asp

  23. Questions ? • Contact info: • Walt Magnussen, Ph.D. • ITEC Director • telecom@tamu.edu • 979-845-5588

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