1 / 49

When will the telephone network disappear?

When will the telephone network disappear?. Henning Schulzrinne Columbia University June 2002. Overview. What is Internet telephony? Why Internet telephony? When? How to transition to IP telephony? What remains to be done?. What is Internet telephony?.

cedric
Download Presentation

When will the telephone network disappear?

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. When will the telephone network disappear? Henning Schulzrinne Columbia University June 2002

  2. Overview • What is Internet telephony? • Why Internet telephony? • When? • How to transition to IP telephony? • What remains to be done?

  3. What is Internet telephony? • Using Internet protocols to transmit voice in real-time • but multimedia (and Internet radio and TV) is almost the same  every telephone can become a "broadcaster" • not necessarily public Internet • similar to streaming media, but typically human on both ends • also known as VoIP, IP telephony • related voice-over-packet: ATM, FR, MPLS

  4. What is Internet telephony? PSTN phones soft phones Ethernet phones

  5. VoIP protocols • Mostly reuse existing protocols, from IP to LDAP • RTP for transporting audio and video • SIP for setting up sessions (calls) • web-like protocol for negotiation and user location • TRIP for finding gateways

  6. Why Internet telephony? • Residential user perspective • cheaper international calls • U.S. to India, China, Mexico • video calls to relatives • integration with IM and presence – no phone tag • (packaged) programmable services • single number, regardless of medium: • mobile phone • home phone • office phone • easy identifier portability • multiple lines cheaper via cable modem, DSL • video monitoring  don't pay for connect time

  7. Why Internet telephony? • Business user perspective • no feature set differences between large and small businesses • automatic call distribution (VoiceXML) • programmable phone services • like web programming (sip-cgi, CPL, servlets) • every company own web page  every company own phone services • easy integration of email, web, IM, databases • single CAT5 Ethernet wiring plant • PBX maintenance costs • PBX growth limits

  8. Why Internet telephony? • Carrier/ISP perspective • classical switches stagnant • but still expensive • Ethernet switch: $0.04/"circuit" • PBX: $218/circuit • Local telephone switch: $270/circuit • avoid separate management infrastructure for voice • new PSTN services hard to deploy • avoid dog-legged routing for mobile calls • mobile = wireline infrastructure

  9. Why should carriers worry? • Application-specific infrastructure  content-neutral bandwidth delivery • GPRS: $4-10/MB • SMS: > $62.50/MB • voice (mobile and landline): $1.70/MB • anybody can offer phone service • only need to handle signaling, not media traffic • no regulatory hurdles

  10. Some differences: VoIP vs. PSTN • Separate signaling from media data path • But, unlike SS7, same network  lower call setup delay • Avoid CTI complexity of "remote control" • Mobile and wireline very similar • Any media as session: • any media quality (e.g., TV and radio circuits) • interactive games

  11. Differences VoIP vs. PSTN • "Switches" (= SIP proxy servers) are service-transparent: • dialog transparency • media transparency • security transparency • topology transparency • functional transparency • May not be true in 3GPP

  12. When will it happen? • Took much longer than anticipated in 1995: • standards (signaling) not really ready until this year • not just a protocol, but a whole industry and infrastructure – eco system: • OSS • billing • testing • features: conferences, voicemail

  13. Technology evolution of PSTN SS7: 1987-1997

  14. When will it happen? • Not too soon by traditional phone companies: • Billions of €/$ deployed infrastructure • $41 billion (est.) for local switches in U.S. • debt-laden carriers • U.S. CLECs killed by monopolies • But others: • (business) ISPs • cable TV companies

  15. Status in 2002 • 2000: 6b minutes wholesale, 15b minutes retail • 2001: 10b worldwide – 6% of traffic (only phone-to-phone) • up to 30% of U.S.-China/India/Mexico traffic • e.g., net2phone: 341m min/quarter

  16. Where are we? • Not quite what we had in mind • initially, SIP for initiating multicast conferencing • in progress since 1992 • still small niche • even the IAB and IESG meet by POTS conference… • then VoIP • written-off equipment (circuit-switched) vs. new equipment (VoIP) • bandwidth is (mostly) not the problem • “can’t get new services if other end is POTS’’  “why use VoIP if I can’t get new services”

  17. Where are we? • VoIP: avoiding the installed base issue • cable modems – lifeline service • 3GPP – vaporware? • Finally, IM/presence and events • probably, first major application • offers real advantage: interoperable IM • also, new service

  18. How to transition? • Several directions at once: • inside out: • inter-PBX trunks • PSTN backbones • signaling links • outside in: • PBX and IP phones • PC-based soft phones

  19. How to transition? • 3GPP and 3GPP2 have chosen SIP and packet audio/video as the technology for 3G Internet multimedia subsystem (IMS) • mostly "real" SIP, with extensions • walled garden mentality – trying to prevent users from choosing other SIP carriers

  20. What remains to be done? • NAT and firewall traversal • cheaper end systems • naming and addressing • quality of service • reliability • security • emergency (112) features • full IM/presence architecture • conferencing

  21. Challenges: NATs and firewalls • NATs and firewalls reduce Internet to web and email service • firewall, NAT: no inbound connections • NAT: no externally usable address • NAT: many different versions  binding duration • lack of permanent address (e.g., DHCP) not a problem  SIP address binding • misperception: NAT = security

  22. Challenges: NAT and firewalls • Solutions: • longer term: IPv6 • longer term: MIDCOM for firewall control? • control by border proxy? • short term: • NAT: STUN and SHIPWORM • send packet to external server • server returns external address, port • use that address for inbound UDP packets

  23. Naming and addressing • Users will have three types of identifiers, several of each: • phone numbers – random # within city  random # within country for mobile • easy to transcribe & key in on 12-button phones • hard to remember • portability across carriers iffy • email addresses = SIP URIs • user@domain, sip:user@domain • portable if own domain ($20/year) or separate from carrier • a pain for existing devices • but need better alpha input in any event

  24. Naming and addressing • Web URLs – http://www.cs.columbia.edu/~hgs • personal domains? • mostly easy to find (Google), but hard to type

  25. Naming and addressing • Have any one of three, need others

  26. Naming and addressing • ENUM: translate +358 8 883 9111 to 1.1.1.9.3.8.8.8.8.5.3.e164.arpa and look up • SIP-to-x: Return on OPTIONS or 302 • Web-to-x: defined business card rather than text search

  27. VoIP applications • Trunk replacements between PBXs • Ethernet trunk cards for PBXs • T1/E1 gateways • IP centrex – outsourcing the gateway • Denwa, Worldcom • Enterprise telephony • Cisco Avvid, 3Com, Mitel, ... • Consumer calling cards (phone-to-phone) • net2phone, iConnectHere (deltathree), ... • PC-to-phone, PC-to-PC • net2phone, dialpad, iConnectHere, mediaring, ...

  28. Challenges: QoS • Bottlenecks: access and interchanges • Backbones: e.g., Worldcom Jan. 2002 • 50 ms US, 79 ms transatlantic RTT • 0.067% US, 0.042% transatlantic packet loss • Keynote 2/2002: “almost all had error rates less then 0.25%” (but some up to 1%) • LANs: generally, less than 0.1% loss, but beware of hubs • voice can tolerate ~10% random loss • averages are misleading – impairments are bursty  really reliability problem

  29. Challenges: QoS • Not lack of protocols – RSVP, diff-serv • Lack of policy mechanisms and complexity • which traffic is more important? • how to authenticate users? • cross-domain authentication • may need for access only – bidirectional traffic • DiffServ: need agreed-upon code points • NSIS WG in IETF – currently, requirements only

  30. Challenges: Security • PSTN model of restricted access systems  cryptographic security • Dumb end systems  PCs with a handset • Objectives: • identification for access control & billing • phone/IM spam control (black/white lists) • call routing • privacy

  31. SIP security • Bar is higher than for email – telephone expectations (albeit wrong) • Potential for nuisance – phone spam at 2 am • Safety – attacker can prevent emergency calls • Denial of service attacks – a billion more sources of traffic

  32. Challenges: service creation • Can’t win by (just) recreating PSTN services • Programmable services: • equipment vendors, operators: JAIN • local sysadmin, vertical markets: sip-cgi • proxy-based call routing: CPL • voice-based control: VoiceXML

  33. Emergency calls • Opportunity for enhanced services: • video, biometrics, IM • Finding the right emergency call center (PSAP) • VoIP admin domain may span multiple 911 calling areas • Common emergency address • User location • GPS doesn’t work indoors • phones can move easily – IP address does not help

  34. Emergency calls common emergency identifier: sos@domain EPAD REGISTER sip:sos Location: 07605 302 Moved Contact: sip:sos@psap.leonia.nj.us Contact: tel:+1-201-911-1234 SIP proxy INVITE sip:sos Location: 07605 INVITE sip:sos@psap.leonia.nj.us Location: 07605

  35. Scaling and redundancy • Single host can handle 10-100 calls + registrations/second  18,000-180,000 users • 1 call, 1 registration/hour • Conference server: about 50 small conferences or large conference with 100 users • Reliability: single expensive 99.999% system  two cheap 99.7% systems • typical reliability of good ISP: 99.5%  dual-homing • For larger system and redundancy, replicate proxy server

  36. Scaling and redundancy • DNS SRV records allow static load balancing and fail-over • but failed systems increase call setup delay • can also use IP address “stealing” to mask failed systems, as long as load < 50% • Still need common database • can separate REGISTER • make rest read-only

  37. Reliability: power • In US, typically about 1.5-4 hours/year of power outage (SAIDI, 99.95%) • plus ~3 short (< 5 min) outages (MAIFIe) • Alternatives: • cell phone  • UPS in Ethernet switches • Ethernet power on spare pairs

  38. Large system stateless proxies sip1.example.com a1.example.com a2.example.com sip2.example.com sip:bob@example.com b1.example.com sip:bob@b.example.com sip3.example.com b2.example.com _sip._udp SRV 0 0 b1.example.com 0 0 b2.example.com _sip._udp SRV 0 0 sip1.example.com 0 0 sip2.example.com 0 0 sip3.example.com

  39. Migration strategy • Add IP phones to existing PBX or Centrex system – PBX as gateway • Initial investment: $2k for gateway • Add multimedia capabilities: PCs, dedicated video servers • “Reverse” PBX: replace PSTN connection with SIP/IP connection to carrier • Retire PSTN phones

  40. Example: Columbia Dept. of CS • About 100 analog phones on small PBX • DID • no voicemail • T1 to local carrier • Added small gateway and T1 trunk • Call to 7134 becomes sip:7134@cs • Ethernet phones, soft phones and conference room • CINEMA set of servers, running on 1U rackmount server

  41. CINEMA components Cisco 7960 MySQL sipconf rtspd user database LDAP server plug'n'sip RTSP conferencing media server server (MCU) wireless sipd 802.11b RTSP proxy/redirect server unified messaging server Pingtel sipum Nortel Cisco Meridian 2600 VoiceXML PBX server T1 T1 SIP sipvxml PhoneJack interface sipc SIP-H.323 converter sip-h323

  42. SIP doesn’t have to be in a phone

  43. Event notification • Missing new service in the Internet • Existing services: • get & put data, remote procedure call: HTTP/SOAP (ftp) • asynchronous delivery with delayed pick-up: SMTP (+ POP, IMAP) • Do not address asynchronous (triggered) + immediate

  44. Event notification • Very common: • operating systems (interrupts, signals, event loop) • SNMP trap • some research prototypes (e.g., Siena) • attempted, but ugly: • periodic web-page reload • reverse HTTP

  45. SIP event notification • Uses beyond SIP and IM/presence: • Alarms (“fire on Elm Street”) • Web page has changed • cooperative web browsing • state update without Java applets • Network management • Distributed games

  46. Controlling devices

  47. Conclusion • Transition to VoIP will take much longer than anticipated  replacement service • digital telephone took 20 years... • 3G (UMTS R5) as driver? • combination with IM, presence, event notification • Emphasis protocols operational infrastructure • security • service creation • PSTN interworking

  48. For more information... • SIP: http://www.cs.columbia.edu/sip • CINEMA: http://www.cs.columbia.edu/IRT/cinema

More Related