When will the When will the telephone network telephone network disappear?disappear?

Henning SchulzrinneColumbia University

June 2002

OverviewOverview

What is Internet telephony? Why Internet telephony? When? How to transition to IP telephony? What remains to be done?

What is Internet What is Internet telephony?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

What is Internet What is Internet telephony?telephony?

soft phones PSTN phones

Ethernet phones

VoIP protocolsVoIP 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

Why Internet telephony?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

Why Internet telephony?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

Why Internet telephony?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

Why should carriers Why should carriers worry?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

Some differences: VoIP vs. Some differences: VoIP vs. PSTNPSTN 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

Differences VoIP vs. PSTNDifferences 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

When will it happen?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

Technology evolution of Technology evolution of PSTNPSTN

0

10

20

30

40

50

60

70

80

90

100

1980 1985 1987 1990 1995 2000 2001

electromechanalogdigital

SS7: 1987-1997

When will it happen?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

Status in 2002Status 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

Where are we?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”

Where are we?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

How to transition?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

How to transition?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

What remains to be done?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

Challenges: NATs and Challenges: NATs and firewallsfirewalls 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

Challenges: NAT and Challenges: NAT and firewallsfirewalls 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

Naming and addressingNaming 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

Naming and addressingNaming and addressing

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

to type

Naming and addressingNaming and addressing

Have any one of three, need othersphone email/

SIPweb

phone -- ENUM ENUM

email/SIP

LDAP?SIP

-- LDAP?SIP

web tel: sip: --

Naming and addressingNaming 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

VoIP applicationsVoIP 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, ...

Challenges: QoSChallenges: 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

Challenges: QoSChallenges: 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

Challenges: SecurityChallenges: 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

SIP securitySIP 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

Challenges: service Challenges: service creationcreation

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

Emergency callsEmergency 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

Emergency callsEmergency calls

EPAD

INVITE sip:sos@psap.leonia.nj.us

Location: 07605

REGISTER sip:sos

Location: 07605

302 MovedContact: sip:sos@psap.leonia.nj.usContact: tel:+1-201-911-1234

SIP proxyINVITE sip:sos

Location: 07605

common emergency identifier: sos@domain

Scaling and redundancyScaling 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

Scaling and redundancyScaling 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

Reliability: powerReliability: 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

Large systemLarge system

_sip._udp SRV 0 0 sip1.example.com

0 0 sip2.example.com

0 0 sip3.example.com

a2.example.comsip2.example.co

m

sip3.example.com

a1.example.com

sip1.example.com

b1.example.com

b2.example.com

sip:bob@example.com

sip:bob@b.example.com

_sip._udp SRV 0 0 b1.example.com

0 0 b2.example.com

stateless proxies

Migration strategyMigration strategy

1. Add IP phones to existing PBX or Centrex system – PBX as gateway

Initial investment: $2k for gateway

2. Add multimedia capabilities: PCs, dedicated video servers

3. “Reverse” PBX: replace PSTN connection with SIP/IP connection to carrier

4. Retire PSTN phones

Example: Columbia Dept. Example: Columbia Dept. of CSof 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

CINEMA componentsCINEMA components

RTSP

sipum

Cisco 7960

sipvxmlSIP

rtspdsipconfLDAP server

MySQL

PhoneJack interface

sipc

T1T1

sipd

mediaserver

RTSP

SIP-H.323converter

messagingserver

unified

server(MCU)

user database

conferencing

sip-h323

VoiceXMLserver

proxy/redirect server

Cisco2600

Pingtel

wireless802.11b

PBX

MeridianNortel

plug'n'sip

SIP doesn’t have to be in a SIP doesn’t have to be in a phonephone

Event notificationEvent 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

Event notificationEvent 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

SIP event notificationSIP 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

Controlling devicesControlling devices

ConclusionConclusion 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

For more information...For more information...

SIP: http://www.cs.columbia.edu/sip

CINEMA: http://www.cs.columbia.edu/IRT/cinema