When will the telephone network disappear?
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Transcript of When will the telephone network disappear?
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): $1.70/MB
anybody can offer phone service only need to handle signaling, not
media traffic no regulatory hurdles
Some differences: VoIP vs. Some differences: VoIP vs. PSTNPSTN Separate signaling from media data path But, unlike SS7, same network lower call
setup delay Avoid CTI complexity of "remote control" Mobile and wireline very similar Any media as session:
any media quality (e.g., TV and radio circuits) interactive games
Differences VoIP vs. PSTNDifferences VoIP vs. PSTN "Switches" (= SIP proxy servers)
are service-transparent: dialog transparency media transparency security transparency topology transparency functional transparency
May not be true in 3GPP
When will it happen?When will it happen? Took much longer than anticipated
in 1995: standards (signaling) not really ready
until this year not just a protocol, but a whole industry
and infrastructure – eco system: OSS billing testing features: conferences, voicemail
Technology evolution of Technology evolution of PSTNPSTN
0102030405060708090
100
1980 1985 1987 1990 1995 2000 2001
electromechanalogdigital
SS7: 1987-1997
When will it happen?When will it happen? Not too soon by traditional phone
companies: Billions of €/$ 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 others
phone email/SIP
web
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:[email protected]: 07605
REGISTER sip:sosLocation: 07605
302 MovedContact: sip:[email protected]: 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._udp SRV 0 0 b1.example.com
0 0 b2.example.com
stateless proxies
Migration strategyMigration strategy1. 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