Computers and the Public Switched Telephone Network

1
Computers and the Public Switched Telephone
Network

• Martin Taylor
• Warwick CS321 22 November 2005

2
Introduction Data Connection Ltd

• Founded 1981
• Based in Enfield
• 320 employees
• Software related to communications
• Protocol stacks
• Applications e.g. messaging, conferencing,
  directory services
• Next-generation voice networks
• Customers
• OEMs (Cisco, Lucent, Nortel, Microsoft . . .)
• Service providers (SBC, Verizon, BT . . .)
• Revenues and profits have grown every year

3
Introduction MetaSwitch

• A division of Data Connection
• 120 employees
• Set up in 1999
• Next generation local exchange systems for Tier
  2 / 3 phone service providers in North America
• Leading vendor in this space
• gt 100 customers
• gt 150 systems deployed

4
Introduction speaker

• Martin Taylor
• VP Product Management Technology Strategy
• Career outline
• University of Cambridge (Engineering)
• GEC (IT management, fibre optic products)
• Madge Networks (IBM-compatible LAN products)
• CopperCom (Silicon Valley startup in Voice over
  DSL)
• MetaSwitch

5
Topics for today

• Public Switched Telephone Network evolution
• Historical perspective
• The Internet changes everything!
• Why voice migrates to IP
• The competitive landscape
• Where things are at now
• Architecture of next-gen voice networks
• VoIP protocols
• Hardware in the PSTN
• Software in the PSTN
• The brave new world IMS (the IP Multimedia
  Subsystem)
• Challenge

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Traditional PSTN (post 1980)
Multiplexed 64 kbps digitalchannels
Long DistanceTandems
AccessTandems
Dedicatedcopperpairs(loops)
LocalExchanges
7
The Internet Changes Everything

• Service providers build new infrastructure for IP
  networking
• Data network operates in parallel with voice
• Two networks to manage and maintain
• Copper loops deliver broadband
• Universal end-to-end IP connectivity
• BT 6.2m DSL connections (Sept 05)
• Data traffic overtakes voice traffic
• And will eventually dwarf it
• Voice can be transmitted over IP
• Not easy, but most technicalproblems now solved

Source Cisco Systems
8
Why voice migrates to IP

• Backbone one IP-based network carries voice and
  data
• Only one network to manage
• Economies of scale
• Access IP-based broadband delivers multiple
  services
• Internet access / Voice telephony / IP-TV
• DSL / Cable / Fiber / Broadband wireless
• Only one network to manage
• Voice services innovation is faster and less
  expensive
• Applying lessons of the Web to the telephony
  world
• Competition separation between infrastructure
  and services
• Barriers to entering the voice services market
  are much lower

9
Competitive landscape

• Competition enabled by regulation
• E.g. BT forced by Oftel to open up their network
• Separation between wholesale and retail
• Competitive voice services using traditional
  technology
• Competition enabled by technology
• Universal, always-on IP connectivity with
  broadband access
• Ability to make voice calls over IP connections
• New entrants to voice market such asSkype /
  Vonage / FreeTalk / PlusTalk
• New business models
• E.g. flat monthly rate, unlimited national
  calling

10
Architecture trends in the PSTN

• Away from
• Separate networks for different services
• Vertical Integration monolithic service-specific
  network devices
• Highly specialized telecom protocols, with
  numerous local variations

• Towards
• Converged networks supporting multiple services
• Horizontal Integration devices that perform a
  specific network function across multiple
  services
• Open, standard, Internet-oriented protocols

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Next-gen architecture for voice ( other services)
Signaling and control
Call Agent
Media Gateway
TV
To legacy PSTN
IPphone
TV head end
Multi-ServiceAccess Node
Analog phone
PC
To Internet
VoiceApplicationServer
Router
Residence
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VoIP protocols transport

• Protocol stack adds a great deal of bandwidth
  overhead
• Voice is compressed to save bandwidth
• Traditional PSTN 64 kbps
• VoIP typically 8 kbps (G.729)
• 8 kbps VoIP with overhead ? 31.2 kbps overall
• VoIP introduces substantial delay ( 25 50 ms)
• Results in audible echo ? needs echo cancellation
• Can reduce bandwidth consumed by suppressing
  packets that contain silence
• For all this need lots of Digital Signal
  Processing!
• Typically 10-15 MIPS per voice channel

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Location of conversions to and from VoIP
Signaling and control
Call Agent
Media Gateway
TV
To legacy PSTN
IPphone
TV head end
Multi-ServiceAccess Node
Analog phone
PC
To Internet
VoiceApplicationServer
Router
Residence
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VoIP protocols peer-to-peer signaling

• Session Initiation Protocol (SIP)
• Used by VoIP endpoints such as IP phones to
  signal call setup requests into the network
• Used by network entities to set up VoIP sessions
  across a long distance backbone
• Text-based protocol, heavily based on email
  standards
• Supports voice, video, IM and presence
• Defined by the IETF (Internet Engineering Task
  Force)

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Example SIP call setup request
INVITE sip5089042190_at_10.255.19.3 SIP/2.0 Via
SIP/2.0/UDP 10.255.10.35060branchz9hG4bK-omty3k
5z831frport From "Henry Blofeld"
ltsip2125489876_at_10.255.19.3gttagd8r3jjuitk To
ltsip5089042190_at_10.255.19.3gt Call-ID
3c267431493e-di6faqvpvm3e_at_10-255-10-3 CSeq 1
INVITE Max-Forwards 70 Contact
ltsip2125489876_at_10.255.10.35060linecarmrlfqgt Us
er-Agent snom190-3.56y Allow INVITE, ACK,
CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK Allow-Events talk, hold, refer Supported
timer, 100rel, replaces Session-Expires
3600 Content-Type application/sdp Content-Length
216 v0 oroot 786984004 786984004 IN IP4
10.255.10.3 scall cIN IP4 10.255.10.3 t0
0 maudio 10008 RTP/AVP 0 101 artpmap0
pcmu/8000 artpmap101 telephone-event/8000 afmtp
101 0-15 aptime20 asendrecv
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VoIP protocols device control signaling

• Used by intelligent network entities to control
  dumb media gateway devices
• Media Gateway Control Protocol
• IETF MGCP
• ITU-T / IETF Megaco (H.248)
• Used to send instructions such as
• Play dial tone and collect dialed digits
• Connect voice port X to VoIP port Y
• Used to receive notification of events
• User Z has gone off hook and dialed the digits
  5089042190
• User Z has hung up

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VoIP signaling protocols SIP MGCP
Signaling and control
Call Agent
Media Gateway
TV
To legacy PSTN
IPphone
TV head end
Multi-ServiceAccess Node
Analog phone
PC
To Internet
VoiceApplicationServer
Router
Residence
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Hardware in the PSTN current generation

• System X standard UK local exchange
• Original design by the Post Office
• Manufactured by GEC / Plessey / STC
• First deployment in 1981
• UK network 100 digital in 1998

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Hardware in the PSTN next generation
MetaSwitch CA9020 Call Agent 1 million Busy Hour
Call Attempts 200,000 subscribers
MetaSwitch MG3510 Media Gateway Connects VoIP
calls to legacy PSTN Supports up to 24,000
concurrent calls
Combine these two devices with Multi-ServiceAcces
s Nodes from other vendors . . . . . . and you
have a system that is capableof replacing the
largest System X exchange
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PSTN hardware evolution

• Then . . .
• Custom bit-slice processors
• 100 proprietary hardware design

• Now . . .
• Intel CPU (dual 3.2 GHz Xeon)
• Standard chassis (CompactPCI)
• Gigabit Ethernet backplane
• Off-the-shelf cards
• CPU
• Ethernet switch fabric
• DSP voice processing

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PSTN software evolution

• Then . . .
• Proprietary operating system
• Telecom-oriented programming language CHILL
• Home-grown development environment

• Now . . .
• Solaris x86
• Programming in C/C and Java
• Wide range of development tools

MetaSwitch product currentlycomprises 4.5M lines
of code
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IMS the grand unifying theory of telephony

• IMS IP Multimedia Subsystem
• Consensus between mobile and fixed network worlds
  on the ultimate direction of VoIP network
  architecture
• A set of standards that builds upon best current
  VoIP practice
• Key objectives
• A single network to support voice and video
  telephony to both fixed and mobile terminals
• A common set of services for fixed and mobile
  users
• Fixed / mobile convergence ? one phone number
  for both
• Faster and easier deployability of new services

23
IMS architecture
(greatly simplified)
Diameter
SIPEndpoints
SubscriberProfiles
SIP CallControl
IP
BreakoutCall Agent
H.248
MediaGateway
SIP everywhere except where shown
PSTN
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Environment for new voice applications in IMS

• Example applications
• Ring back when free
• Find me / follow me
• Call forwarding with time-of-day rules
• All such applications are handled by SIP
  application servers
• Standard APIs and protocols for common functions
• Java SIP Servlet API and deployment configurator
• Read / write access to subscriber data
• Write access to charging records
• Fault tolerance and failover
• Application developer only needs tocapture
  service-specific logic in a Java class

Borrowing ideas from the world of Web apps, such
as J2EE containers and HTTP servlets
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BT 21cn Architecture (based on IMS)
Source ETSI TISPAN
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Challenge 8

• Compare a selection of VoIP-based and traditional
  phone services using these criteria
• Sign-up or installation cost
• Pricing for different types of calls
• Minutes-based calling plans
• Availability and cost of various add-on features
  like Caller ID
• Limitations on the service (e.g. countries you
  cant call)
• Limitations on the phone number you can get
• Interesting differentiators (e.g. Web control of
  calling features)
• Focus on broad differences between VoIP-based
  services and traditional services
• Comment on likely impact of VoIP-based services
  on competition in the phone market

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Some of the players
Traditional
VoIP-based
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Find out more about us
http//www.dataconnection.com
5th overall
3rd overall
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