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					     Voice-over-IP

The Future of Communications


           By:
      D. R. LUHAR
    SIGMA TRAINERS
      AHMEDABAD


                               1
              What is internet telephony?


The ability for people talk to each other using the
internet rather than a traditional voice network
carrier.

      Hello                            Hello


                       The
                     Internet




                                                      2
     What does Internet Telephony Cover?

• Internet Carrier Services
• Peer to Peer Communications




                                           3
              What are Internet Carriers?
They are just like a traditional carriers except…
…you connect to them via your existing
internet connection.
                                              Fixed Line
                                            Interconnects




                                                Mobile
           The                              Interconnects
         Internet

                              Internet       International
                                            Interconnects
                               Carrier

                                                             4
                  What do you need?

• an internet connection                         Phone
                                                 Adaptor
a phone adaptor




      The
    Internet          Cable or DSL
                        Modem
                                   Ethernet
                                 Switch/Router




                                                           5
                           How does it work?
• The phone adaptor authenticates and registers with the
  carrier’s server
  When a user makes a call, the phone adaptor converts the
  DTMF & analogue voice and uses SIP to sends it to the carrier
  The Voice Switch converts the data into the appropriate format,
  extracts the number and routes the call to the appropriate PSTN
  interconnect
                                            Authentication
                                               Server
    Phone
    Adaptor

                        Cable or DSL
                          Modem
                                         The                          PST
                                       Internet                        N

                                                             Voice
                                                             Switch
          Ethernet
        Switch/Router

                                                                            6
                           What about incoming?

• The carrier receives a call from the PSTN
It polls the authentication server to establish where to route the call
If the user is not logged on or unavailable, it may provide additional
network services such as voice mail.
If the user is present, it routes the call to the Phone Adaptor which
converts it back to analogue and the phone rings

                                             Authentication
                                                Server
   Phone
   Adaptor

                         Cable of DSL
                           Modem
                                          The                          PST
                                        Internet                        N

           Ethernet
                                                              Voice
         Switch/Router                                        Switch

                                                                             7
          What are their advantages?

• Possible cost savings (more likely on international
  calls than on UK ones)
• Compatible with existing networks
• Do not need a PC (except with the soft phone
  version)
• Portability
• Short lead time
• No long-term contract




                                                        8
         What are their disadvantages?

• Emergency Calls
• Currently only targeting small businesses and
  residential
• Requires specialist hardware (except with soft phone
  option)
• Resilience Issues
• Bandwidth Issues




                                                         9
      Who offers Internet Carrier Services?
• Broadband Providers




Specialist Internet Carriers




Equipment Providers




                                              10
   What is Peer to Peer Communications?

Peer to Peer communications is where one person can
talk to another using only their PC and an Internet
Connection.


         Hello                        Hello


                      The
                    Internet




                                                  11
                       What do you need?
a PC

• an internet connection
a client application
a headset




         The
       Internet                Cable/DSL
                             Modem or Router




                                               12
                            How does it work?
•   The client application logs into the authentication server and
    advertises its presence on the network
    The client application regularly queries a distributed network to update their
    contacts presence and routing information
    When a user wants to talk to someone, they click on the person’s details in their
    directory or search a global directory (if available)
    Depending on their connectivity, the client either contacts the recipient directly or
    via a routing server
                             Distributed
                                                                  Authentication
                              Network
                                                                     Server




            Cable/DSL                                         Cable/DSL
          Modem or Router                                   Modem or Router
                                             The
                                           Internet



                                                                                        13
            What are their advantages?

•   Calls within peer networks are ‘free’
•   No specialist hardware required
•   Portability
•   No lead-time, just download and install




                                              14
        What are their disadvantages?

• Emergency Calls
• Currently more consumer than business focused
• Requires a PC
• Not compatible with other peer to peer networks
• Basic service not compatible with traditional
  landlines and mobiles
• Resilience Issues
• Bandwidth Issues




                                                    15
Who offers Peer to Peer Services?




                                    16
          What about traditional calls?

Most Peer to Peer services either offer (or will shortly
offer) compatibility with traditional telephone services
as a costed option:
   – Skype offer SkypeOut
   – Yahoo offer BT Communicator in the UK and Dialpad in the US
   – Microsoft have bought Teleo




                                                                   17
       What about enterprise solutions?

Most solutions are currently aimed at consumers
rather than enterprises but options are as follows:
   –   deploy an enterprise solution like Microsoft Live
       Communications Server
   –   see what products your current PBX or IPT system will support
   –   introduce Internet Telephony on a small scale using specialist
       devices




                                                                        18
                   What is Microsoft Live
                  Communications Server
It is an enterprise telephony system which supports:
    – Voice
    – Instant Messaging
    – Application Sharing
It features:
    Point of Presence
    Server-based contact lists
    PSTN Integration
    Instant Message Integration with 3rd party providers


                                            PST
                           Live              N
                       Communications
                          Server




                                                  The
                                                Internet


                                                           19
             Handset Products


PSTN




       PBX




                      Switch/Router
                                        The
                                      Internet




                                                 20
       PBX Products


              Skype
 PBX         Exchange
                        Switch/Router
                                          The
                                        Internet




PSTN




                                                   21
                      Security

• SPIM – Spam over Instant Messaging
• SPIT – Spam over Internet Telephony




                                        22
                          Future

• Wireless access (Skype Zones in beta)
• PBX Integration
• Possibly a Virtual PBX or Centrex type solution




                                                    23
          From Circuits to Packets
        Moving Telecom to the IP World


M. V. Pitke
Axes Technologies
IEEE Mumbai ComSoc Chapter


TIFR May 31 2002




                                         24
                        The PSTN

• Circuit Switched, Global global coverage
• Highly modular and standardized
• Highest reliability (1/2 hr down time in 25 years! Five
  Nines!)
• All digital transmission- 64kbps basic channel rate)
• Guaranteed performance- Bandwidth and distortion
• Focus on telephony




                                                            25
                Switching and Signaling


•   Circuit Switch mode
•   Full Duplex and Symmetric
•   Very low latency (delay) <20ms
•   No switch off -modification or upgraded in hot condition
•   Powerful online fault diagnosis and maintenance
•   Extensive use of duplication and N+1 redundancy
•   Tone Signaling
•   Common Channel Signaling SS7 globally connected




                                                               26
                    Transmission

•   All digital except the 'last mile'
•   2 wire local loop 300-3300Hz/40db dyn range
•   64kbps per channel- after compression
•   Higher order 'trunks' to Gbps
•   Variety of media-Cable, Wireless, Optical
•   Access network very important-major cost component




                                                         27
                      Special features

•   Handset powered by the central office
•   All telecom equipment runs on -48V
•   1500 ohm loop resistance
•   Bidirectional balanced loop
•   G3 fax (9600bps) or higher
•   Transparent to signalling and administrative information
•   Smooth movement to ISDN
•   Very strict quality-of-service (QoS) standards --'carrier
    class'




                                                                28
          Impact of Digital Technology

• Dominant role for computers
  /microprocessors/software
• Cheaper, hardware/VLSI, etc
• Impact of DSP--broadband communication on basic
  copper pair, DSL, ADSL, etc
• Limited video facilities
• Ease of development and manufacture--
  opportunities for small entrepreneurs
• Emergence of new innovative products and services,
  leading to the Internet
• Rapid increase in data services




                                                       29
                     ISDN and ATM

• ISDN-Integrated Services Digital Network
• 2-64kbps and 1-8kbps channel on the same copper pair
• Relatively limited impact, except for higher data rates for
  computer networks
• High impact of SS7 signalling system
• Unable to keep pace with rapid developments elsewhere
• ATM-Asynchronous Transfer Mode for broadband service
• User selectable bandwidth
• Both designed by telecom engineers-very high QoS
• More impact on data than on telephony



                                                                30
         Intelligent Networks IN and AIN

• Separated the switching, signalling and service functions
• Ease of creation of virtual and other services
• Some control delegated to the customer (from the service
  provider)
• Improved utilization of resources (800 and 900 type of
  services)
• The network is no more switch-centric, switch becomes a
  peripheral like a voicemail server
• Ability to use products from multiple vendors-- drastic
  fall in prices
• IN model helpful in moving legacy services to the IP
  world

                                                              31
                   The Wireless

• Rebirth after a long gap, large scale adoption of
  mobile communication and help quick expansion of
  the network
• Low cost digital/broadband devices.. to several Gbps
• Impact of low cost DSP solutions, software radios,
  etc.
• Very powerful spectrum conservation techniques
• Emergence of wireless Lan and computer networks
  with low cost devices such as blue-tooth threatening
  3G and PCS services




                                                         32
                  The Internet/IP

• Rapid worldwide coverage and acceptance.
• Very high impact of the Web and its potential
• Message based (unfettered) medium not too many
  constraints on delays and QoS. Problems for
  telephony
• No regulation, very low set up and operating costs
• Universal connectivity across space and time
  moving towards web-centric services
• Ability to integrate very diverse products,
  technologies and services--convergence
• Bits, packets and streams


                                                       33
      Circuit Switched Networks Benefits
                and Drawbacks
Regulatory issues         Infrastructure Cost
Infrastructure in place   Less feature Rich Designed
Ubiquitous                for voice
It works (Reliability)    Expensive
Guaranteed QOS            Poor use of assets
Highly Scalable           High access fees
                          High international settlement
                          charges




                                                          34
Benefits and Drawbacks of Packet Networks

•   Convergence              • Poor Reliability (Echo,
•   New applications           Delay & jitter)
•   Cost reduction           • Limited scalability
•   Consolidate management   • Emerging standards
•   Regulations              • Not ubiquitous
                             • Basic services
                             • At some point all the
                               advantages of today's
                               circuit switched net




                                                         35
 Circuit Switched Voice vs. Packetized Voice

T1.5 /24channels/64K EACH/24 Voice calls max

                 Circuit Voice

T1.5 full use of the bandwidth

              Packetized Voice
      Video                      Data
                    Voice



                                               36
Data Traffic is Growing to Dominate all Public Network Traffic
      (shaded = voice component of total PSTN traffic)

    60%

    50%

    40%

    30%

    20%

    10%

     0%
          98   99     '00    '01    '02    '03    '04



                                                                 37
             Market Forecasted at $300 Billion

$300.0                                    140.00%
             $263.7
$250.0                            121%    120.00%
                                          100.00%   2004 Telephony
$200.0                                              Market Revenue in
                                          80.00%    Billions
$150.0
                                          60.00%    2004 Telephony
$100.0                                              Market Growth Rate
                                          40.00%
                              $37.0
 $50.0                                    20.00%
         4.60%
  $0.0                                    0.00%
         Circuit Switched   IP Switched




                                                                         38
                 Major Applications
                  for Voice Over IP

•   Pre-paid calling cards
•   Voice /Data/Video Integration
•   Voice Enhancements
•   Unified Messaging
•   Internet Call Centers




                                      39
              Telecom v/s Computer

• Clash of cultures
• Telecom- Very conservative, stds first, service later
• Emphasis on legacy, reliability, modularity and inter-
  operability
• Long time scales, unable to keep up with technology
  developments
• Higher costs and investment
• Computers: Tolerance of occasional breakdowns
• Service first and stds (if any) later
• Agile-ability to adapt and innovate
• Low budgets

                                                           40
       Special Problems of Telephony

• Speech does not tolerate delay but can tolerate
  distortion
• Data cannot tolerate distortion (errors), but can
  tolerate delays
• Speech needs lower bandwidth while data is
  generally transmitted at much higher speeds
• In a mixed environment, low speed data (speech) has
  to be given higher priority over high speed data
  packets
• Special techniques employed for handling telephony




                                                        41
      Special Problems of Video/Multimedia

• Very high data rates
• Very high volumes of data (streams)
• Protocols for packet handling are too slow with high
  overhead
• processing by hardware to overcome speed
  limitations
• Handling streams as opposed to packets
• Special care for handling functions like multicasting.




                                                           42
                Multimedia Service

• Very unique problems
• Information handled as streams
• Special multimedia servers
• Special software
• Problems relating to generation, caching and
  distribution of contents
• Network support for multicasting




                                                 43
The Powersurfer




                    PoP/Receiving station

                         Substation



Clusters of buildings


                                            44
             Quality of Service (QoS)

• Very high standards set by telecom (carrier class)
• Unpredictability of the route and nodes traversed ,
  each node adding some (variable) delay
• Emergence of MPLS- Multi-Protocol Label Switching
  technique for management of control of bandwidth
  and delay
• Provision of differentiated services--Guaranteed
  quality and 'best effort' quality
• Control of total transit delay
• Asymmetry of communication
• Carrier class availability 99.999%
• Crisis management--Graceful degradation of service-
  no crashes
• Network security/protection against vandalism
                                                        45
                  General Trends

• Cheaper bandwidth due to large scale worldwide
  fiber deployment
• Disappearance of conventional
  switching/architecture.
• Increasing use of wireless at the last mile
• Speed limitations set by software/processing speeds
• Moving software operations to hardware
• Bits, bytes, packets and now, streams'
• New stream based techniques likely to emerge



                                                        46
                Some Concerns

• Telecom-- dominated by engineers
• TV--dominated by media people
• Information Technology- dominated by management
  and finance people?
• Large gaps between claims and delivery
• Wide scope for bogus products and fraud
• Unreal customer expectations, result of dot-com
  hype and misinformation
• Vast potential of the Internet is yet to be tapped



                                                       47
     Important elements TDM (telecom) to IP
            (internet) transformation

• The Soft switch-- All call processing software
  resides here. It is essentially a platform
• The Gateway --Different type, depending upon
  application
• Enterprise, media, trunk, etc. PSTN links, tel lines, E1
  trunks on one side and Ethernet interface on the
  other (IP cloud)
• Special servers like interactive voice response
  systems, etc,.
• (IN model)

                                                             48
                  Ip Gen Soft switch Solution
              Components of
                                                      Descriptions
           Softswitch Technology

Partners                              • Features and applications such as Class 4/5,
   +          Application/Feature       mobility management, prepaid, voice portals,
                    Server              unified messaging, voice activated dialing etc...

                                      • Flexible, programmable environment able to
                                        interface with servers and applications to
             Applications Interface     support creative and innovative applications
              and Service Creation    • Supports multiple applications interfaces (e.g.
                                        JAIN, JTAPI, S100, etc…)

                                      • Often referred to as the ‘operating system’ of
                                        IP switching, basic call and session processing
                                      • Manages and monitors signal control for
                                        multiple sessions across multiple networks
             Call Control Engine      • Capable of calling upon OA&M functions and
             (Operating System)         customized applications while maintaining the
                                        flow of the call
                                      • Mediates sessions, requests and calls across
                                        multiple networks and protocols (e.g. SIP,
                                        H.323, Megaco, SS7, etc…)

                                       • Converts from TDM to Packets
Partners       Media Gateway           • Examples include IAD, RAS gateways,
                                         ATM switches, traditional CO switches




                                                                                            49
                                                                             Generation MSP
                                                Robust API open to 3rd party application developers
                                                Only flexible call model in the industry(patents pending)
                                                Most complete protocol inter working function(patents pending)
                                                Application Creation Toolkit coupled to Soft switch
                                                              Genovation Application
 User / Craft                                                 Creation Tool Kit (ACT)
                              .
Please complete the informationWhen completed hit . w
                            belo                er


                                    Use add
                                    newr1
                                    feature




Please complete the informationWhen completed hit . w
                            belo
                              .                 er
                                                                             Native Wireline
                                                                                                 Wireless            Wireline              Multimedia
                                                                             & Wireless/WIN
                                                                                                Application         Application            Application
                             OAMP Interface                                   Applications
                                                                                                                                                          Platform
                                                                                                                                                          Mobility
                                                                                                                                                          Services
                                                                OAMP                                                 Open APIs
                                                              Application                                 (Parlay, JTAPI, XML, WML, JAIN)                 Platform
                                                                                                                                                            Call
                                                                                                                     Security                             Services
                                                                                                     Application Interface Adapter ( AIA )
                                                                                                                                                          Platform
                                                                                                                                                           Media
                                                                   D                                   Native API ( NAPI )                                Services
          Hardware and                                             a
         O/S Maintenance                                           t                                                                       Platform      Services
                                                                                                                                                          Platform
                 &                                                 a                                                                       Transaction   Services
                                                                                                                                                         Transaction
       Configuration Interface                                     b                                Session Composer ( SC )                               Services
                                                                                                                                           Services
                                                                   a
                                                                   s
                                                                                                                                                          Platform
                                                                   e                                                                                        Mail
                                                                                                Universal Protocol Engine ( UPE )                         Services
                                                        DB
                                                                   I / I/F   IS-634       IS-41       MGCP            SIP           TCAP          SS7     Platform
                                                                   F         UMTS       GSM-MAP      MEGACO          SIP-T          SMTP         Q.931   Framework
                                                        VLR                                                          H.323                     SIGTRAN    Services



                                                                                                                     SIP
                                                                                                                  Application
                                                                                          HLR

                                                                                                                                                                       50
Gateway Application




                      51
               Intelligent Building Telecom Network

  Second Floor
                                                           Packet VoX   PV 9024-E       Packet VoX   PV 9024-E




                               Ethernet   10/100 Mbps
                                Switch

                               F                                        LAN                          LAN

                               I
                               B
                               E
                               R                           Packet VoX   PV 9024-E       Packet VoX   PV 9024-E




         First Floor           Ethernet   10/100 Mbps
                                Switch


                                                                        LAN                          LAN




    To C.O lines


  Packet VoX       PV 9024-E




Enterprise
                                                                                    To Internet
 Gateway
                               Ethernet
      Ground Floor              Switch                  Router
                                                                                                                 52
                    In Conclusion

• Internet provides an unprecedented opportunity for
  new, innovative, cost effective communication
  services with high impact
• India should benefit from the low cost of hardware
  and the high software and labour content in the new
  worldwide businesses
• There is considerable scope to develop
  products/solutions that are important to us here and
  also have a vast global market




                                                         53
         SIP (Session Initiation Protocol)
The Session Initiation Protocol (SIP) is an
application-layer control protocol that can establish,
modify and terminate multimedia sessions or calls .
These multimedia sessions include multimedia
conferences, distance learning, Internet telephony
and similar applications. SIP can invite both persons
and "robots", such as a media storage service. SIP
can invite parties to both unicast and multicast
sessions; the initiator does not necessarily have to
be a member of the session to which it is inviting.

     SIP Establishes a session connection across the Internet


                                                                54
                        VOIP

• Voice Over Internet Protocol (VoIP) refers to the
  integration of data and voice onto a single Internet
  Protocol (IP) based network.




                                                         55
                     Why VoIP?

• Network Integration:
   – Voice, Data, Video
• Cost
   – Utilizes existing Network Infrastructure and
     investments
   – Existing Networking and Telecomm Support
• Strengths:
   – QoS (Quality of Service)
   – Security
   – Redundancy
   – Improved Features



                                                    56
              Popular VoIP Features

• Caller ID
• Conference Calls
• Extension Mobility (including wireless)
• Unified Messaging:
   – Convert Outlook text mail to voice mail (text to
     speech)
   – Store voice mail in Outlook mail (audio files)
• Voice Recognition (call center, directory assistance)
• XML applications
   – Portal information imported to display
   – Bulletin broadcasts (building or campus)
   – Outlook Contact information
                                                          57
                VoIP Phone Sets
• Hard Sets
              Avaya 4602w IP Telephone
              2 programmable call appearance/feature keys
              2 x 24 character based Eurofont display
              Avaya 4620w IP Telephone

              Large screen graphic display (168x132 dots)
              WML browser capability using standard XML
              LDAP directory access via browser
              24-line appearance buttons




                                                            58
                        SoftPhones




The IP Softphone telephone client can be configured for
“Road
Warriors” or Telecommuters.

In Softphone, an application is installed in your laptop or
desktop and can act as your telephone.

The SoftPhone configuration is suited for users working
from a remote office or while on business trips away from
the office.


                                                              59
           Regular Telephone vs VoIP

• THE ONLY DIFFERENCE BETWEEN VOIP AND
  REGULAR TELEPHONY IS THE PRICE.
   – Internet telephony and regular telephony are unlike
     one another in almost every possible way. Internet
     telephony depends on turning voices into packets of
     data and sending them through a relatively dumb
     network—the Internet. Those packets are sent to
     relatively smart devices: computers, PDAs, and IP
     phones.



                                                           60
                Is VoIP Secure?


To the extent that VoIP is just another data
application, it has no inherent protection against
eavesdropping, but in practice VoIP is even more
secure than old-style telephony. That wasn't
always the case.




                                                     61
          SIP (Establishing a Call)




SIP Establishes a session connection across the Internet

                                                           62
SIP (Establishing a Call, using a
      redirection Server)




                                    63
                        SIP Messages
• Encoding: SIP is a text-based protocol and uses the ISO 10646
    character
• Format : SIP-message = Request | Response
• generic-message = start-line
                        *message-header
                        CRLF
                        [ message-body ]
• start-line = Request-Line | Status-Line
• message-header = ( general-header
                | request-header
                | response-header
                | entity-header )
•   Method =      "INVITE" | "ACK" | "OPTIONS" | "BYE"
                | "CANCEL" | "REGISTER"

                                                                  64
                Multiple Perspectives

         Business                     Technology
• Enables an existing         • Promise of ease of build
  service to be delivered       for enhanced services
  with greater benefits and     that combine voice, data,
  lower costs.                  and video
• Enormous Potential for      • Unified Messaging
  new applications/new
  markets/new players
• eCommerce




                                                            65
                     What is VoIP

VoIP and Internet Telephony are methods which
convert voice signals into digital data and send this
data on IP network as a series of 1s and 0s.

We will use VoIP broadly and understand it to mean the
use of data packets to transmit voice.




                                                         66
           Why is VoIP Important?

IP technology and packet transmission of VoIP
traffic presents a fundamentally different set of
opportunities and challenges than traditional circuit
technology.

The confluence of IP technology and its complexity,
and the financial incentives to use it, reflect
regulatory challenges.



                                                        67
VoIP some examples




                     68
                     Some definitions

PSTN: Public Switched Telephone Network is the telephone
network available for public use including telephone lines, mirco
wave, and other modes of transmission. Both Circuit and IP
networks can and do operated on the PSTN.

Circuit Switched network is the traditional telephone network
that sends information via a fixed transmission line linking the
caller and the recipient. A temporary link on the PSTN is formed
between the information sender (caller) and the information
recipient (call receiver) for the duration of the communication.
No others can use this transmission line during this time.

IP Network transmits data via packets. Here a communication is
divided into smaller packets and the packets are sent
independently from each over, some time via different
transmission lines or routes, and reassembled together at the
end of transmission line.
                                                              69
                     Some definitions

PSTN: Public Switched Telephone Network is the telephone
network available for public use including telephone lines,
mircowave, and other modes of transmission. Both Circuit and
IP networks can and do operated on the PSTN.

Circuit Switched network is the traditional telephone where
information is sent via transmission line linking the caller and
the recipient. A temporary link on the PSTN is formed between
the information sender (caller) and the information recipient (call
receiver) for the duration of the communication. No others can
use this transmission line during this time.

IP Network transmits data via packets. Communications are
divided into smaller packets and the packets are sent
independently from each over, some time via different
transmission lines or routes, and reassembled together at the
end of transmission line.
                                                                70
               Why VoIP is attractive

Cost savings are achieved through more efficient use
of a IP network than a circuit network.
Efficiency is gained with IP since multiple users can
use the same transmission line while on a ciruit
network a transmission line can only be used by the
sender and the recipient.




                                                        71
              Why VoIP is attractive

Cost savings are achieved through more efficient use
of a IP network than a circuit network.
Efficiency is gained with IP since multiple users can
use the same transmission line while on a circuit
network a transmission line can only be used by the
sender and the recipient.




                                                        72
               Why VoIP is attractive

Cost savings are achieved through more efficient use
of an IP network than a circuit network.

Efficiency is gained with IP since multiple users can
use the same transmission line while on a circuit
network a transmission line can only be used by the
sender and the recipient.




                                                        73
Circuit and IP Networks Transmission lines
Circuit Network           IP Network




                                             74
       How VoIP is Used—some examples

• Two party Traffic—two individuals.
• Third Party Traffic—two individuals with transport
  service provided by a third party.
• Private Networks—Banks, Government, Educational
  institutions.
• Call Center—Circuit call initiated and routed from a
  PSTN to an IP network.
• Pirates—use of VoIP to by-pass Telephone company
  and avoid Telephone company termination tariff,
  taxes, US/UA fees.


                                                         75
                 Key Facts re VoIP
• VoIP is a nascent technology….

• VoIP is a “borderless” technology…unlike the circuit-
  switched network, the IP network is
  “connectionless”…traffic is global and no longer
  defined within the limited jurisdiction of states.

• VoIP is part of an IP network that is being built-out
  and interconnected by robust intermodal
  competition…there is no one dominant player.

• VoIP is spurring price competition and new service
  offerings…(e.g., Cablevision offering unlimited
  enhanced VoIP service and e911 for $34.95/mo…¾ of
  broadband customers taking).
                                                          76
                 Guiding Principles
• Salute Capitalism. In a competitive market,
  economic regulation is a disincentive to the
  investment that will be required to build-out the IP
  networks of the future.

• Competition Benefits Consumers. Policy should
  recognize & respect that intermodal competition
  (i.e., phone vs. cable vs. VoIP vs. wireless) benefits
  consumers.

• Emerging Technology. As VoIP is an emerging,
  competitive technology, VoIP providers should not
  be subject to rules designed to forge competition in
  established, monopoly markets.
                                                           77
            Guiding Principles (cont.)
• No Economic Regulation. Where VoIP is provided
  purely as an application over a broadband network
  (pure VoIP), there should be no economic regulation
  (i.e., regulation of prices, service quality, etc.).

• Regulatory Parity. VoIP providers – whether new
  firms, IXCs, or LECs – should be subject to the same
  (de)regulatory regime.

• Interstate in Nature. Because VoIP technology is
  borderless, VoIP services should be presumed to be
  inherently interstate in nature (at least absent clear
  evidence to the contrary in a particular case).
                                                           78
            Guiding Principles (cont.)

• There’s VoIP and Then There’s VoIP. Distinctions
  between pure VoIP providers and POTS providers
  that use VoIP merely as means of transport may call
  for policy differences.

• Limited “Necessary” Regulation. VoIP providers do
  not have to be classified as CLECs, and VoIP need
  not be subjected to full range of telecom regulation in
  order to address public safety and welfare issues
  (e.g., E911 and USF).



                                                            79
          Devil is in the Details


• Access Charges     • Consumer Protection

• E911               • Numbering

• Service Quality    • TDD Compatibility

• USF Issues         • VoIP as Transport




                                             80
                  Access Charges


VoIP Scenarios:

•   Pure VoIP: VoIP Phone/Computer to VoIP
    Phone/Computer without touching the PSTN.

•   POTS: Plain old telephone to telephone with IXC
    using VoIP for transport in the IXC’s enterprise.

•   VoIP Phone to Plain Old Telephone: VoIP is used to
    transport portion of the call, but the PSTN is relied
    upon for delivery of the call.
                                                            81
                 Access Charges (cont.)

• The entry of new providers & new types of providers is
  an opportunity for reform of rules, but in the
  meantime…

• Access Charges Should Only Apply to VoIP Where
  PSTN is Accessed (and Only to Extent
  Accessed)…Intercarrier compensation/access rules
  would apply to that portion of a VoIP call that relies
  upon the switched network.

• Simple “But For” Test Could Apply. If a VoIP call could
  not be made but for access to the PSTN, then the VoIP
  provider would be subject to access charges under this
  approach.
                                                            86
                          E911

• Guiding Principle # 1 – Public Safety Argues for a
  Ubiquitous 911 System. Consumers want 911
  services. At the time of need, callers may forget
  they are using a VoIP phone. A child in danger
  should not be deemed outside the 911 system
  because her parent opted for a VoIP system.

• Guiding Principle # 2 – Those Utilizing the 911
  System Should Support the 911 System. A VoIP
  provider that transfers calls to the 911 system
  should bear its “fair share” of maintaining the 911
  system. Regulatory parity argues that those who
  use the system should, regardless of the platform
  used, support the system.
                                                        88
                    E911 (cont.)
• Guiding Principle # 3 – Afford a Reasonable
  Opportunity for Industry to Develop Standards.
  Public safety regulation ultimately applied to VoIP
  should allow a reasonable opportunity for
  providers to develop & implement solutions (e.g.,
  for syncing VoIP with the circuit-switched e911
  system).

• Guiding Principle # 4 – Shared Responsibility.
  Industry has responsibility to fully inform
  consumers. Consumers have a duty to educate
  themselves and understand that their use of a
  competitive, emerging communications service
  may have a different 911 functionality than their
  plain old telephone.
                                                        89
                  Universal Service
• The Debate: Nascent technologies should not be
  burdened with old taxes, but the country has
  established universal service policies that require
  funding.

• The Problem: As consumers increasingly turn to
  substitutes for a taxed service, not subjecting those
  substitutes to universal service fund obligations
  picks winners and losers. Some competitors but not
  others would bear the brunt of funding the program.

• The Need: Reassess how competitive market should
  impact universal service business plan.
                                                          90
            Universal Service (cont.)
• Guiding Principle # 1 – Expansive Regulation is Not
  Required. VoIP (like wireless) does not have to be
  subjected to the full range of common
  carrier/telecom regulation in order to require VoIP
  providers to contribute to the USF.

• Guiding Principle #2 – Revenue Neutrality. Any
  extension of USF obligations to VoIP providers (or
  others) should not constitute new/additional revenue
  or a new/additional tax. Rather, it should reflect a
  reallocation of a burden amongst some group of
  similarly-situated competitors.

• Guiding Principle #3 – Regulatory Parity. Those who
  make contributions ought to be considered for
  distributions.
                                                         91
           Challenges for Regulators
• Understand that the rules addressing the
  combination of established networks & regional
  monopolies are not suited to (and not intended to
  govern) emerging technologies.

• Where a competitively available service may
  substitute for basic local exchange service, resist
  the urge to regulate the new service like the old and
  consider deregulating the old.

• Resist the notion that regulators can “create”
  competition and innovation between market
  participants if they just keep tweaking the model.

• In short…let the market work.
                                                          94
             Moving into the IP World
               (Image and Reality)


M.V. Pitke

IEEE Mumbai Section
IEEE ComSoc Chapter

July 28 2001 Ahmedabad
July 29 2001 Vadodara




                                        95
             I Shall Attempt to Cover


•   Communication Networks
•   Narrow Band and Broadband Communications
•   Internet and the Web
•   Transporting Services to the Web
•   Issues in Promises and Delivery




                                               96
         The Telecommunication Network

       PSTN - Public Switched Telephone Network
 Circuit Switched
 Global Coverage
 High Standardization ( FCC and ITU )
 Highest Reliability
 Rapid transition to an all digital network
 Guaranteed Performance
    Bandwidth / bit rate
    Distortion / bit error rates
    Telephony
    Digital Terminals with modems
    Traditionally analog
                                                  97
              Transmission
To the subscriber: 300-3300 Hz bandwidth
                   ~ 40 db Dynamic range
                         in the base-band

In the network: Multiplexed transmission
                   - openwire
                   - coaxial
                   - microwave
                   - optical fibres
       Analog - ~ 25 KHz/channel
       Digital - ~ 32/64 Kbps/channel
Multiplexed systems to more than 10 Gps

                                            98
                      Switching

Blocking and Non-blocking switches
Analog - Strowger (Relays)
       - Crossbar (Relays)
       - Semiconductor (Cross points)
Digital - Time Switch
        - Store and forward bits in a memory
        - Address corresponds to the time position of
          a bit in the sequence
        - Space Switch
        - Interconnection through gates
Both have limitations.
Large digital switches are built by a combination of time
and space switches. Distributed and Centralized
Control
                                                            99
                    Signalling
The most important function in the network
   In-band
   Out-of-band
   Channel associated
   Common channel e.g. SS7
   Subscriber (loop)
   Inter-office signalling
   Dial pulses (10 pps)
   Tones (DTMF)
   MF signalling
   SS7 (Computer Channel)0

                                             100
                  Special Features


   Power Supply for the exchange: -48V
   ~ 1500  loop resistance
   Balanced transmission (300 - 3000 Hz)
   G3 Fax 9600 bps (or higher)
   Transmission of signalling and administrative
    information
   Half hour down time in 25 years
   Modification/upgradation in hot condition (switch)
   Extensive use of duplication/N+1 redundancy
   Smooth movement to ISDN


                                                         101
          Impact of Digital Technology

 Digital Transmission
             - 64 Kbps per voice channel
           - order 32 ch/2 Mbps EI and higher

 Digital Switching
            - Large, fully non-blocking switches
            - ~ 100,000 lines
 DSL, broadband on standard telephone cable pair
 Microprocessors/distributed control
 Dominant role for software
 Ease of development and manufacture


                                                    102
           Impact of Digital Technology


 Powerful common channel signalling systems (SS7)
 Integration of telephone network with computer
  networks
 Overcome geographical limits
 Emergence of novel, innovative value-added services
 Opportunities for new developments and entrepreneurs
 Evolution of radically new, unforseen developments
  like the internet.




                                                    103
                 Advantages of IN

 Ability to rapidly prototype, test and introduce new
  facilities
 Creation of Virtual Private Networks
 Lot of control delegated to the customer (from the
  operator)
 Improved utilization of resources (e.g. 800 type
  services)
 A switch becomes an intelligent peripheral like a voice
  mail server
 Ability to use products from multiple vendors
 Improved overall management and control


                                                        104
                    ISDN and ATM

ISDN - Integrated Services Digital Network
• 2 64 Kbps + 1 8 Kbps channels
• PCM based, circuit switched
• `same’ copper pair

ATM - Asynchronous Transfer Mode
• Packet based for high data rates - 155 Mbps
• Bandwidth (bit rate) selectable as per customer
  requirement
• Designed by telecom engineers
• Same very high quality of service as telecom
                                                    105
                Emergence of New
             Technologies and Services

 Computer/data networks are dominating and
  influencing the evolution of new networks and
  services
 Paging
 Cellular/Mobile
 Broadband
 Cable Telephony
 The Internet – reversal of the trend – voice in packets
  over data network



                                                            106
              Telecom v/s Computer


 Cultural difference
 Telecommunication services result from long
  discussions, committee reports, slow but long-term
  impact. Not agile to catch up with rapid advances in
  technology. Standards come first, services come
  later. Very high availability, no downtime tolerated.
  Dominated by large organisations. Minor role for
  small groups and individuals.
 Relatively large investments



                                                          107
                Computer Approach

 Tolerance for occasional breakdowns
 Service first and standards (if any) emerge later if the
  `service’ gets established
 Ability to adopt and absorb a wide range of new
  techniques and technologies
 Scope for local innovation
 Opportunities for smaller groups and individuals
 Lower costs and investments




                                                         108
           A New Challenge - Cable TV

 Possible advantage for multimedia/video based
  services
 In many areas it has reached the user ahead of the
  telephone
 Presently the cost of cable modem is high, but this
  may change in the near future
 Integrating computer networks with TV, Scope for
  new, innovative services




                                                        109
               Wireless Technology


 Rebirth of wireless technology as a result of
     Low cost access
     Cellular and mobile communication
     Satellite communication
            VSATs
            LEOs
            Mobile services
 Emerging personal communication systems



                                                  110
           Analog Access Technology


 NMT, TACS/ETACS, AMPS
 FDMA – Frequency Division multiple Access
 400 – 900 MHz frequency range.
 Requirements differ in each country – hence
  widely differing specs
 AMPS, the North American standard well
  established
     Meets well the wireline specs, hence well
        suited for Wireless Local Loop applications



                                                      111
            Digital Access Technology

 Based on TDMA – Time Division Multiple Access and
  CDMA
    Code Division Multiple Access or a combination of
     both
 TDMA – channels assigned to a chain of time slots
 CDMA – channels assigned to a series of
  pseudorandom codes spread over a wider band of
  frequency
 DAMPS – TDMA mostly in North America
 GSM – TDMA – International roaming. Expanding
  rapidly all over the world.
    16 Kbps – 13 Kbps for speech and 3 Kbps for
     control
                                                         112
             Cellular Communications
 Mobile communication services over a wide area
  implemented by :
    - Dividing the area into cells with base stations
    - Base stations (MTSO) networked for
       communicating with the mobile subscriber and the
       PSTN
    - A signalling system with protocols to provide
       uninterrupted communication as the subscriber
       moves from cell to cell
 System optimized for size of cell, frequency band,
  number of users in the cell, frequency reuse and grade
  of service
 A separate operation and management system for
  administration, billing, etc.
                                                      113
     Personal Communication Service (PCS)
           Personal phone/pocket communicator


 Upgradation of the cellular phone
 Two approaches
 International Mobile Telecommunication in the
  year 2000 CIMT-2000
 Universal Mobile Telecommunication System
  UMTS (European Initiative)
 Moving to higher data rates
     2 Mbps for low mobility users
 New `Palm’ based solutions
                                                114
          Teleservices in UMTS/3G

 Telephony               8-32 kbps
 Voice Mail              32 kbps
 Video                   128 kbps
 Video Conferencing      384-768 kbps
 Database access         2.4-768 kbps
 Message broadcast       2.4 kbps
 Unrestricted Digital
  Information             64-1920 kbp
 Navigation              2.4-64 kbps
 Location                2.4-64 kbps



                                         115
              Emergence of the IP
 Email Service
 Impact of the World Wide Web - WWW
 Started as an academic network – has
  transformed the telecom industry
 Internet is now what radio was with hams
 Presently piggy-back on PSTN, but will soon
  grow into large networks on its own
 Very low setup and operating cost
 Possible combination with PSTN will provide
  new, low-cost services
 Basically message oriented medium, hence
  problems for voice telephony. Speech tolerates
  distortion, but no delay. Data communication
  tolerates delays but no distortion. (bit errors)
                                                     116
                   New Scenario

 Bits    -     old analog telephony
 Packets -     The Internet and Messaging Systems
 Streams -     Video/Multimedia streaming and
                distribution

 As speeds go up and the medium becomes reliable, it
 becomes difficult or even impossible to use
 conventional protocols. Problems with real time
 processing and standardization.



                                                       117
              ….New Scenario


 Widespread deployment of fibers -
  bandwidth getting cheaper and cheaper
 Problems with the last mile
 More and more services on the wireless - with
  limited spectrum, new techniques for
  sharing/conserving bandwidth
   - WAP
   - Bluetooth/ Home R.F.
   - 3G Services



                                                  118
                Advantages of IP

• Universal Connectivity across Space and Time
• Ability to integrate very diverse products,
  technologies and services
• Ability to work with all types of media
• Information communication of all kinds - one-to-one,
  one-to-many, broadcast, text, audio, visual, moving
  images, different bandwidths, on-line, etc.




                                                         119
                   The Internet

 Presently based essentially on the public telecom
  network (Narrow band)
 Broadband Internet is still evolving
 Many attempts to provide some broadband facilities
  using existing infrastructure
 No regulation. Very poor quality of service (Q0S)
 Present networks cannot automatically get upgraded
  to broadband, just by increasing the speeds
 Problems for telephony on internet
 Many services are getting web based


                                                       120
     Architecture of a Broadband Network


Basic Components
 Servers (large number, different functions)
 Special server for multimedia
 Routers and switches
 Primary/Core network (very high bandwidth)
 Secondary/auxiliary ring
 Access network - fiber, copper, wireless, etc.




                                                   121
The Powersurfer Network




                        PoP/Receiving station

                             Substation



    Clusters of buildings


                                                122
                  Primary Core


 Very high bandwidth (to terabits/sec)
 Basic fiber network with capacity expansion with
  DWDM - Dense Wave Division Multiplexing, etc.
 Access network - access to customers generally
  through a secondary network/ring
 Duplication/Redundancy for full availability
 Routers/Switches for connecting secondary rings,
  etc.



                                                     123
                Secondary Rings

 Buffer between the core and the access serving
  customers
 High traffic (up to Gbps) depending upon the needs
 Essentially provides a concentration function for
  traffic from customers
 Capable of serving different types of access
  technologies and grades of service




                                                       124
                 Access Network

 Provides access to the customer - the last mile
 Accounts for a significant portion (2/3 to 3/4) of the
  total cost
 Different types of technologies
   Fibre Copper - DSL
   Wireless - P - MP
   Other
 Handles wide range of speeds and quality of service




                                                           125
              Multimedia Service

 Very unique problems
 Needed streaming: bits - packets - streams
 Special multimedia servers
 Special software
 Problems relating to generation, caching and
  distribution of content
 Multicasting - network support




                                                 126
        Special Problems of Telephony

 Speech does not tolerate delay but can tolerate
  distortion (bit errors)
 Always interactive
 Data does not tolerate any distortion (bit errors) but
  can tolerate delay - Message oriented
 Speech is relatively narrow band
 Data generally at high speeds
 In a mixed packet environment speech packets have
  to be given priority over data packets. VoIP
  algorithms, etc
 Special `tricks’ to minimize internal delay for speech
  packets
                                                           127
       Special Problems of Video/Multimedia

 Very high data rates
 Very high volume of data
 Protocols for packets are not suitable - too much of
  overhead
 Speeds too high for protocol processing by software
 All processing required to be done mainly by
  hardware
 Special care for handling multicasting functions
  (minimize repetitive traffic)



                                                         128
           Quality of Service (QoS)


 Very high quality standards set by Telecom
 Normally Internet communication being message
  based, the QoS issues have been relaxed
 The speech enforces one of the parameters (delay)
 ‘Carrier class’ availability 99.99% ?
 MTTR and MRBF




                                                      129
                  Important Factors


   QoS - carrier class
   Security - network, access and intrusion detection
   MTBF and MTTR
   Total Transit Delay
   Delay v/s loading
   NMR and billing and customer care




                                                         130
            Important Factors, contd...

 Multimedia streaming and multicasting
 Bandwidth management
 Smooth incorporation of telephony and other
  services
 Handle different interfaces
 Matching investment with customer base expansion
 Crisis handling. Graceful degradation of service - no
  crashes
 Protection against vandalism



                                                          131
                 General Trend


• Cheaper Bandwidth - cost of transmission
• Disappearance of conventional switching
• Higher and higher speeds
• Speed limitation of software
   moving software operations to hardware
• Bits, bytes, packets and now, streams




                                             132
                   Problem Areas
•    Clash of cultures
     Telecom - conservatives
     Computer/IT - radicals
•    Management domination
     Telecom - engineers
     TV/Cable – media
     IT - management (finance)
•   More and more complex products and services:
    less and less time to develop, prototype or even
    test
•   Unreal customer expectation
•   Large gaps between claims and delivery
•   Wide scope for bogus products and fraud1           133
             Another Important Problem


•   Products first, standards (if any) later
•   Compatibility
•   A product has to be `born’ defect free
•   Not possible to start with an unreliable product and
    hope to make it reliable by additions, upgradations
    and patches/




                                                           134
               Present Scenario

• Novel IP and Webcentric services
• Custom tailored services - bandwidth, special
  networks, VPNS, etc.
• Problems moving from the present TDM world to
  IP world
• Urgent need for protocols and signalling systems
  for high speed, high quality networks
• Unreal customer expectations, result of dotcom
  hype and misinformation
• IP has become a new religion. It is perceived as a
  solution to every problem

                                                       135
            Important Issues to Focus



• Quality of Service
   (Telecom or Carrier Class)
• Bandwidth Management
• Delay Management
• Reliability and Availability




                                        136
            Result of these problems


• The vast potential of the Internet is yet to be fully
  tapped
• Broadband services are still on a very small scale.
  Very few houses have real broadband connectivity
• Quality-of-Service issues are not yet solved. We
  really need the ATM broadband (telecom) quality on
  broadband IP networks
• Difficulty in verifying claims of service quality and
  performance



                                                          137
                 In Conclusion

• IP provides an unprecedented opportunity for new,
  innovative, cost effective communication service
  that can have significant impact on our lives
• India should benefit from the low cost of hardware
  and the high software and labour content in the new
  worldwide business
• There is considerable scope to develop
  products/solutions that are important to us here and
  also have a vast market




                                                         138
Voice over IP Protocols


      An Overview




                          139
                 What is in this Module
Module Title:
Voice over IP Protocol – An Overview
Objectives:
This module provides an introductory overview of the
voice over IP protocols: SIP, H.323 and MGCP. At the end
of this module, you will:
• Understand the basics of SIP and its architecture.
• Understand H.323 and how it compares to SIP.
• Understand MGCP.

Target Audience:
Marketing or business development professional who would
like an introductory yet technical overview of the voice over IP
protocols.


                                                                   140
Voice over IP Protocols


   Pictorial Overview




                          141
                         SIP, H.323 and MGCP

 Call Control and Signaling                 Signaling and                  Media
                                           Gateway Control
                                                                            Audio/
           H.323                                                            Video

               H.225

 H.245    Q.931        RAS       SIP               MGCP              RTP   RTCP      RTSP




         TCP                                            UDP


                                             IP



H.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP.
H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP.
SIP supports TCP and UDP.


                                                                                            142
Session Initiation Protocol




                              143
                 What is SIP?


“
Session Initiation Protocol - An application
layer signaling protocol that defines initiation,
modification and termination of interactive,
multimedia communication sessions between
users.


                  IETF RFC 2543 Session Initiation Protocol
                                                              ”
                                                              144
              SIP Framework

– Session initiation.
– Multiple users.
– Interactive multimedia
  applications.




                              145
             SIP Distributed Architecture
               SIP Components




             Location   Redirect       Registrar
              Server     Server         Server




                                                             PSTN

User Agent                                         Gateway
               Proxy               Proxy
               Server              Server


                                                                    146
                       User Agents

• An application that initiates, receives and terminates
  calls.
   – User Agent Clients (UAC) – An entity that initiates a call.
   – User Agent Server (UAS) – An entity that receives a call.

    Both UAC and UAS can terminate a call.




                                                                   147
                Proxy Server

– An intermediary program that acts as both a
  server and a client to make requests on behalf of
  other clients.
– Requests are serviced internally or by passing
  them on, possibly after translation, to other
  servers.
– Interprets, rewrites or translates a request
  message before forwarding it.




                                                      148
             Location Server

– A location server is used by a SIP redirect or
  proxy server to obtain information about a called
  party’s possible location(s).




                                                      149
                Redirect Server

– A server that accepts a SIP request, maps the
  address into zero or more new addresses and
  returns these addresses to the client.
– Unlike a proxy server, the redirect server does not
  initiate its own SIP request.
– Unlike a user agent server, the redirect server
  does not accept or terminate calls.




                                                        150
             Registrar Server

– A server that accepts REGISTER requests.
– The register server may support authentication.
– A registrar server is typically co-located with a
  proxy or redirect server and may offer location
  services.




                                                      151
          SIP Messages – Methods and Responses
SIP components communicate by exchanging SIP
messages:
•   SIP Methods:                           •   SIP Responses:
     – INVITE – Initiates a call by             – 1xx - Informational Messages.
        inviting user to participate in
                                                – 2xx - Successful Responses.
        session.
     – ACK - Confirms that the client           – 3xx - Redirection Responses.
        has received a final response to        – 4xx - Request Failure
        an INVITE request.                         Responses.
     – BYE - Indicates termination of           – 5xx - Server Failure Responses.
        the call.                               – 6xx - Global Failures
     – CANCEL - Cancels a pending                  Responses.
        request.
     – REGISTER – Registers the user
        agent.
     – OPTIONS – Used to query the
        capabilities of a server.
     – INFO – Used to carry out-of-
        bound information, such as
        DTMF digits.                                                           152
                     SIP Headers
– SIP borrows much of the syntax and semantics from HTTP.
– A SIP messages looks like an HTTP message – message
  formatting, header and MIME support.
– An example SIP header:
     ----------------------------------------------------------------
       -
                             SIP Header
     ----------------------------------------------------------------
       -
     INVITE sip:5120@192.168.36.180 SIP/2.0
     Via: SIP/2.0/UDP 192.168.6.21:5060
     From: sip:5121@192.168.6.21
     To: <sip:5120@192.168.36.180>
     Call-ID: c2943000-e0563-2a1ce-2e323931@192.168.6.21
     CSeq: 100 INVITE
     Expires: 180
     User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled
     Accept: application/sdp
     Contact: sip:5121@192.168.6.21:5060
     Content-Type: application/sdp
                                                                        153
               SIP Addressing

– The SIP address is identified by a SIP URL, in the
  format: user@host.
– Examples of SIP URLs:
   • sip:hostname@vovida.org
   • sip:hostname@192.168.10.1
   • sip:14083831088@vovida.org




                                                       154
    Process for Establishing Communication

• Establishing communication using SIP usually
   occurs in six steps:
  1. Registering, initiating and locating the user.
  2. Determine the media to use – involves delivering a
     description of the session that the user is invited
     to.
  3. Determine the willingness of the called party to
     communicate – the called party must send a
     response message to indicate willingness to
     communicate – accept or reject.
  4. Call setup.
  5. Call modification or handling – example, call
     transfer (optional).
  6. Call termination.                                     155
                      Registration
– Each time a user turns on
  the SIP user client (SIP IP
  Phone, PC, or other SIP
  device), the client registers
  with the proxy/registration
  server.                                                Proxy/                    Location/
                                  SIP Phone
– Registration can also occur        User                Registration              Redirect
  when the SIP user client                               Server                     Server
  needs to inform the                         REGISTER                  REGISTER

  proxy/registration server of                  200                       200
  its location.
– The registration information
  is periodically refreshed and         SIP Messages:
  each user client must re-             REGISTER – Registers the address listed in the To
  register with the                     header field.
                                        200 – OK.
  proxy/registration server.
– Typically the
  proxy/registration server
  will forward this information
  to be saved in the
  location/redirect server.                                                                 156
                Simplified SIP Call Setup and Teardown

        User Agent              Proxy             Location/Redirect               Proxy Server        User Agent
                     INVITE     Server            Server
                                               INVITE
                                                302
                                         (Moved Temporarily)
                                                ACK
                                                          INVITE
Call                                                                  INVITE
Setup                                                                   302
                                                                 (Moved Temporarily)
                                                                        ACK
                                                                                            INVITE
                180 (Ringing)                         180 (Ringing)                      180 (Ringing)
                  200 (OK)                              200 (OK)                           200 (OK)
                    ACK                                    ACK                               ACK

Media
                                                   RTP MEDIA PATH
Path
Call                  BYE                                  BYE                               BYE
Teardown             200 (OK)                            200 (OK)                          200 (OK)




                                                                                                                   157
            SIP – Design Framework

• SIP was designed for:
   – Integration with existing IETF protocols.
   – Scalability and simplicity.
   – Mobility.
   – Easy feature and service creation.




                                                 158
       Integration with IETF Protocols (1)

• Other IETF protocol standards can be used to build a
  SIP based application. SIP can works with existing
  IETF protocols, for example:
   – RSVP - to reserve network resources.
   – RTP Real Time Protocol -to transport real time
     data and provide QOS feedback.
   – RTSP Real Time Streaming Protocol - for
     controlling delivery of streaming media.
   – SAP Session Advertisement Protocol - for
     advertising multimedia session via multicast.


                                                         159
      Integration with IETF Protocols (2)

– SDP Session Description Protocol – for
  describing multimedia sessions.
– MIME – Multipurpose Internet Mail Extension –
  defacto standard for describing content on the
  Internet.
– HTTP – Hypertext Transfer Protocol - HTTP is the
  standard protocol used for serving web pages
  over the Internet.
– COPS – Common Open Policy Service.
– OSP – Open Settlement Protocol.


                                                     160
                 Scalability

– The SIP architecture is scalable, flexible and
  distributed.
   • Functionality such as proxying, redirection,
     location, or registration can reside in different
     physical servers.
   • Distributed functionality allows new processes
     to be added without affecting other
     components.




                                                         161
                Simplicity
SIP is designed to be:
– “Fast and simple in the core.”
– “Smarter with less volume at the edge.”
– Text based for easy implementation and
  debugging.




                                            162
                   Mobility

– SIP supports user mobility by proxying and
  redirecting requests to a user’s current location.
– The user can be using a PC at work, PC at home,
  wireless phone, IP phone, or regular phone.
– The user must register their current location.
– The proxy server will forward calls to the user’s
  current location.
– Example mobility applications include presence
  and call forking.




                                                       163
                  Feature Creation

• A SIP based system can support rapid feature and
  service creations.
• For example, features and services can be created
  using:
   – Call Processing Language (CPL).
   – Common Gateway Interface (CGI).




                                                      164
              Feature Creation (2)

• SIP can support these features and applications:
   – Basic call features (call waiting, call forwarding,
     call blocking etc.).
   – Unified messaging.
   – Call forking.
   – Click to talk.
   – Presence.
   – Instant messaging.
   – Find me / Follow me.



                                                           165
                    References

• For more information on SIP refer to:
• IETF
   – http://www.ietf.org/html.charters/sip-charter.html
• Henning Schulzrinne's SIP page
   – http://www.cs.columbia.edu/~hgs/sip/




                                                          166
H.323




        167
              What is H.323?


“
Describes terminals and other entities that
provide multimedia communications services
over Packet Based Networks (PBN) which may
not provide a guaranteed Quality of Service.
H.323 entities may provide real-time audio,
video and/or data communications.

                  ITU-T Recommendation H.323 Version 4
                                                         ”
                                                         168
              H.323 Framework

• H.323 defines:
   – Call establishment and teardown.
   – Audio visual or multimedia conferencing.




                                                169
                 H.323 Components



           Gatekeeper               Multipoint
                                   Control Unit




                   Packet Based                             Circuit Switched
                        Networks                               Networks
Terminal                                          Gateway




                                                                               170
                    H.323 Terminals

• H.323 terminals are client endpoints that must
  support:
   –   H.225 call control signaling.
   –   H.245 control channel signaling.
   –   RTP/RTCP protocols for media packets.
   –   Audio codecs.

    Video codecs support is optional.




                                                   171
                H.323 Gateway

• A gateway provides translation:
   – For example, a gateway can provide translation
     between entities in a packet switched network
     (example, IP network) and circuit switched
     network (example, PSTN network).
   – Gateways can also provide transmission formats
     translation, communication procedures
     translation, H.323 and non-H.323 endpoints
     translations or codec translation.




                                                      172
                    H.323 Gatekeepers
• Gatekeepers provide these functions:
   –   Address translation.
   –   Admission control.
   –   Bandwidth control.
   –   Zone management.
   –   Call control signaling (optional).
   –   Call authorization (optional).
   –   Bandwidth management (optional).
   –   Call management (optional).
• Gatekeepers are optional but if present in a H.323
  system, all H.323 endpoints must register with the
  gatekeeper and receive permission before making a
  call.

                                                       173
            H.323 Multipoint Control Unit

• MCU provide support for conferences of three or
  more endpoints.
• An MCU consist of:
   – Multipoint Controller (MC) – provides control functions.
   – Multipoint Processor (MP) – receives and processes audio, video
     and/or data streams.




                                                                       174
              H.323 is an “Umbrella” Specification
                                                                    H.323
Media                                                Media          Data/Fax     Call Control and
H.261 and H.263 – Video codecs.                                                     Signaling
G.711, G.723, G.729 – Audio codecs.
RTP/RTCP – Media.                            Audio
                                                     Video
                                             Codec
Data/Fax                                     G.711
                                                     Codec
                                                     H.261                        H.225 H.225
T.120 – Data conferencing.                   G.723
                                                     H.263
                                                             RTCP   T.120 T.38
                                                                                  Q.931   RAS
                                                                                                H.245
                                             G.729
T.38 – Fax.

•   Call Control and Signaling                   RTP
•   H.245 - Capabilities advertisement,
    media channel establishment, and                  UDP              TCP        TCP     UDP   TCP
    conference control.
•   H.225                                                               IP

•   Q.931 - call signaling and call setup.
•   RAS - registration and other
    admission control with a gatekeeper.
                                                                                                        175
          Other ITU H. Recommendation that work with
                            H.323

   Protocol                                  Description

H.235          Specifies security and encryption for H.323 and H.245 based terminals.
H.450.N        H.450.1 specifies framework for supplementary services. H.450.N
               recommendation specifies supplementary services such as call transfer,
               call diversion, call hold, call park, call waiting, message waiting
               indication, name identification, call completion, call offer, and call
               intrusion.
H.246          Specifies internetworking of H Series terminals with circuit switched
               terminals.




                                                                                        176
                     H.323 Components and Signaling
    H.225/RAS messages                                           H.225/RAS messages
    over RAS channel                                             over RAS channel


    H.225/Q.931 (optional)              Gatekeeper                H.225/Q.931 (optional)

    H.245 messages (optional)                                     H.245 messages (optional)


                                H.225/Q.931 messages over
                                call signaling channel
                                                                                 PSTN
                                H.245 messages over
         Terminal               call control channel             Gateway


•     H.245 – A protocol for capabilities advertisement, media channel establishment and
      conference control.
•     H.225 - Call Control.
•     - Q.931 – A protocol for call control and call setup.
•     - RAS – Registration, admission and status protocol used for communicating between an
      H.323 endpoint and a gatekeeper.

                                                                                              177
    Process for Establishing Communication

• Establishing communication using H.323 may
  occurs in five steps:
  1. Call setup.
  2. Initial communication and capabilities exchange.
  3. Audio/video communication establishment.
  4. Call services.
  5. Call termination.




                                                        178
                  Simplified H.323 Call Setup

–   Both endpoints have previously
    registered with the gatekeeper.       Terminal A             Gatekeeper        Terminal B
–   Terminal A initiate the call to the                1. ARQ
    gatekeeper. (RAS messages are
                                                       2. ACF
    exchanged).
–   The gatekeeper provides                                   3. SETUP
    information for Terminal A to                          4. Call Proceeding
    contact Terminal B.                                                   5. ARQ
–   Terminal A sends a SETUP                                              6. ACF
    message to Terminal B.                                    7.Alerting
–   Terminal B responds with a Call                           8.Connect
    Proceeding message and also                            H.245 Messages
    contacts the gatekeeper for
                                                           RTP Media Path
    permission.
–   Terminal B sends a Alerting and                             RAS messages
    Connect message.                                            Call Signaling Messages
–   Terminal B and A exchange H.245         Note: This diagram only illustrates a simple
    messages to determine master            point-to-point call setup where call signaling
    slave, terminal capabilities, and       is not routed to the gatekeeper. Refer to the
    open logical channels.                  H.323 recommendation for more call setup
–   The two terminals establish RTP         scenarios.
    media paths.
                                                                                                179
                       Versions of H.323
      Version          Date             Reference for key feature summary

H.323 Version 1   May 1996         New release. Refer to the specification.
                                   http://www.packetizer.com/iptel/h323/

H.323 Version 2   January 1998     http://www.packetizer.com/iptel/h323/whatsnew
                                   _v2.html

H.323 Version 3   September 1999   http://www.packetizer.com/iptel/h323/whatsnew
                                   _v3.html

H.323 Version 4   November 2000    http://www.packetizer.com/iptel/h323/whatsnew
                                   _v4.html




                                                                                   180
                            References

• For more information on H.323 refer to:
• ITU-T
   – http://www.itu.int/itudoc/itu-t/rec/index.html
• Packetizer
   – http://www.packetizer.com/iptel/h323/
• Open H.323
   – http://www.openH323.org




                                                      181
Comparing


SIP and H.323




                182
  Comparing SIP and H.323 - Similarities

Functionally, SIP and H.323 are similar. Both SIP
  and H.323 provide:
– Call control, call setup and teardown.
– Basic call features such as call waiting, call hold,
  call transfer, call forwarding, call return, call
  identification, or call park.
– Capabilities exchange.




                                                         183
 Comparing SIP and H.323 - Strengths

– H.323 – Defines sophisticated multimedia
  conferencing. H.323 multimedia conferencing can
  support applications such as whiteboarding, data
  collaboration, or video conferencing.
– SIP – Supports flexible and intuitive feature
  creation with SIP using SIP-CGI (SIP-Common
  Gateway Interface) and CPL (Call Processing
  Language).
– SIP – Third party call control is currently only
  available in SIP. Work is in progress to add this
  functionality to H.323.

                                                      184
                     Table 1 - SIP and H.323
                                   SIP                                  H.323

Standards Body     IETF.                                ITU.


Relationship       Peer-to-Peer.                        Peer-to-Peer.
Origins            Internet based and web centric.      Telephony based. Borrows call
                   Borrows syntax and messages          signaling protocol from ISDN
                   from HTTP.                           Q.SIG.
Client             Intelligent user agents.             Intelligent H.323 terminals.
Core servers       SIP proxy, redirect, location, and   H.323 Gatekeeper.
                   registration servers.
Current            Interoperability testing between     Widespread.
Deployment         various vendor’s products is
                   ongoing at SIP bakeoffs.
                   SIP is gaining interest.
Interoperability   IMTC sponsors interoperability events among SIP, H.323, and MGCP.
                   For more information, visit: http://www.imtc.org/


                                                                                        185
                  Table 2 - SIP and H.323
 Information                    SIP                                H.323

Capabilities    SIP uses SDP protocol for            Supported by H.245 protocol.
Exchange        capabilities exchange. SIP does      H.245 provides structure for
                not provide as extensive             detailed and precise information
                capabilities exchange as H.323.      on terminal capabilities.

Control         Text based UTF-8 encoding.           Binary ASN.1 PER encoding.
Channel
Encoding Type
Server          Stateless or stateful.               Version 1 or 2 – Stateful.
Processing                                           Version 3 or 4 – Stateless or
                                                     stateful.
Quality of      SIP relies on other protocols such   Bandwidth management/control
Service         as RSVP, COPS, OSP to                and admission control is managed
                implement or enforce quality of      by the H.323 gatekeeper.
                service.                             The H323 specification
                                                     recommends using RSVP for
                                                     resource reservation.


                                                                                        186
                Table 3 - SIP and H.323
 Information                  SIP                              H.323

Security       Registration - User agent         Registration - If a gatekeeper is
               registers with a proxy server.    present, endpoints register and
                                                 request admission with the
               Authentication - User agent
                                                 gatekeeper.
               authentication uses HTTP
               digest or basic authentication.   Authentication and Encryption -
                                                 H.235 provides recommendations
               Encryption - The SIP RFC          for authentication and encryption
               defines three methods of          in H.323 systems.
               encryption for data privacy.
Endpoint       Uses SIP URL for addressing.      Uses E.164 or H323ID alias and a
Location and   Redirect or location servers      address mapping mechanism if
Call Routing   provide routing information.      gatekeepers are present in the
                                                 H.323 system.
                                                 Gatekeeper provides routing
                                                 information.



                                                                                     187
                        Table 4 – SIP and H.323
 Information                          SIP                                    H.323

Features             Basic call features.                    Basic call features.

Conferencing         Basic conferencing without              Comprehensive audiovisual
                     conference or floor control.            conferencing support.
                                                             Data conferencing or
                                                             collaboration defined by T.120
                                                             specification.

Service or           Supports flexible and intuitive         H.450.1 defines a framework for
Feature              feature creation with SIP using         supplementary service creation.
Creation             SIP-CGI and CPL.

                     Some example features include
                     presence, unified messaging, or
                     find me/follow me.

Note: Basic call features include: call hold, call waiting, call transfer, call
forwarding, caller identification, and call park.

                                                                                               188
                   Reference

• This section cites a document that provides a
  comprehensive comparison on H.323 and SIP:
    Dalgic, Ismail. Fang, Hanlin. “Comparison of
    H.323 and SIP for IP Telephony Signaling” in Proc.
    of Photonics East, (Boston, Massachusetts), SPIE,
    Sept. 1999.
    http://www.cs.columbia.edu/~hgs/papers/others/
    Dalg9909_Comparison.pdf




                                                         189
           MGCP


Media Gateway Control Protocol




                                 190
             What is MGCP?


“ Media Gateway Control Protocol - A
   protocol for controlling telephony
   gateways from external call control
   elements called media gateway
   controllers or call agents.

            IETF RFC 2705 Media Gateway Control Protocol
                                                           ”
                                                           191
                       Components
•   Call agent or media gateway
    controller                           Call Agent or             Call Agent or
                                                          SIP
     – Provides call signaling,         Media Gateway
                                                         H.323    Media Gateway
        control and processing            Controller                Controller
                                             (MGC)                     (MGC)
        intelligence to the gateway.
     – Sends and receives
        commands to/from the
        gateway.                               MGCP                    MGCP
•   Gateway
     – Provides translations between
        circuit switched networks and
        packet switched networks.
     – Sends notification to the call   Media Gateway            Media Gateway
        agent about endpoint events.        (MG)                     (MG)
     – Execute commands from the
        call agents.

                                                                                   192
                 Simplified Call Flow
                                                 Call Agent
                                          Media Gateway Controller
–When Phone A goes offhook
 Gateway A sends a signal to
 the call agent.
–Gateway A generates dial
 tone and collects the dialed             MGCP                MGCP
 digits.
–The digits are forwarded to
 the call agent.                                  RTP/RTCP
–The call agent determines
 how to route the call.
–The call agent sends                 Gateway A              Gateway B

 commands to Gateway B.
–Gateway B rings phone B.
                                Analog
–The call agent sends           Phone A
                                                                         Analog
                                                                         Phone B
 commands to both gateways
 to establish RTP/RTCP
 sessions.
                                                                               193
               MGCP Commands

• Call Agent Commands:        • Gateway Commands:
  –   EndpointConfiguration     – Notify
  –   NotificationRequest       – DeleteConnection
  –   CreateConnection          – RestartInProgress
  –   ModifyConnection
  –   DeleteConnection
  –   AuditEndpoint
  –   AuditConnection




                                                      194
           Characteristics of MGCP

• MGCP:
  – A master/slave protocol.
     • Assumes limited intelligence at the edge
       (endpoints) and intelligence at the core (call
       agent).
     • Used between call agents and media gateways.
     • Differs from SIP and H.323 which are peer-to-
       peer protocols.
  – Interoperates with SIP and H.323.



                                                        195
               MGCP, SIP and H.323
– MGCP divides call              In this example, an H.323 gateway is
  setup/control and media        “decomposed” into:
  establishment functions.       –A call agent that provides signaling.
– MGCP does not replace SIP      –A gateway that handles media.
  or H.323. SIP and H.323        MGCP protocol is used to control the
  provide symmetrical or peer-   gateway.
  to-peer call setup/control.     H.323 Gateway
– MGCP interoperates with                             H.323
  H.323 and SIP. For example,
   • A call agent accepts SIP        Call Agent/                 H.323
                                       Media
     or H.323 call setup              Gateway
                                                                Gateway
     requests.                       Controller

   • The call agent uses MGCP                MGCP
     to control the media
     gateway.                                        Media RTP/RTCP
   • The media gateway                 Media
                                      Gateway
     establishes media
     sessions with other H.323
     or SIP endpoints.                                                    196
                           Example Comparison
 H.323                                           MGCP
 1.      A user picks up analog phone and        1.   A user picks up analog phone and dials a
         dials a number.                              number.
 2.      The gateway determines how to           2.   The gateway notifies call agent of the phone
         route the call.                              (endpoint) event.
 3.      The two gateways exchange               3.   The Call agent determines capabilities,
         capabilities information.                    routing information, and issues a command
 4.      The terminating gateway rings the            to the gateways to establish RTP/RTCP
         phone.                                       session with other end.
 5.      The two gateways establish
         RTP/RTCP session with each other.


                       3                                        2    Call Agent/
                     5.RTP/                                            Media
                     RTCP                4                            Gateway
  1
       H.323                   H.323                                 Controller
      Gateway                 Gateway
                                                        1
                                                                        RTP/
                                                            Gateway A   RTCP Gateway B
Analog                                  Analog
Phone                                   Phone          Analog                            Analog
                                                       Phone                             Phone

                                                                                                  197
                 What is Megaco?

• A protocol that is evolving from MGCP and
  developed jointly by ITU and IETF:
   – Megaco - IETF.
   – H.248 or H.GCP - ITU.

• For more information refer to:
   – IETF - http://www.ietf.org/html.charters/megaco-
     charter.html
   – Packetizer - http://www.packetizer.com/iptel/h248/



                                                          198
                        References

• For more information on MGCP refer to:
• IETF
   – http://www.ietf.org/rfc/rfc2705.txt?number=2705




                                                       199
Summary




          200
                  Summary

– SIP and H.323 are comparable protocols that
  provide call setup, call teardown, call control,
  capabilities exchange, and supplementary
  features.
– MGCP is a protocol for controlling media
  gateways from call agents. In a VoIP system,
  MGCP can be used with SIP or H.323. SIP or
  H.323 will provide the call control functionality
  and MGCP can be used to manage media
  establishment in media gateways.



                                                      201
Additional References




                        202
            General VoIP Reference

• Pulver – IP Telephony News
  – http://www.pulver.com
• Internet Telephony
  – http://www.internettelephony.com
• An overview poster of the SIP, MGCP, and H323
  protocols.
  – http://www.protocols.com/voip/posvoip.pdf




                                                  203
                  End of Module


• This is the end of the VoIP Protocol Overview
  training module.
• For additional training and documentation visit us
  at www.vovida.org.




                                                       204
     Did you think VoIP was just old telephony
                somewhat cheaper?

          Not with the IX67!
 Live IP communication is much more.



Get even more with the SIP Switch!




                                                 205
You’ve Got an Account from a SIP Service Provider
                          All SIP compliant clients work! No requirement for
                          clients to implement insecure/unreliable
sip:ben@provider.com
                          UPnP/STUN/ICE to open holes in firewall.
                                                                               Each client has
                                                                               an account


sip:ann@provider.com
                                                        Internet


sip:carl@provider.com

                            But don’t you want global IP-SIP connectivity?
                            And you want to use all the SIP features…
sip:sue@provider.com




 sip:ken@provider.com                      •    Use any services from the provider
                                                on your LAN.
 sip:beth@provider.com                     •    Use multiple SIP clients on your
                                                broadband connection.
 sip:lisa@provider.com                     •    IX67 is SIP transparent by default.
                         Ordinary
                         NAT/firewall
 sip:john@provider.com

                                                                                                 206
                                    Set Up Your Own SIP Server!
                                                                                      A service for your
              sip:ben@smartcompany.com
                                          SIP server for domain
                                          smartcompany.com                            broadband!
                                                                               SIP
                                                                                      Internet is global!
              sip:ann@smartcompany.com
                                                                    Internet


              sip:carl@smartcompany.com                                        Is it really that easy?
                                                         (Dyn)DNS
                                                                               Hmm, email is global – I have it…
              sip:sue@smartcompany.com
                                                                               Why not SIP?
              sip:sue@provider.com

                                                            Become part of the open SIP community!
To do list:
1. Make the IX66 your SIP server – GUI:
                                                            •     Communicate with anyone in the open
                                                                  SIP community.
2. Make your SIP domain DNS point to your IX66.
        Static IP address: Ask your ISP to point out       •     Get also video, presence (client based),
         your SIP domain.                                         instant messaging and data
        Dynamic IP address: Get a DynDNS account                 collaboration for free!
         and use the IX66 DynDNS client.
3. Use your SIP domain names for the clients (see           •     Combine with SIP service provider
   example). You may also mix with service                        accounts.
   provider’s SIP addresses.
                                                                                                            207
                       Just Another Internet Service…
     inGate
    Firewall
                       Sweden
                                             Internet
          Networks               Boston                                                  PSTN
         Telecom                  VON                        Sweden           SIP/PSTN
                                                  ENUM                        Gateway
                              Booth
                              #421

                So,fine - but how do IUSA PSTN connectivity?
                                       get
  +43 1 25397 531     IX67         Sweden                IX67
                And it is soooo difficult to dial a URL,
                even on a SIP phone! IX66
 SOHO LAN
                                                    Intertex Stockholm LAN

                                                    Home Office Users
                         XP
                                                          inGate        DMZ
                                                                                          DNS
+43 1 25397 511 +43 1 25397 521 +43 1 25397 512          SIParator
                                                                                          SRV

      Enterprise LAN                   inGate            Ingate Linköping LAN
                                      Firewall
                          XP
                                                                                XP
+43 1 25397 513     +43 1 25397 522
                                  Get PSTN Connectivity
                            Using a Gateway Account (Alternative 1)
              sip:ben@smartcompany.com

                                                             SIP
              sip:ann@smartcompany.com
                                                          Internet


              sip:carl@smartcompany.com


                                          And how can PSTN users
              sip:sue@smartcompany.com
              sip:sue@provider.com
                                          reach my SIP address?
                                          And nobody wants to give me
                                          a gateway account! 
To do list:
 Find a suitable provider that is
  offering such subscriptions.               •   You get PSTN connectivity through the
 Make sure they have good rates                 service provider’s gateway(s) and at the
  to the places you often dial (i.e.
  gateways at those locations)                   rates they offer.
 Buy the IX66 SIP Switch
  software addition.                         •   All users of the IX67 are known and billed
 Download a Dial Plan and fill in               by the provider.
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                                                                                              209
  Let’s do something about it!




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                                                                      211
 The SIP World with Operator Services…
   ITSP                Internet
              The SIP Switch Integrates:SIP/PSTN
                       Sprint                                 PSTN
                                                 Gateways
IX67      Easy dial from SIP HW Phones
          - to soft from SIP PC clients
          Easy dialSIP PC clients: Now !#*
                           USA
          - to soft SIP phones
          - to SIP HW PC clients
                         Sweden                      IX67
                     from
          Easy dial DNS PSTN
          - to PSTN phones
          - to SIP HW CONNECTIVITY
                     PSTN IX68
          PBX functions…                         Intertex Stockholm LAN
       XP - to PSTN Home Office Users
                     PHONE NUMBERS
                      “extension” numbers
          - internal ENUM features…
          Typical Phone
          - 0 to reach the PSTN
          - voice mail forwarding
                      the MAIL
          And keepVOICE new features…
          - direct PSTN numbers
                                                        DMZ      inGate
          - forwardSIP PRESENCE SERVER
           - Presence                                           SIParator

           - IM inGate      Ingate Linköping LAN
           - Video
                Firewall
         XP Forking – ring several
           -
                                                  XP
                     How can we achieve that?

 To get general SIP traversal
   :                                   Firewall & NAT
 Dynamic Firewall Engine
 SIP Proxy Server, controlling the
  firewall

 SIP Registrar, user location
  information
 Communication between SIP Proxy                       Firewall
  and firewall                                          Control
What have you got?                                      Protocol
In the Ingate and Intertex products:
You got a SIP server!
Use it just for firewall traversal          SIP           User
AND/OR as your                             Proxy        Location
- SIP Server
- Outbound proxy
- Inbound proxy
- PBX (The SIP Swich)                                              213
                          Add PBX Functionality
                                              Locally on your LAN and globally over
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                                              numbers, dial plan, ENUM and more!

                                                      SIP


                                                 Internet




                          •   Mimic conventional PBXs, 0 or 9 to reach the PSTN
                          •   Forward incoming calls individually to one or several
                              phones
To do list:
                          •   Call internally using extension numbers, URLs, or
 Buy the SIP Switch          conventional phone numbers
  software option for     •   Check ENUM before handing call to PSTN
  the IX66.
 Select your dial plan   •   Outgoing PSTN calls routed to best gateway
 Set up the user         •   New: Fork, ENUM, blacklist and whitelist callers
  accounts

                                                                                  214
                                 Get PSTN Connectivity Using
                                Service Provider’s SIP Account
              sip:ben@smartcompany.com

                                                             SIP
              sip:ann@smartcompany.com
                                                          Internet


              sip:carl@smartcompany.com


                                                   • You get PSTN connectivity through
              sip:sue@smartcompany.com
                                                     one or several service provider’s
                                                     SIP user accounts.
To do list:
 Get accounts with suitable SIP providers         • All SIP Switch users can use the
  supported by the IX66 SIP Switch and having
  good PSTN connectivity in the countries you        accounts for ordinary (PSTN)
  communicate with.
                                                     phone calls.
 Buy the IX66 SIP Switch software addition.
 Download a Dial Plan and fill in the SIP
  accounts to be used for different countries or
                                                   • PSTN users can call in locally
  areas.                                             (where you have your SIP
 Allow the SIP Switch to forward the specific
  country user numbers to one or several             accounts) and be routed to any
  SIP and regular phones.                            phone by the SIP Switch.
                                                                                        215
                                      Virtual Foreign Offices
                                                                     Get phone number with PSTN
                                                                     access in various countries!
              sip:ben@smartcompany.com


                                                              SIP                          Sweden
                                                                                           (local access)
              sip:ann@smartcompany.com

                                                          Internet
                                                                                                   USA
              sip:carl@smartcompany.com
                                                                                           (local access)



              sip:sue@smartcompany.com                •    Customers and partners can call a
              sip:sue@provider.com
                                                           local number in the specific country.
To do list:                                                You will get the call via IP for free.
 Get accounts with suitable SIP providers
  supported by the IX66 SIP Switch and having         •    You can call your customers and
  good PSTN connectivity in the countries you              partners, only paying for local
  communicate with.
 Buy the IX66 SIP Switch software addition.
                                                           access. (International part over IP).
 Download a Dial Plan and fill in the SIP accounts   •    Incoming calls can be routed to
  to be used for different countries or areas.
 Allow the SIP Switch to forward the specific
                                                           specific phones or soft clients for
  country user numbers to one or several                   various countries.
  SIP and regular phones.
                                                                                                    216
                                    Lowest Rate PSTN Calls
                                                    Long distance over IP – Old
                                                    telephone network only locally!


              sip:ben@smartcompany.com           SIP                      Sweden
                                                                          (local access)


                                             Internet
              sip:carl@smartcompany.com
                                                                                USA
                                                                          (local access)
• You can lower you phone rates
  radically!
  Not just 30%. We talk up to 90%…                                         Austria
                                                                          (local access)

To do list:
 Get accounts with suitable SIP
  providers supported by the IX66
  SIP Switch and having good
  PSTN connectivity in the
  countries you communicate with.
 Buy the IX66 SIP Switch software
  addition.
 Download a Dial Plan and fill in
  the SIP accounts to be used for
  different countries or areas.
                                                                                   217
             ENUM – Dial Phone Number to SIP Phone

                                                          Bypass the PSTN!
                          3) IX66 uses returned           IP all the way!
                          SIP URL to bypass PSTN

                                                                  Other SIP
                                        Internet                  phone
         2) ENUM lookup




                                                   PSTN


1) Dial Phone Number

• SIP devices can have E.164 phone numbers assigned
  (just like ordinary phones)
• IX67 SIP Switch checks dialled phone numbers with
  EMUM to see if a SIP URL exists
                                                                          218
     Dial Plan with ENUM and Authentication




Mix URLs, “extension” and E.164 numbers conveniently
Mimic PBX, e.g. dial 0 for PSTN (or 9 in US)
ENUM checking before passing to PSTN gateway
                                                       219
                    User Accounts




Extension numbers      Mapping of incoming PSTN calls
Authentication         Using Operator’s SIP account for
                       everyone

                                                          220
     And every user can control his settings




And more will come…

                                               221
          Or the administrator can do it…




Forwarding, Forking, Sequence   Forward to Voice MailZ


                                                         222
            Restriction of Incoming Callers
SPAM calling may become a problem – prepare for it…




Allow callers based on various criteria     Or blacklist unwanted
                                          (Although easy to bypass)


                                                                  223
           Hacking VoIP




Is your Conversation confidential?

    by Nick von Dadelszen and Darren Bilby

                                             224
                 VoIP Trends

• VOIP becoming more popular and will increase in
  future
• Many ISPs and Teleco’s starting to offer VoIP
  services
• Like most other phone calls, it is presumed to be
  confidential




                                                      225
              Types of Phones

• SoftPhone




• HardPhone




                                226
Typical VoIP Architecture




                            227
                    Attacks Against VoIP

• Multiple attack avenues:
   –   Standard traffic capture attacks
   –   Bootp attacks
   –   Phone-based vulnerabilities
   –   Management interface attacks




                                           228
             Consequences of Attacks

• Consequences of VoIP attacks include:
  –   Listening or recording phone calls
  –   Injecting content into phone calls
  –   Spoofing caller ID
  –   Crashing phones
  –   Denying phone service
  –   VoIP Spamming




                                           229
                     VoIP Protocols

• H.323
  – Earlier protocol used, though still used today
  – Provides for encryption and authentication of data
• SIP
  – Digest authentication based on HTTP, but many times not enabled
  – No encryption
• MGCP
  – Relies on IPSEC for security, but most current phones don’t
    support IPSEC




                                                                      230
                Use of VLANS

• Cisco recommends separate VLANs for data and
  voice traffic
• To ease implementation, many phones allow sharing
  of network connections with desktop PCs
• VoIP allows the use of SoftPhones installed on
  desktop PCs
• Therefore cannot separate voice traffic from the rest
  of the network




                                                     231
            Capturing VoIP Data


• Ethereal has built-in support for some VoIP
  protocols
• Has the ability to capture VoIP traffic
• Can dump some forms of VoIP traffic directly to WAV
  files.




                                                   232
                      Other Tools

• Vomit
  – Injects wave files into VoIP conversations
• Tourettes
  – Written by a staff member of a customer for fun
  – Injects random swear words into a conversation




                                                      233
          Example Phone Exploit

• CAN-2002-0769
• Cisco ATA-186 Web interface could reveal sensitive
  information
• Sending a POST request consisting of one byte to
  the HTTP interface of the adapter reveals the full
  configuration of the phone, including administrator
  password

• IP Phones – Another thing to patch!




                                                    234
               Caller ID Spoofing

• Caller ID is based on a Calling Party Number (CPN)
• This is always sent when a call is placed
• A privacy flag tells the receiver whether to show the
  number or not
• Have always been able to spoof Caller ID but needed
  expensive PBX equipment to do so.
• With VoIP PBX software, spoofing is easier
• Has repercussions for phone authentication




                                                          235
                  VoIP System Model

• Two issue from the previous diagram

  – Voice coding: Voice Packet, Packet Voice

  – Signaling: who is called and where is the called party on the
    network?




                                                                    236
            The two models in brief


        H.323                   SIP
Gatekeeper          UAC (user agent Client)
Gateway             UAS (User agent server)
H.323 terminal      SIP Terminal
MCU                 Proxy
                    Redirect Server
                    Location server




                                              237
           Voice packet routing and Delay

• Source of delay include:
   – Accumulation delay: caused by need to collect a frame of voice samples
     to be processed by the voice coder (from some microsecond to many
     milliseconds.
   – Algorithmic Delay: caused by specific voice encoding delay.
   – Processing delay: result from the two previous delay plus collecting the
     sample in to packet for transmission.
   – Network delay: Processing that occurs as packets are sent across a
     network.(from protocol, medium, and buffer use to remove packet
     jitter on the receive side)




                                                                                238
      Voice packet routing and Delay

– Echo: is generated toward the packet network from
  the telephone network…as it always greater than 50
  ms, it is not acceptable for good audition.
– Jitter: is the variable inter-packet timing cause by the
  fact that packets do not all cross the network at the
  same speed.




– Lost packet: Under peak load and congestion, voice
  frame are dropped at the same rate as data frame.


                                                             239
                    VoIP Signaling

• Three distinct area:
   – Signaling from PABX  router
      • Network seize the PABZ with any of the signaling used to seize
        a trunk (as the local network appear to the BABX as a trunk) –
        FXS or E&M.
   – Signaling from Router  router
      • Dial plan Mapper.
   – Signaling from Router PABX
      • Line seizure signaling.




                                                                         240
                VoIP application

• In today’s networking, there are several attractive
  alternatives both to conventional public telephony
  and to leased lines. Among the most interesting are
  networking technologies based on a different kind of
  voice transmission, called packet voice and in our
  case Voice over IP.
• VoIP can be used in two broad context differentiated
  by geography or by the type of users to be served.




                                                         241
                   VoIP application

• Within a national administration or telephony jurisdiction,
  – to support its own voice calling among its own sites.
  – to support the activities of a single company — to
     connect two or more company locations in
     multiple countries —
  – to connect public calls within a company, the packet
     voice provider is technically providing a local or
     national telephone service and is subject to
     regulation as such.




                                                                242
                   VoIP application

• Between different administration or telephony
  jurisdiction
   – to connect public calls between countries, the packet
     voice provider is subject to the national
     regulations in the countries involved and also to
     any treaty provisions for international calling to
     which any of the countries served are signatories.
• Be aware of the valid law applicable in your country
  until setting up VoIP application to avoid any
  inconvenience.



                                                             243
                VoIP application

• Example:
CAFE Informatique & Telecommunications S.A.
   – Two application:
      • International Communication
      • Call center
         • 30 local worker.
         • Tele-marketing for America and Canadian
           company
         • Data scramble for foreign company



                                                     244
         VoIP Technology (CODECs)

• G.711 (PCM) – 64Kbps (4.4 MOS)
• G.721, G.723, G.726 (ADPCM) – 16, 24, 32, 40 Kbps
  (4.2 MOS)
• G.728 (LD-CELP) – 16 Kbps (4.2 MOS)
• G.729, G.729a (CS-ACELP) – 8Kbps (4.2 MOS)
• G.723.1 (CS-ACELP) – 5.3Kbps (3.5 MOS), 6.3Kbps
  (3.98 MOS)




                                                      245
            VoIP Technology (Call Setup)

• H.323 Standard (Video conferencing over
  Packet/Data Networks)
  – H.225
  – H.245
• Real-Time Transport Protocol (RTP)/Real-Time
  Transport Control Protocol (RTCP)
• Session Initiation Protocol (SIP)
• Multimedia Gateway Control Protocol (MGCP)




                                                 246
             VoIP Technology (QoS)

•   Dejitter Buffer
•   Type of Service (TOS)
•   Weighted Fair Queuing (WFQ)
•   Random Early Detection (RED)
•   Weighted RED (WRED)
•   Multilink PPP (MLPPP)
•   Resource Reservation Protocol (RSVP)




                                           247
            Concerns To Be Addressed

•   Powering for End-user Devices
•   Reliability
•   Troubleshooting VoIP Services
•   Access to Emergency Services and related issues
    (Liability)




                                                      248
              What has changed?

• IP Telephones powered by proprietary switching
  devices
• Converters for Current devices
• Some more layers of reliability have been added
• New services added to IP PBXs and gateways




                                                    249
        Things that we are still missing

• Device location. Similar to cell phone triangulation.
  How about GPS?
• Standardization for power provisioning
• Plan for an open environment similar to the current
  data and voice networks
• Management platforms




                                                          250
                 UO VoIP Services

• Equipment:
   – Redundant IP/PBX Call Manager
   – Cisco Digital/Analog Gateways
   – Cisco’s IP Telephones
• Services Offered:
   – VoIP Trunks to Remote Sites
   – On Campus Mobility via IP Telephones




                                            251
        Tracking Technology Changes
• Websites
  – Internet Technical Resources
    (http://www.cs.columbia.edu/~hgs/internet)
  – Hello Direct (http://www.phonezone.com)
  – Internet Telephony
    (http://www.fokus.gmd.de/research/cc/glone/proje
    cts/ipt)
  – Computer Telephony
    (http://www.telecomlibrary.com)
  – European Telecommunications Standards
    Institute (http://www.etsi.org/tiphon)
  – Internet & Telecomms Convergence Consortium
    (http://itel.mit.edu)
                                                       252
        Tracking Technology Changes

• Hardware Manufacturers
   – Cisco Systems (http://www.cisco.com)
   – Lucent Technologies (http://www.lucent.com)
   – 3Com (http://www.3com.com)
   – Nortel Networks (http://www.nortelnetworks.com)
   – Vocaltec (http://www.vocaltec.com)
   – RadVision (http://www.radvision.com)




                                                       253
       Internet and Telecoms Convergence
• PSTN designed for reliable voice
   – Data added by making it behave like voice (modems, ...)
• ISDN designed for reliable data and reliable voice
   – Voice treated as data using CS paradigm (2B+D, ...)
• Internet designed for “best effort” data transfer
   – Pretty good, but good enough?
   – Much effort being applied towards QoS, security/fraud/privacy,
     charging, legacy interworking, ...
• Major changes in access capabilities
   – xDSL; WWANs (UMTS, CDMA, ...) & WLANs (IEEE 802.11x)

No approach fully satisfactory by itself
  – Can be addressed using a “managed” internet
  – “Next Generation Networks” discussions transitioning from
    theoretical to practical

                                                                      254
                       GII is All About Convergence
                     Internet, Broadcasting, Telephony, ...
                                          The NGN 2004 Project will establish
                                          implementation guidelines and standards
     TODAY                                for the realization of Next Generation
                        Consumer
                       Entertainment      Networks based on GII concepts.

                                                        NEAR FUTURE
Telecommunications    GII
                                                                               Consumer
                                                                              Entertainment
                             Computer
                            information
                                              Telecommunications            GII

              AIM
                        Consumer                                                Computer
                       Entertainment                                           information

                          GII
     Telecommunications                    • Y-series Recommendations: GII
                    Computer               • Per Fig. 5-1/Y.110 – GII is at the centre of
                   information               the threefold industry convergence



                                                                                              255
                                  The Wireless Landscape
                         Wireless Wide Area Network
                         (WWAN)
                         Metro/Geographical area
                         “Always On” Services
                         Ubiquitous public connectivity with
                         private virtual networks
                                                                             Wireless Local Area Network
 Mobility                                                                    (WLAN)
                                                                             Public or Private Site or Campus
                                                                             Enterprise / premises application voice &
          Vehicle                                                            data network extension
                    GSM/GPRS
                    CDMA2000 1X




                                                                             Nomadic / “pull” services
Campus
Outside




                                                                             Non-licensed spectrum
           Walk
                                                     >3G

           Fixed
                                                                 HiperLAN2

                        DECT
                                                      802.11b
Campus




                                                                  802.11a



           Walk
Within




           Fixed/      Bluetooth                                             LAN
          Desktop

                           0.1             1                    10            100        Mbps
                                                                                                                         256
          Enhancing End User Experience:
              Blending User Devices

• PC, phone(s) and PDA: different user interfaces to
  the same network-based application
• Common, network-based directory for:
   –   Phone numbers
   –   Buddies & presence
   –   Email address book
   –   All applications
• Just one address to reach the user
• Unified, network-based, user profile applying to all
  terminals
   – E.g., set presence location, call routing preferences, etc., on any
     terminal and it applies to all


                                                                           257
 The Un-Wiring of the Future

              • Mobility / WWAN
              • A Million nodes @
                $50k


       • S                          • Nomadic / Mesh /
         e                            WLAN
         n                          • Millions of Nodes @
         s                            $100
         o
         r

          /

         A
… connected through the Wireless Packet Network
         d
         -                                                  258
                                      Network Transformation

              Gaming
              Console
                                                      Existing
     PDA
                                                      Multiple networks
                         Home                         Simple devices
                        Computer
                                                      Disparate services
Business                Home Office
                          Phone
 Mobile



            Office
                                                      Transition
 Office
 Phone
           Computer
                        TV / PVR                      • Converged packet network
                                                      • Multimedia devices
             Simplify




                                                      • Linked services

                                                      Transformed
                                                      • Dynamic packet/
     Network Profile                                    optical network
                                                      • Secure multimedia services
                                                      • Ubiquitous broadband
                                                      • Integrated functionality

                                                                                   259
                                               Convergence

                                                                                                 Application Servers
This animated chart is
  provided as three
discrete charts at the
 end of this package.



  Infrastructure
                                                    Call Server                                           Call Server
                                                                  PDF
                                       HLR/                             Internet                                          Internet
                                                                        Intranet                                          Intranet
                                       HSS
                         Call Server                                               Call Server
                           MGCF                                                      MGCF
                                              GGSN
     Services               MGW
                                                        PDSN       HA
                                                                         PDG          MGW
                                                                                                           PDSN
                                                                                                                            PDG

                                         PSTN

                                                                                                   PSTN
                                                      SGSN
                               R4 BICN



                                              GSM    UMTS CDMA                                                          WLAN DSL/Cable

  Architectural


                                                                                                                                     260
  Transformed Network Architecture
                                                 ISV
                                                Apps
                                                                    Applications

 Access                             Voice            Media
           Service       Content                             Interactive
                        Switching                            Multimedia    Services
            Edge                            Policy



            Security


            Mobility                                                        Packet

             QoS


            IP VPN
                                                                            Optical

           Subscriber
            Control
LAN
                                    Broadband
                                                                                 261
       Requirements of Service Architecture

• For Users:
   –   Services available everywhere
   –   Choice of services from multiple sources
   –   Performance guarantees / one number to call for support
   –   Immediate activation / one bill to pay
• For Service Providers:
   – Open service creation on one service infrastructure
   – “Stickiness” with Users
   – Performance against SLAs
• For Service Developers:
   – A convenient level of abstraction
• For Service Transporters:
   – A slice of revenue: no free lunch!
• For everyone:
   – Security from malicious attack
                                                                 262
      Key Attributes of a Service Architecture

• Supports dynamic and static services
• Enables access-independent service delivery
• Provides seamless service execution across
  enterprise and carrier domains
• Enables a dynamic communications services value
  chain
• Ensures services are billable
   – revenue essential for the bottom line!
• Supports digital rights management
• Simplifies the end-to-end user experience


                                                    263
                    NGN Standards

• Many organizations working on NGNs, future
  generation technologies, etc.
  – Leverage ITU global perspectives for an overall framework
  – Leverage near term detailed and well-focussed technical work of
    relevant bodies into this consistent global framework
  – Example: ITU-T addressing forward looking global NGN framework,
    3GPPs working IMS, ETSI TISPAN being based on 3GPP IMS Rel. 6,
    OMA working application areas




                                                                      264
The Transformed Network




 • Always on
 • Anytime, anywhere and in any
   form
 • Voice and multimedia
 • Self service, intuitive
 • Simple for the end user
 • Secure, trusted and reliable




                                  265
                       Selected Acronyms

3G        Third Generation                    MGCF   Media Gateway Control Function
3GPP(2)   Third Generation Partnership        MGW    Media Gateway
          Project (2)                         NGN    Next Generation Network
BICN      Bearer Independent Core Network     PC     Personal Computer
CDMA      Code Division multiple Access       PDA    Personal Digital Assistant
CSCF      Call State Control Function         PDF    Packet Data Function
DECT      ?? Digital Electronic Cordless      PDG    Packet Data Gateway
          Telephony                           PDSN   Packet Data Serving Node
FA        Foreign Agent                       POTS   Plain Old Telephone Service
GGSN      Gateway GPRS Support Node           PSTN   Public Switched Telephone
GII       Global Information Infrastructure          Network
GPRS      General Packet Radio Service        QoS    Quality of Service
GSM       Global System for Mobility          SCM    Session Control Manager
HA        Home Agent                          SGSN   Serving GPRS Support Node
HLR       Home Location Register              SIP    Session Initiation Protocol
HSS       Home Subscriber Server              SLA    Service Level Agreement
IMS       IP Multimedia Subsystem             UMTS   Universal Mobile Terrestrial
IP        Internet Protocol                          Access
ISDN      Integrated Services Digital         WLAN   Wireless Local Area Network
          Network                             WWAN   Wireless Wide Area Network
LAN       Local Area Network

                                                                                      266
         Convergence - Non-animated - Step 1



                                           Call Server                                         Call Server
                                                         PDF
                              HLR/                             Internet                                        Internet
                                                               Intranet                                        Intranet
                              HSS
                Call Server                                               Call Server
                  MGCF                                                      MGCF
                                     GGSN                       PDG                                              PDG
                   MGW                                                       MGW
                                               PDSN       HA                                    PDSN
                                PSTN

                                                                                        PSTN
                                             SGSN
                      R4 BICN
Architectural
                                     GSM    UMTS CDMA                                                        WLAN DSL/Cable




                                                                                                                          267
         Convergence - Non-animated - Step 2
                                                                                        Application Servers




                                           Call Server                                           Call Server
                                                         PDF
  Services                    HLR/
                              HSS
                                                               Internet
                                                               Intranet
                                                                                                                 Internet
                                                                                                                 Intranet

                Call Server                                               Call Server
                  MGCF                                                      MGCF
                                     GGSN                       PDG                                                PDG
                   MGW                                                       MGW
                                               PDSN       HA                                      PDSN
                                PSTN

                                                                                          PSTN
                                             SGSN
                      R4 BICN
Architectural
                                     GSM    UMTS CDMA                                                          WLAN DSL/Cable




                                                                                                                            268
           Convergence - Non-animated - Step 3
                                                       Application Servers



Infrastructure                                     Intelligent Infrastructure
                      Packet Based      CSCF/SCM
                                                       PDF
                              HLR/                              Internet
                              HSS                               Intranet
                 Call Server
                   MGCF
                                     GGSN                        PDG
  Services          MGW                       PDSN        HA
                                 PSTN
                                            SGSN
                       R4 BICN


Architectural                     GSM       UMTS CDMA WLAN DSL/Cable

                                        Access Independence


                                                                                269
                     Name confusion

• Commonly used interchangeably:
   – Internet telephony
   – Voice-over-IP (VoIP)
   – IP telephony (IPtel)
• Also: VoP (any of ATM, IP, MPLS)
• Some reserve Internet telephony for transmission
  across the (public) Internet
• Transmission of telephone services over IP-based
  packet switched networks
• Also includes video and other media, not just voice



                                                        270
             New Internet services

• tougher: replacing dedicated electronic media vs.
  new modes (web, email)
• distribution media (radio, TV): hard to beat one
  antenna tower for millions of $30 receivers
• typewriter model of development
• radio, TV, telephone: a (protocol) convergence?




                                                      271
           The phone works – why bother with VoIP

user perspective                         carrier perspective

variable compression: tin can to         better codecs + silence suppression –
broadcast quality  no need for          packet header overhead = maybe
dedicated lines                          reduced bandwidth
security through encryption              shared facilities simplify management,
                                         redundancy

caller & talker identification           advanced services

better user interface (more than 12      cheaper bit switching
keys, visual feedback, semantic rather
than stimulus)
no local access fees (but dropping to    fax as data rather than voiceband data
1c/min for PSTN)                         (14.4 kb/s)

adding video, application sharing is
easy
                                                                                  272
                  Telephone

• 1876 Alexander Graham Bell transfered voice over
  wire for the first time.
• Direct connection; telephones are sold in pair

                                    F

                             A             E




                             B             D

                                    C
                                                     273
                 Switches

• As the number of users increases, switching
  centers are more economical

                                      F

                               A                E




                               B                D

                                      C


                                                    274
              Digital Switches

• Took more than 100 years from
  analog to digital voice
  transmission
   – Better quality for long
     distance calls
• Demands to telephone network
  become constantly higher
   – World-wide communication
     network




                                  275
           Mobile Communications


• Bell Laboratories introduced the idea of
  cellular communications in 1947
• Motorola and Bell Labs in the 60s and early 70s
  were in a race to design portable devices
• Dr. Cooper, 2-pound Motorola handset (1973)




                                                    276
                      What is VoIP?

• Use a LAN and/or WAN to carry voice in the same
  way as the telephone system.
• Why?
  – Save costs
  – Improve facilities.




                                                    277
                   VoIP Gateway
 The interface between VoIP and PSTN




•An essential feature for VoIP



                                       278
Cheap phone-cards/voice carriers




                                   279
               Carrier Grade VoIP
• Carrier grade and VoIP
   – mutually exclusive
   – A serious alternative with enhanced features
• Carrier grade
   – The last time when it fails
   – 99.999%, five-nines reliability
   – Verizon network supports 70M voice access
     lines
   – AT&T serves 300M voice calls a day
   – Short call setup time, high speech quality
   – no perceptible echos, noticeable delay or
     annoying noises
   – Self-healing, highly scalable and manageable
                                                    280
                       VoIP

• Transport voice traffic using IP
• Voice over the Internet?
   – Interconnected networks
   – Applications: e-mail, file transfer, e-com
• The greatest challenges
   – Voice quality and bandwidth
   – Control and prioritize the access
• Internet: best-effort transfer
   – The next generation
   – VoIP != Internet telephony


                                                  281
                                    IP

• A packet-based protocol
   – Routing on a packet-by-packet base
• Packet transfer with no guarantees
   – May not receive in order
   – May be lost ore severely delayed
• TCP/IP
   –   Retransmission
   –   Assemble the packets in order
   –   Congestion control
   –   Useful for file-transfers and e-mail




                                              282
                     Data and Voice

• Data traffic
   – Asynchronous – can be delayed
   – Extremely error sensitive
• Voice traffic
   – Synchronous – the stringent delay requirements
   – More tolerant of errors
• IP is not for voice
• VoIP must
   – Match the PSTN
   – Offer new and attractive capabilities at a lower cost




                                                             283
                      Why VoIP?

• Why carry voice?
  – Internet support instant access to anything
     • Everything can be done on the net? “Dot-com guy”
     • Many new services and applications
  – However, voice services provide more revenues
• Why use IP for voice?
  – Why try to fix something that is not broken?
  – Circuit-switching is not for datacom
  – IP
     • Equipment cost, integrated access, less bandwidth, and
       widespread availability




                                                                284
           Lower Equipment Cost

– PSTN switch
   • Proprietary – hardware, OS, applications
   • High operation and management cost
   • Training, support and feature development cost
– Mainframe computer
– The IP world
   • Standard hardware and mass-produced
   • Application software is quite separate
   • A horizontal business model
– IN
   • does not match the openness and flexibility of
     IP
   • A few highly successful services
                                                      285
– Moore’s Law
  • Processing power
    doubles every 18
    months
  • Frame     10
  • Router    20
  • ATM       40
  • Circuit   80




                       286
             Voice/Data Integration

– Click to talk application
   • Personal communication
   • E-commerce
   • CTI – Computer Telephony Integration
– Web collaboration
   • Shop on-line with a fried at another location
– Video conferencing
– IP-based PBX
– IP-based call centers



                                                     287
         Lower Bandwidth Requirements
– PSTN
   • G.711 - 64 kbps
   • Human speech bandwidth < 4K Hz
   • The Nyquist Theorem: sample rate twice the bandwidth
   • 8K * 8 bits
– Sophisticated coders
   • 32kbps, 16kbps, 8kbps, 6.3kbps, 5.3kbps
   • GSM – 13kbps
   • Save more by silence-detection
– Traditional telephony networks can use coders too
   • But it is difficult
   • So many switches
– VoIP – two ends of the call negotiate the codec
                                                            288
      The Widespread Availability of IP

– IP
   • LANs and WANs
   • The ubiquitous presence
– VoFR or VoATM
   • Only for the backbone of the carriers
– Voice over WLAN
   • Voice over WiFi for now
   • Voice over WiMax could be a real threat for
     PLMN



                                                   289
             The VoIP Market

• The revenue projection
  – Value-added service




                               290
• Revenue breakdown
  – VoIP
  – Fax over IP




                      291
                  VoIP Challenges

• Speech quality
   – Must be as good as PSTN
   – Delay
      • The round-trip delay
      • International calls through satellite – 500-600 ms
      • G.114 – < 300 ms
   – Jitter
      • Delay variation
      • Different routes or queuing times
      • Adjusting to the jitter is difficult
      • Jitter buffers add delay

                                                             292
– Packet loss
   • Traditional retransmission cannot meet the real-time
     requirements
   • Packets must be played in order
– Speech-coding techniques
   • MOS, Mean Opinion Score >= 4
   • P.800, but subjective in nature
   • G.711           64kbps          4.3
   • G.726           32kbps          4.0
   • G.723 (celp)    6.3kbps         3.8
   • G.728           16kbps          3.9
   • G.729           8kbps           4.0
   • GSM             13kbps          3.7
   • iLBC            13.33/15.2kbps high robustness to packet loss
   • iSAC            10-32kbps       wideband codec

                                                                     293
     Network Reliability and Scalability

– PSTN system fails
   • Five-nines reliability
– The office computer network fails
– Today’s VoIP solutions
   • Redundancy and load sharing
   • Scalable too – easy to start small and expand
   • Fiber-optic transport, gigabit router, high-speed
     ATM base




                                                         294
  Managing Access and Prioritizing Traffic


– A single network for a wide range of applications
– Call admitted if sufficient resources available
– Different types of traffic are handled in different
  ways
– QoS has required huge efforts




                                                        295
            VoIP Implementations
• IP-based PBX solutions
  – A single network
  – Enhanced services




                                   296
• IP voice mail
   – One of the easiest applications
• Hosted PBX solutions
   – For SOHO
   – Internet and telephony access
• IP call centers
   – Use the caller ID
   – Automatic call distribution
   – Load the customer’s information on the agent’s
     desktop
   – Click to talk
                                                      297
298
• IP user devices
   – VoIP protocols, SIP
   – Integrated functions
      • Telephony, WWW, e-mail, voice mail,
         address-book
   – WiFi phone




                                              299
• Skype
   – A peer-to-peer VoIP client developed by KaZaa in
     2003
   – Skype can
      • work almost seamlessly across NATs and
        firewalls
      • has better voice quality than the MSN and
        Yahoo IM applications
      • encrypts calls end-to-end, and stores user
        information in a decentralized fashion
      • SkypeOut, SkypeIn

                                                        300
                          New applications

• The networks are converging
• Possible applications
   –   Video Phones
   –   Conferencing
   –   Collaboration Tools
   –   Distance Learning / Training
   –   Tele-medicine, tele-repair, tele-…
   –   On-line gaming
   –   Dating Applications
• Skype is rolling out developer kits and programs to
  encourage innovation, similar to the wireless industry
  promoting application development on their platforms



                                                           301
                Why Internet Telephony?

• The business case
   – Integration of voice and data
   – Bandwidth consolidation
   – Tariff arbitrage
• Universal presence of IP
• Maturation of technologies
• The shift to data networks




                                          302
                            VoIP Spectrum
• Traditional Telecomm Segments in transition to VoIP
   – International Low cost calling
   – Internal networks of large carriers
   – Numerous equipment makers, software providers
   – Residential VoIP phone service
      • This area is exploding: Vonage, Packet8, Broadvoice …
   – Office PBX systems
      • Using VoIP inside a company location, and between corporate branches
   – Call Center
• Instant Messaging
   – Not only the traditional big 3, but newcomers like Skype …
• Consumer and Business Application Areas
   – Voice applications
   – Wireless Internet applications




                                                                               303
                  Course Overview

•   VoIP and RTP
•   Voice codecs
•   H.323
•   SIP – simple and flexible
•   MGC and softswitch
•   SS7, UMTS
•   QOS
•   Voice over WLAN
•   P2P IP communications
•   Charging and payments


                                    304
                  VoIP – Big Picture

• User’s voice converted
  from analog to digital
  signal.
• Digital signal is
  compressed.
• Compressed signal is
  assembled into
  packets.
• Packets transported
  over IP networks.


                                       305
              Technical Issues

• For good voice quality we need to ensure that
  latency does not exceed 200ms.
• IP Networks have several sources of delay which
  increase latency




                                                    306
PSTN Vs. VoIP




                307
Current VoIP Implementations




                               308
                   Conclusion

• VoIP provides a cost effective solution
• Can envision a wide array of applications that can
  complement VoIP
• However, previous slide shows several issues that
  need to be resolved before widespread deployment.




                                                       309
                        Circuit-Switched Telephony
                          Traditional PSTN Approach
                                                                  SCP
                                SS7 Signaling Network
                                                            Most service logic in
              Signaling                                     local switches, rest
                                                 Class 4          in SCPs
                                                 Switch




              Class 5                                           Class 5
                                 Circuit-based Trunks           Switch
              Switch
                                  64 kb/s digital voice
   Typically analog
“loop”, conversion to                        Media stream
digital at local switch



                 Data travels over a parallel (but separate) network

                                                                                    310
                                VoIP
               Goals and Potential Benefits

• Consolidation of voice, data on a single network
   – Simplify infrastructure, operations; provide bundled services
• Support for intelligent terminals as well as phones
• Increased flexibility
   – Multiple bit rates, multiple media types, richer signaling
   – Distinguish calls from connections (add/modify streams during
     call)
• Separation of service control from
  switching/routing
   – Accelerate new service development, increase end-user control,
     evolve from VoIP towards advanced services
• Expansion of competition


                                                                      311
                  Packet Voice Transport

• Key targets for voice call service quality:
   – Average packet loss: < 2%
   – Consecutive packet loss: < 200 ms burst
   – End-to-end (lip-to-ear) delay: < 150 ms for comfortable conversation
• Packet loss cannot be corrected by retransmission (TCP),
  because the packets arrive too late to be useful
• Use RTP (Real-time Transport Protocol) over UDP (User
  Datagram Protocol) for voice or video transport
   – Payload ID, sequence numbers, timestamps, monitoring via RTCP
• Packet and buffer lengths limited by constraint on end-to-
  end delay
• Typical codecs: G.711 (64 kb/s), G.729 (8 kb/s) G.723 (~ 6
  kb/s)
   – Transmitted bit rates depend on overheads, optional silence
     suppression
                                                                            312
                         H.323 Architecture
                                     ITU-T
                                                         3 stages of signaling:
                                                         • RAS to Gatekeeper
                              H.323 Gatekeeper
                                                         • H.225 call signaling
                                                         • H.245 media stream control
                                                         (can be simplified for VoIP)


                         H.323                                   PSTN
                        Terminal
                                                H.323
                                               Gateway
          H.323 Zone
                                      H.323
                              Multipoint Control Unit

•   Telco-centric multimedia,multiparty conferencing (initially for LANs)
•   Gatekeeper for network control, heavy-weight protocols
•   Widely deployed in first wave of VoIP standardization

                                                                                        313
            SIP (Session Initiation Protocol)
                  IETF Multimedia Architecture

• Internet-centric alternative, initially for large multicast
  conferences
    – SIP for call signaling, SDP (Session Description Protocol) for media
• Initially very simple, light-weight, loosely-coupled sessions;
  oriented towards direct signaling between endpoints
• Network servers for additional capabilities:
    – Registrar for terminal registration, aliases
    – Redirect returns contact address directly to end user
    – Proxy forwards signaling (requests, responses)
• Evolution towards greater use of proxy/registrar for locating
  users, vertical services, call tracking, network control
• Strong, rapidly growing support (e.g., Microsoft XP, 3GPP)


                                                                             314
                        SIP Call Setup
                        Simplified View
 lts.ncsc.mil                                                          telcordia.com
                           DNS
                                                         Location
                                                          server

                                   INVITE
                                   Ringing
                Proxy
                                   200 OK                 Proxy
             INVITE                 ACK                       INVITE
peter@telcordia.com
                                                              Ringing

                                                     200 OK


                                 Media Streams
          Linda                                                  Peter

          INVITE SDP proposes media type(s), IP & ports to send to
        200 OK SDP accepts/rejects media, gives IP & ports to send to

                                                                                 315
               Where Do Services Live?

• Some implemented at the endpoints
   – Last-number redial, call hold...
• Others may be better supported from the network
   – Avoid need for PC or IP phone to be turned on (call forwarding)
   – More complex services, such as conferencing
   – Integration with web-based services (unified messaging)
• Example: SIP Proxy runs a script for each incoming
  call for Peter
   – Parallel forking: forward INVITE to multiple endpoints
     simultaneously
   – Sequential forking: try his office PC first, then lab, then cell phone,
     …

                                                                               316
           SIMPLE (SIP for IM and Presence)
                          Simplified Example
 lts.ncsc.mil                                                         telcordia.com
                                                           Presence
                                                            server

                                SUBSCRIBE
                                   NOTIFY
                Proxy
                                   NOTIFY
                                                            Proxy
       SUBSCRIBE
peter@telcordia.com                                            Update
                                                               Presence




          Linda                                                  Peter

        Linda subscribes to notifications of changes in Peter’s status:
                  Off-line, on-line, busy, away, available, ...
                                                                                      317
                  NGN Architecture
               Next-Generation Network
• Oriented towards application of VoIP (or VoATM) to
  large-scale public networks
• Focus on scalability, network control, support for
  traditional phones, sophisticated gateway (GW) to the
  PSTN and its services
• Media GW interfaces voice stream to PSTN trunk or
  phone line
• Signaling GW allows signaling directly to SS7
  network
• Softswitch controls Media GWs and does call
  processing
   – Allows smaller, cheaper Media GWs (e.g., for
     individual homes)
   – Control via MGCP (Media Gateway Control
     Protocol) or H.248                                   318
                            NGN Example
                      Voice over DSL or Cable Modem

                                                NGN             PSTN



                                 Softswitch
                                                                       SCP

 IP Phones,
     PCs
                                                                         SS7
                                                       SS7
                                                                      Signaling
                                                     Gateway
                                                                      Network
                               Core Packet
                                Network
                DSL or                                Trunk
Customer
              PacketCable       Voice Streams        Gateway
Gateway
                Access
                                                            Class 5
                                                            Switch



Can also use to interconnect PSTN clouds (long-distance),
        or PSTN switches (interoffice backbone)
                                                                              319
International Voice Market
 Calls Terminated on PSTN




                   Source: Telegeography 2001
                  (2001 figures were projections)


                                                    320
               Carrier Applications of VoIP
• First major inroads for VoIP have been in long-
  distance
    – Avoid regulation, high international PSTN tariffs
    – VoIP invisible to end user, doesn’t rely on him to do anything
    – Installed base dominantly H.323, movement now towards NGN
•   Local-carrier interest for interoffice connections
    – Consolidate voice and data networks (typically ATM)
    – Use NGN, or packet-enable existing switches
• Many trials of VoIP to residences, but deployments
  few
    – Cable TV has laid groundwork for NGN approach (DOCSIS
      1.1)
    – Decline of CLECs likely to slow multi-line VoDSL


                                                                   321
                 Enterprise VoIP
                                                        Location B
                             PSTN
  Location A                                              Centrex
                                                          or PBX

                           Core IP
                           Network
                                              GW
                                              GW
    IP PBX
                                                       Softswitch


                                                            IP
      IP                                                  phone
    phone

Many possible combinations of VoIP and circuit-switched telephony



                                                                     322
          Enterprise Applications of VoIP
• Leverage spare data-network capacity, minimize phone
  bills, create platform for multimedia conferencing
• H.323 and SIP both being deployed, softswitches and
  IP-PBX options emerging, unclear which will prevail
• Examples: Telcordia/SAIC (H.323), Telia (SIP)
• Carrier-managed VPN networks last year from AT&T
  (H.323) and Worldcom (SIP)
• VoIP adoption slower than expected, partly due to
  plunging PSTN long-distance prices, QoS concerns



                                                         323
                        Peer-to-Peer VoIP
                                PC-to-PC
• Internet Telephony revisited, often facilitated by
  software or network servers from new types of voice
  service providers
   – Microsoft, Net2Phone, Dialpad, AOL, Yahoo!
   – Mass market alternative to telcos, requiring limited network
     infrastructure, capital costs, operating expenses
• What’s the business case for “free” VoIP?
   – Sell advertising, software, or enhanced services
   – Charge for PC-to-phone, phone-to-phone
   – Give away as a competitive differentiator
• Mostly H.323 today, likely to move towards SIP
• Could be key industry driver, even if penetration were
  limited
                                                                    324
                    Outlook for VoIP
                 Current Status and Trends

• VoIP is not monolithic – many applications, with different
  drivers, will maintain a heterogeneous mix of technologies
• H.323 is most widely implemented today, but trends are
  towards SIP for intelligent terminals, NGN for most carrier
  networks
• Most success thus far in long-distance networks, perhaps
  with local carrier backbones to follow in next few years
• Footholds made in enterprise and access markets, but
  VoIP has not taken off as fast as initially expected
• Adoption being slowed by economic conditions,
  plummeting long distance rates, declining advertising
  market (peer-to-peer)
                                                          325
               Continuing Challenges
• Quality of Service
   – Diffserv, MPLS, traffic engineering, bandwidth brokers, call
     admission…
   – What is really needed for consolidated voice and data
     networks?
• Security, reliability
• Extending SIP to provide conference control
• Operations (configuration of IP phones, version
  control and upgrading of highly distributed
  software, accounting/billing,…)
• Packet-level interconnection of VoIP islands which
  use competing architectures and protocols
• Controlling feature interactions in a distributed-
  services environment
• Traversal of NATs and firewalls
• Support for services beyond voice                                 326
                         NAT Traversal

• Network Address Translators (NATs) map a
  private IP address space to externally visible
  (public) IP addresses
   – Conserve scarce public IP addresses
   – Shield internal hosts from outside world
• Useful for enterprises, cable modem networks,
  broadband access routers, internet cafes…
• NATs interfere with peer-to-peer protocols such as
  SIP
   – SIP clients must identify the IP address and ports they will use
     to receive media streams (in payload of their signaling
     messages)
   – But they don’t know their externally visible addresses
• “One of the SIP community’s biggest problems”

                                                                        327
                         STUN – Simple Traversal of
                            UDP Through NATs
                     draft-rosenberg-midcom-stun-01.txt
                                        STUN Server


Private Network A                                Internet
                                                                   Private Network B

                                   STUN
                               Request/Response
                      NAT                                    NAT


                              SIP
  STUN Client               Signaling                                    STUN Client
   SIP Client                                                             SIP Client

                                 SIP Proxy/Registrar
                                                            Source: P. Thermos, Telcordia

STUN client contacts STUN server, discovers NAT, address translation
SIP client uses “external” address in signaling for setup of media streams
This approach being implemented and tested at Columbia and LTS
                                                                                            328
                    Advanced Services
• VoIP: natural platform for evolution to advanced
  services
   –   Supports intelligent terminals and rich signaling
   –   Separates calls from connections
   –   Multimedia capabilities already in the protocols (SIP/H.323)
   –   Removes bottleneck by separating call control from switching
• Thus far, focus is almost entirely on voice
   – For many players (but not all), voice is the killer app
   – Solve the simpler problem first
• This simplifies many network control issues, because
  of predictability of voice bandwidth, traffic patterns
   – But current solutions are likely to require significant
     extensions to accommodate more flexible advanced services


                                                                      329
             Moving Beyond Two-Party Voice
         What’s Different About Advanced Services?

• Flexibility in media streams, participants, “ownership”; service
  not pre-defined at call setup
   – Multiple media per call, differing (and very wide range of) bandwidths
   – Dynamic reconfigurability during call
   – Potential for multicast conferencing, streaming
• Implications
   – Call admission control becomes more complex
   – Much less aggregation, localization of flows than with NGN voice
   – Usage, traffic patterns may be highly variable and hard to predict
• New approaches to traffic engineering, resource allocation and
  network control will be needed to address even a modest
  penetration of these new services



                                                                              330
What is Voice over IP (VoIP)?
       1010101000010        IP Packet
)))    1001010101010
       1001010101010
       0101010001001
                            1010101000010
                            1001010101010
                            1001010101010
                            0101010001001




                            Internet/
                       Private IP Network


       1010101000010

(((    1001010101010
       1001010101010
       0101010001001
                            IP Packet
                            1010101000010
                            1001010101010
                            1001010101010
                            0101010001001

                                            331
                    What is IP Telephony ?

      ITU Definitions

     VoIP : Use of Private Networks
     Internet Telephony : Use of Public Network
     IP Telephony : VoIP + Internet Telephony


       SG Definitions


                                    This deployment is VoIP
                                     (end points are PSTN)
IDA has framework for VoIP

                                     End points are IP?
                                          Hmm…

                                                              332
   VoIP Deployments

                   Phone to Phone
1998               is mainly provided by
                    Service Provider or
                      Private Network
                     (E.g. Singtel’s 019)


1999             PC(Web) to Phone
                   is mainly provided
                   by Service Provider
                    (E.g. Web2Phone)

                IP Phone to IP Phone
2002               is mainly provided by
                    Service Provider or
                       self Managed
                   (E.g. Free World Dial)

post              Wireless IP Phone
2003             to Wireless IP Phone
                 will be provided by whom?


                                             333
Demo: IP-to-IP Calls

                 Internet




                            334
Demo: IP-to-PSTN Calls


                 Internet



    PSTN



                            335
          Implications


    No more time-based
     telephony charges


No concept of local call, long-
  distance call or IDD call

                                  336
                 Change in Business Paradigm
             Traditional Telecom Model                     New Telecom Model


                        Value-Added
                                                                 Voice
                           Service


                         Data Service                        Value-Added Service

                   Voice Service                                  Data Service


                   Infrastructure                                Infrastructure

“The most powerful paradigm shift is the fact that applications are not woven into the
platforms. Now to be a phone company, you don't have to weave tightly the voice service
into the infrastructure. They (Vonage) turn voice into a application and shoot it across one
of these platforms. And, suddenly, you're in your business.”
                                                         – Michael Powell, Chairman of FCC
                                                                                          337
                   Technical Issues
      Technical Architecture

To achieve interoperability between different IP Telephony services,
         What should be the technical common platform?
         What are the technical specifications to adopt?


         Quality of Service
What is the acceptable QoS for IP Telephony?
How to ensure QoS?


       Security and Privacy
How secure is IP Telephony?
What about wiretapping requirements?



                                                                       338
                    Business Issues
          Interconnection
Who can interconnect to PSTN?
What is the pricing model?


           Market Studies
What is the economic impact?
How would it change the telco landscape?
(cost to setup IP Telephony service ~= cost to setup Email service)


            Numberings
What’s is the numbering plans for IP Telephony?
How are numbers assigned for IP Telephony services?
What if there is no service provider?


                                                                      339
                  Regulatory Issues
            Classification
How do we classify all the different IP Telephony services?
Are they subjected to similar “regulation”?


              Licensing
Who needs to apply for license?
What if there is no service provider?



          Universal Access
Is it applicable to IP Telephony?
How about emergency numbers?


                                                              340
                   What is ENUM


• RFC 3761 defined by IETF - The E.164 to Uniform Resource
  Identifiers (URI) Dynamic Delegation Discovery System (DDDS)
  Application (ENUM)
• This document discusses the use of the Domain Name System
  (DNS) for storage of E.164 numbers. More specifically, how
  DNS can be used for identifying available services connected
  to one E.164 number.




                                                                 341
                     What is ENUM


  Electronic Numbering (ENUM) provides
  a mechanism to assign an E164 phone
         number to an IP resource.




sip:dickson@poc.tech.org.sg   sip:xiayang@sp.edu.sg
       +65 6411 1000               +65 6411 1201
                                                      342
     How does ENUM work

take a phone number

           +65-6411-1234

…and turn it into
a domain name!

     4.3.2.1.1.1.4.6.5.6.e164.arpa

                                     343
           How does ENUM work


           4.3.2.1.1.1.4.6.5.6.e164.arpa
ask the DNS

                       mailto: dickson@ida.gov.sg
get a list or URIs
    returned         sip: dickson@poc.tech.org.sg

                         http: www.ida.gov.sg


                                                    344
                     Three Tier Model


Tier 0 – e164.arpa
ITU
Zone Delegation, i.e.
e164.arpa to 5.6.e164.arpa
 Tier 1 – 5.6.e164.arpa
 Country Level (Singapore)

Zone Delegation, i.e.
5.6.e164.arpa to 1.1.4.6.5.6.e164.arpa

  Tier 2
  Has the NAPTRs in it, i.e. for
  4.3.2.1.1.1.4.6.5.6.e164.arpa

                                         345
How does ENUM work




                     346
             ENUM in IP Telephony

                                          DNS-Server
             Query
4.3.2.1.1.1.4.6.5.6.e164.arpa.?


         Response
sip:dickson@poc.tech.org.sg
                                                       “Call setup”



             Dial                          SIP
        +65 6411 1234             dickson@poc.tech.org.sg




                        Gateway                         Sip server


                                                                      347
            What problem does ENUM solve?

• Legacy phone calls to IP phone
• Two different routing domains, need mapping


                              IP-Network
                     STP
   Switch
               SS7




                                                348
            Technology adoption life cycle

                                       Late majority
       Early adopters                 (conservatives)
        (visionaries)
                                               (Geoffrey
                                                Moore,
                        Main market             Chasm
                                                Group)



                Early majority
                (pragmatists)             Laggards
    Innovators                            (sceptics)
(technical people)
                                                           349
          Technology adoption life cycle

                                       Late majority
       Early adopters                 (conservatives)
        (visionaries)
 ENUM
                        Main market


                Early majority
                (pragmatists)             Laggards
    Innovators                            (sceptics)
(technical people)
                                                        350
         Country Code Delegations

•   31   Netherlands   86     China
•   33   France        246 Diego Garcia
•   36   Hungary       247 Ascension
•   41   Switzerland
                       290 Saint Helena
•   40   Romania
                       353 Ireland
•   43   Austria
•   44   UK
                       358 Finland
•   46   Sweden        374 Armenia
•   48   Poland        420 Czech
•   49   Germany       421 Slovakia
•   55   Brazil        423 Liechtenstein
•   65   Singapore     971 UAE

                                           351
             Worldwide ENUM Activities

•   Austria http://enum.nic.at/
•   China http://www.enum.cn/index-en.html
•   France http://www.numerobis.prd.fr/welcome.shtml
•   Germany http://www.enum-center.de/
•   Netherlands http://www.enuminnederland.nl/
•   Korea http://www.enum.or.kr/en/
•   Sweden http://enum.autonomica.se/
•   UK http://www.ukenumgroup.org/
•   US http://www.enum-forum.org/



                                                       352
              AP ENUM Activities (Australia)

• ACA released “Expression of Interest” for Tier 1
  Registry Operator for Australian ENUM Trial
  closing date 7 July 2004
• ENUM Trial will run for 12 months, with the option
  of a 12 months extension.
• Workshops and seminars organised, and ENUM
  discussion group formed
• ENUM Website:
  http://www.aca.gov.au/telcomm/telephone_numbering/enum_ns
  g2/index.htm



                                                              353
             AP ENUM Activities (China)

• Led by Ministry of Information Industry
• Study of technical and administrative issues
• Detailed ENUM trial plan is being prepared
• Active involvement in ITU-T (SG2) discussions
• Security and reliability issues very important
• Several carriers / vendors / service providers show great
  interest
• Web Site: http://www.enum.cn/Enum/EnumReg/English/home.php




                                                               354
             AP ENUM Activities (Japan)

•   MPHPT studying regulatory issues
•   Signed MoU with South Korea on ENUM DNS in Feb 2004
•   Considering ENUM trial
•   Interest in security and privacy issues
•   Administrative and operational arrangements under study
•   Web Site:
    http://etjp.jp/english/index.html




                                                              355
          AP ENUM Activities (Korea)

• ENUM National Trial implementation (funded by Korean
  Ministry of Information and Communication)
• ENUM test bed project established Feb 03
• Broad involvement from DNS and telephony community
• Interest in commercial applications and ENUM APIs
• ENUM Website: http://www.enum.or.kr/kr/intro.html




                                                         356
         AP ENUM Activities (Taiwan)


• Joint government, research,
  telecommunications company and ISPs
  involvement
• Establishment of ENUM test bed
• Seminars, workshops and study groups
• Interest in determining successful business
  models
• http://trial.enum.org.tw




                                                357
         ENUM Activities (Singapore)


• Singapore received its country code top level
  delegation
• Exploring the technical issues relating to ENUM
• Working towards an ENUM trial
• Liaising with industry regarding ENUM issues




                                                    358
              ENUM Activities (APEET)

• Asia Pacific ENUM Engineering Team consists of
  ccTLD administrators:
   – SGNIC, CNNIC, JPRS, KRNIC, TWNIC
• Objective to conduct ENUM/SIP trials using the
  common golden root of “apenum.org”
• Website: http://www.apenum.org/




                                                   359
               Infrastructure ENUM

• “Infrastructure ENUM” (e.g. ibm.com, apenum.org)
  taking off more rapidly than “Individual ENUM”
  (e164.arpa)
• May be driving force due to ease of mirroring DNS
  records from one domain to another




                                                      360
        ENUM Implications


     ONE number for telephone, fax,
       e-mail, web site address


E164 numbers available for IP Telephony


Driving factor for IP Telephony, leading to
convergence of PSTN and Data Networks



                                              361
      Considerations for ENUM Implementation

          Administration

Who should administer ENUM database at a national level?
How do we choose the administrator?
Should there be multiple entities managing multiple number ranges?




          Charging Model
What should be the cost of ENUM registration?
Should there be a bidding process for golden numbers?




                                                                     362
            Considerations for ENUM Implementation

          Numbering Plan

Should there be a new range of numbers to delineate ENUM
numbers?
How do we ensure there is no “clash” with current PSTN numbers?
How to ensure ENUM services complies with number portability
arrangements



        Privacy & Security
Should participation in ENUM by telephone users be compulsory or
should there be an opt-in?
What are the procedures for protecting the security, integrity and
privacy of the ENUM database?


                                                                     363
What VoIP SIG have done




                          364
                 IP Telephony/Enum Trial Architecture




Intranet                                                              ENUM
                                                                       NS
                             ENUM        SIP Server    SIP Server
                              NS


                                                                    SIP Server
      PSTN Gateway   SIP Server
                                           Internet                              PSTN Gateway




                                       ENUM
                                        NS
                                                 SIP Server




                                  PSTN Gateway
                                                                                                365
          SingAREN VoIP Network


                                                            ENUM
                                                             NS
            ENUM                                   1.1.1.1.4.6.5.6.e164.arpa
             NS                                                                          PSTN
          5.6.e164.arpa                                                                 Gateway

                                                                    SIP Server




                                       SingAREN
 PSTN     SIP Server
Gateway


                                                                     ENUM
                                                                      NS
                          SIP Server
                                       ENUM                 3.1.1.1.4.6.5.6.e164.arpa
                                        NS
                              2.1.1.1.4.6.5.6.e164.arpa




                                                                                                  366
               SingAREN VoIP Network

• ENUM Delegation to
   – NTU – 65-6411-11xx
   – Singapore Poly – 65-6411-12xx
   – I2R – 65-6411-13xx
• Trial deployment of open-sourced SIP server
  (Asterisk) at NTU, Singapore Poly, I2R and IDA
• Local PSTN connectivity via Cisco 2600 in IDA




                                                   367
                Singapore Polytechnic SIP Network Setup
                                                                                                      6564111221
                                                                                               IP Address:164.78.247.121

                          6564111203       6564111204
6564111202

                                                     Telcordia
                                                     Softswitch
                                                 IP Address:164.78.247.103
                    FXS Media Gateway
6564111201     IP Address:164.78.247.104
                                                                           Wireless Access Point



   PSTN                                                                                                   6564111228
              FXO Media Gateway
          IP Address:164.78.247.105                                                                IP Address:164.78.247.128
                                                 Backbone
                                                  Network
   SINGAREN Network
                                                                                             6564111210
                                                                                      IP Address:164.78.247.110




          Vocal Server
                                                                                      6564111211
    IP Address:164.78.247.101    Asterisk PBX
                                                                               IP Address:164.78.247.111
                            IP Address:164.78.247.102          6564111212
                                                        IP Address:164.78.247.112
                                                                                                                       368
                 DEMONSTRATION OF ENUM CALLS

 6564111210

                                                                         I2R SIP PROXY        I2R ENUM DNS
                Asterisk PBX
                                                         ENUM DNS


                                                                                   I2R
                                                                                 Network
6564111211


                 SP
               Networ
                  k                                                       IDA ENUM DNS
                                              SINGAREN
                                               Network



      NTU
     Network                                                          IDA
                                                                    Network
                               Network Connection
                               Request
                               Response                                                  IDA SIP PROXY
                               RTP Session
               NTU SIP
               PROXY                                                                                     369
        Various Codec Bandwidth Consumptions

                 Encoding/           Result
                 Compression         Bit Rate
  Standard       G.711 PCM         64 kbps (DS0)
Transmission     A-Law/u-Law
Rate for Voice
                 G.726 ADPCM      16, 24, 32, 40 kbps

                 G.727 E-ADPCM    16, 24, 32, 40 kbps

                 G.729 CS-ACELP        8 kbps

                 G.728 LD-CELP        16 kbps

                 G.723.1 CELP       6.3/5.3 kbps
                                      Variable

                                                        370
     Media Link Layer Overhead

Layer 2 Media             Layer 2
                        Header Size
Ethernet                  14 bytes


PPP/MLPPP                   6 bytes

Frame Relay                 6 bytes

ATM (AAL5)                  5 bytes + waste

MLPPP over FR              14 bytes

MLPPP over ATM      5 bytes for every ATM cell
                   + 20 bytes for MLPPP/AAL5



                                                 371
                      H.323 & SIP


• Different protocols to achieve the same goal
• Reasonably easy to translate between them
• Expect proxy/translation units to be common




                                                 372
                      Soft PaBX

• New features and functionality now possible
   • Voice mail Available via email
   • Soft phone (eg: receive a call on my laptop)
• Extra features easily added (unlike PaBX)
   • LDAP/X.500 directory service




                                                    373
                 Video Conference


• This is H.323 – not just VoIP, but video conferencing




                                                          374
             Multipoint Control Unit

• Allows multiple people in a conference (where H.323
  is normally point-to-point).
• Blends Video over IP, Voice over IP, ISDN video and
  PSTN voice conferencing.
• One way to easily handle remote teaching




                                                        375
                         Introduction

• Voice over IP and IP telephony
• Network convergence
   – Telephone and IT
   – PoE (Power over Ethernet)
• Mobility and Roaming
• Telco
   – Switched -> Packet (IP)
   – Closed world -> Open world
• Vendors and Time to Market
• Security and privacy
   – IPhreakers
   – VoIP vs 3G


                                        376
              Architecture : protocols

• Signaling
   – User location
   – Session
      • Setup
      • Negotiation
      • Modification
      • Closing

• Transport
   – Encoding, transport, etc.


                                         377
             Architecture : protocols
• SIP
   – IETF - 5060/5061 (TLS) - “HTTP-like, all in one”
   – Proprietary extensions
   – Protocol becoming an architecture
   – “End-to-end” (between IP PBX)
      • Inter-AS MPLS VPNs
      • Transitive trust
   – IM extensions (SIMPLE)
• H.323
   – Protocol family
   – H.235 (security), Q.931+H.245 (management), RTP,
     CODECs, etc.
   – ASN.1
                                                        378
              Architecture : protocols

• RTP (Real Time Protocol)
   – 5004/udp
   – RTCP
   – No QoS/bandwidth management
   – Packet reordering
   – CODECs
      • old: G.711 (PSTN/POTS - 64Kb/s)
      • current: G.729 (8Kb/s)




                                          379
             Architecture : network

• LAN
   – Ethernet (routers and switches)
   – xDSL/cable/WiFi
   – VLANs (data/voice+signaling)

• WAN
  – Internet
  – VPN
     • Leased line
     • MPLS

                                       380
             Architecture : network

• QoS (Quality of service)
  – Bandwidth
  – Latency (150-400ms) and Jitter (<<150ms)
  – Packet loss (1-3%)




                                               381
            Architecture : systems

• Systems
   – SIP Proxy
   – Call Manager/IP PBX
      • User management and reporting (HTTP, etc)
      • Off-path with IP
   – H.323: GK (GateKeeper)
   – Authentication server (Radius)
   – Billing servers (CDR/billing)
   – DNS, TFTP, DHCP servers



                                                    382
               Architecture : systems

• Voice Gateway (IP-PSTN)
   – Gateway Control Protocols
   – Signaling: SS7 interface
      • Media Gateway Controller
         • Controls the MG (Megaco/H.248)
         • SIP interface
      • Signaling Gateway
         • Interface between MGC and SS7
         • MxUA, SCTP - ISUP, Q.931
   – Transport
      • Media Gateway: audio conversion
                                            383
             Architecture : firewall/VPN
• Firewall
   – “Non-stateful” filtering
   – “Stateful” filtering
   – Application layer filtering (ALGs)
   – NAT / “firewall piercing”
      • (H.323 : 2xTCP, 4x dynamic UDP - 1719,1720)
      • (SIP : 5060/udp)
• Encrypted VPN
   – SSL/TLS
   – IPsec
   – Where to encrypt (LAN-LAN, phone-phone, etc) ?
• Impact on QoS
• What is IPv6 going to change ?
                                                      384
              Architecture : phones

• IP phones
   – Softphone or Hardphone ?
   – “Toaster”
      • Updates/patches
      • Intelligence
   – Intelligence removed from the network and put on
     the end device
   – Flows between the phone and other systems
      • SIP
      • RTP
      • (T)FTP
      • CRL
      • etc.
                                                        385
                  Architecture : example
                         POTS

   SIP



                  LAN                            PSTN
                            IP PBX



IP PBX

         IP VPN                                                 POTS

         (MPLS)                                    GSM
                                     VGW


                            internet
                                                        voice
                                                        signaling
                                           SIP
            SIP
                   SIP
                                                                       386
                    Other phone networks

• POTS/PSTN [TDM]

• “Wireless”/DECT phone

• GSM

• Satellite

• Signaling (SS7)



                                           387
                           Attacks

• IPhreakers
  –   IP knowledge
  –   Known weaknesses
  –   Evolution 2600Hz -> voicemail/int’l GWs -> IP telephony
  –   Internal or external threat ?
  –   Targets: home user, enterprise, government, etc ?


• Protocol implementations
  – PROTOS


• The human element



                                                                388
              Attacks : denial of service

• Denial of service
   –   Network
   –   Protocol (SIP INVITE)
   –   Systems / Applications
   –   Phone


• Availability (BC/DR)
   –   Requires: power
   –   Alternatives (Business Continuity/Disaster Recovery) ?
   –   E911 (laws and technical aspect)
   –   GSM
   –   PSTN-to-GSM



                                                                389
                      Attacks : fraud

• Call-ID spoofing

• User rights takeover
   – Fake authentication server


• Effects
   –   Access to voicemail
   –   Value added numbers
   –   Social engineering
   –   Replay




                                        390
                   Attacks: interception

• Interception
   – Discussion
   – “Who talks with who”
      • Network sniffing
      • Servers (SIP, CDR, etc)


• LAN
   –   Physical access to the LAN
   –   ARP attacks
   –   Unauthenticated devices (phones and servers)
   –   Different layers (MAC address, user, physical port, etc)




                                                                  391
                  Attack: interception

• Where to intercept ?
   – Where is the user located ?
   – Networks crossed ?


• Lawful Intercept
   – CALEA
   – ETSI standard
   – Architecture and risks




                                         392
               Attacks : systems

• Systems
  – Mostly none is hardened by default
  – Worms, exploits, Trojan horses




                                         393
                    Attacks : phone

• (S)IP phone
  – Startup
     • DHCP, TFTP, etc.
  – Physical access
     • Hidden configuration tabs
  – TCP/IP stacks
  – Firmware/configuration
  – Trojan horse/rootkit




                                      394
                              Defense

• Signaling: SIP
    – Secure SIP vs SS7 (physical security)
•   Transport: Secure RTP (with MiKEY)
•   Network: QoS [LLQ] (and rate-limit)
•   Firewall: application level filtering
•   Phone: signed firmware
•   Identification: TLS
    – Clients by the server
    – Servers by the client
• 3P: project, security processes and policies



                                                 395
                            Conclusion

• Conclusion

• Other presentations
  – Backbone and Infrastructure Security
     • http://www.securite.org/presentations/secip/

  – (Distributed) Denial of Service
     • http://www.securite.org/presentations/ddos/


• Q&A


                  Image: www.shawnsclipart.com/funkycomputercrowd.html


                                                                         396
                   Things Change!

• Ten years ago
  – The Public Switched Telephone Network (PSTN) was just
    completing the transition to digital
  – The Internet was starting to move from academia
• Three years ago
  – Dot.com mania ruled
  – It was “reliably” forecast that the Internet was about to take
    over as the sole communications medium
• Today
  – There is a strike of capital, but
  – convergence is becoming a reality




                                                                     397
                        In the Future

• Telephony and multi-media may be just another
  application over the Internet, but
• There will need to be changes to support user
  requirements, based current expectations.
• To make this happen, there needs to be
   – Substantial resource investment, and
   – Substantial standards work
• Much current telco standards work directly relates
  to NGN (next generation networks)



                                                       398
                Network Generations

  Fixed Network             Mobile                  Data


Analogue PSTN        Analogue Mobile         X.25 Packet
                          (AMPS, NMT)


Digital PSTN          Digital Mobile (GSM,   Frame Relay
                           CDMAOne)
                                             Internet



“Carrier Grade IP”            3G             “Carrier Grade IP”
                       (CDMA2000, UMTS)




                                                                  399
      PSTN

                                                        2G Mobile

• The PSTN/ISDN is based on 64 kbit/s digital connections,
  with a separate “common channel” signalling system
   • Access may be analogue (telephony), 64 kbit/s digital (ISDN) or
     low speed digital (mobiles)
   • The network establishes an end-to-end digital connection for
     the duration of each call
• The PSTN/ISDN is designed for high reliability, specified at
  the national level and connecting to form a global network,

                                                                       400
• The (public) Internet is based on the set of protocols defined by
  the Internet Engineering Task Force (IETF)
   • The primary protocol is the Internet Protocol (IP) which
     describes a simple connectionless packet protocol able to
     operate over a range of media
   • Other protocols work in association with the IP, for example,
     TCP to assist reliable end-to-end operation
• The Internet is defined by the Internet protocols rather than by a
  standardised architecture
• The Internet provides open interfaces, supporting rapid
  innovation


           ISP

                             “The Internet”          ISP
                           (best-endeavours
                               network)

                                                                       401
             Telco Networks                    Current Internet
        • 64 kbit/s circuit switching    • Packet switching over
        • Well defined architecture,       diverse media
          fixed and mobile.              • Defined by protocols
        • Designed for high                rather than architecture
          reliability and QoS              (TCP/IP)
        • Specified at national level    • Specified at global level
          growing to global              • Best endeavours network
        • Main area for national           – no QoS guarantee
          regulation                     • Open interfaces support
        Dumb terminal, smart               rapid innovation
          network
                                             Smart terminal, dumb
Next Generation Networks                        network
 Largely Packet based (IP & ATM), with necessary extensions to give
 a level of service equal to or better than current carrier networks

                                                                        402
                    Network Evolution

• There are different paths (not mutually exclusive)
  by which “an NGN” could evolve:
   –   Interconnection of enterprise IP VPNs
   –   IP expansion of existing carrier networks
   –   New IP-based networks providing integrated service
   –   Addition of QoS support to the existing public Internet
• What is the underlying demand, the business
  case and the likely timing? (The economics of
  adding QoS to the existing Internet do not seem
  compelling.)




                                                                 403
              Quality of Service Provision

• Future networks need to provide adequate Quality
  of Service to support real-time interactive services
  (e.g. voice)
• There has been extensive work on “adding” QoS to
  the Internet
• Implementation of QoS can be
   – standards driven (primarily IETF work),
   – based on proprietary approaches
   – provided by traffic segregation and traffic engineering (over-
     provision of underlying resources)
• Almost all work in the IETF has been directed with a
  single network rather than across networks (NNI or
  inter-domain)
                                                                      404
                         Carrier Networks are not
                              homogeneous
               Carrier networks consist of multiple domains
                                           Domain 3

                                                Service
                                                Domain
             Service network

  Client


            Transport Network            Transport Network     Client




                      Domain 1             Domain 2

each domain may have its own policies                         Relevant
each domain may have its own commercial goals
                                                             interfaces
and possibly its own protocols & transport
Source: ITU-T SG 11
                                                                          405
                       Why Change?

  The current circuit switched PSTN provides good
  service. What are the likely drivers for change to
  a packet-based network?
• Flexibility
   – The PSTN is based on carrying 64 kbit/s circuits.
      • Services at bit-rates below this can be carried (but not
        efficiently)
      • Services at bit-rates above this can only be carried by
        combining 64 kbit/s circuits
   – Open interfaces supporting innovation
• Economics


                                                                   406
PSTN
                                    2G Mobile
       ISP




  “Carrier
 Grade” IP-
   based
 Networks



             ISP   “The Internet”
                      (best-          ISP

                    endeavours
                     network)
                                                407
     What is Needed for “Carrier Grade IP”?

• The current Internet
   – Does not provide differentiated quality connections
   – Queues packets at peak times for maximum efficiency
• If the Internet is not congested, real-time (e.g.
  voice) packets can be delivered, but if there is
  congestion, real-time services cannot be
  supported reliably.
• There is a need for connection-oriented support
  to provide a required level of QoS for the duration
  of a connection (or, in telco terms, a call)


                                                           408
                       To provide QoS…..



  Backbone transport



• An underlying backbone transport is required (for
  example, by SONET/SDH over optical fibre or
  radio)
• Backbone resource control protocols such as
  GMPLS with RSVP-TE or CR-LDP can be used to
  provide support for resource allocation



                                                      409
                       To provide QoS…..

  Bearer Control


  Backbone transport




• It is then necessary to establish specific support
  for end-to-end connections for the duration of the
  connection/call.
• This can be provided by MPLS enabled routers,
  or by the use of the virtual circuit capabilities of
  ATM


                                                         410
                       To provide QoS…..


  Call Control


  Bearer Control


  Backbone transport


• Per call (or session) call control is needed for the
  duration of each call, to set up, supervise and
  clear-down.
• Possible protocols include
   – BICC (from the ITU-T)
   – SIP (from the IETF)
   – H.323 (from the ITU)
                                                         411
                        To provide QoS…..

  Service/application


  Call Control


  Bearer Control


  Backbone transport



• SIP and H.323 are end-to-end protocols
• An alternative approach is to use centralised
  control from a Media Gateway Controller /
  Softswitch, combining bearer and call control
   – Megaco/MGCP H.248 has been developed by
     IETF and ITU
                                                  412
                    QoS Support




• Caller must specify requirements
• Access network and subsequent networks must
  provide the required QoS for the duration of the call
  Current Internet protocols can support this within one
  network, but not across different networks
                                                           413
PSTN
         ISP
                                          2G Mobile

                                    ISP
  “Carrier
 Grade” IP-
   based
                  ?             ?
 Networks


         ISP

               “The Internet”               ISP
                  (best-
                endeavours
                 network)                             414
           End-to-End Connectivity?




There is no shortage of possible approaches – and
they are all in use!
The problem
– How to guarantee end-to-end service with the required QoS across
  multiple networks using incompatible implementations
  [the subject of current international work]

                                                                     415
                Today’s Network Architectures
                            Frame
                            Relay           IWF
                                                        PSTN/ISDN
                           Networks

                     IWF


                                                                      IWF
          IP/MPLS                                 IWF
          Networks                    IWF

                                                                            Radio
                                                         IWF
                                                                           Access
                                                                          Networks
                              IWF
          IWF

                                                                    IWF



    Ethernet               Wireless                        ATM
    Networks               Access                        Networks
                                                                            Source: ITU-T SG 13




•   Multiple, interworked, interdependent networks
•   Diversity of control and management architectures
•   Capacity and performance bottlenecks
•   Each network has its own control plane and management plane
                                                                                                  416
                                 Near Term Evolution
                                                                    PSTN/ISDN

                                                                                       Q & X series Rec.
  Rec. Q.931
                                                                                              Frame
                                                                                              Relay
                                                              IWF   Rec. I.580               Networks
   PSTN/ISDN                            Rec. Q.2931, PNNI
                                                                           IWF            FR OSF & NM
                          IWF

                                                                          Rec. I.555
                   Rec. I.580
 PSTN/ISDN                                      ATM                                         IETF RFCs
OSF & NM, M                                   Networks
 series Rec.
                                IWF                                              IWF              IP-based
               Wireless                                                                           Networks
                                                                       Rec. Y.1310
               access                 ATM OSF & NM, M series Rec.
                                                                                              SNMP based

For                                                         Against
• Convergence on ATM core                                   • Lack of service transparency
  networking enables initial stage of                         between IP based services
  unified management and control                              and ATM/PSTN services
• Enhanced performance and QoS
  capabilities for multi-services over                       OSF = Operating Support Function

  common platform                                                           Source: ITU-T SG 13
                                                                                                             417
                      Medium-term - Convergence
                            on MPLS Core
       ATM
     Networks                                                                Frame
                              MPLS NETWORK                                   Relay
                                                                            Networks
                      IWF                             IWF


  Frame
  Relay
 Networks       IWF                                                        Ethernet
                                                             IWF
                                                                           Networks




 Ethernet
                      IWF
 Networks                                              IWF

                                                                            ATM
                                                                          Networks

    Label Switching Router (LSR)   Label Switched Path (LSP)

• Requires well defined interworking mechanism for all services
    • Transfer plane functions
    • Control plane functions
    • Management plane functions
                                                                                       418
                                                            Source: ITU-T SG 13
           Inter-Network Resources




• Successful solutions have to combine
   – End to end operation control
   – Inter-domain resource negotiation
                                         419
            Inter-network Negotiation




Alternative approaches include
   – Requiring each network to support a limited range
     of QoS/network services (inflexible and
     prescriptive)
   – Network by network negotiation (but how to
     ensure required service is available?)
                                                         420
                       Ongoing Work

• International and national work is need to
  introduce interoperable next generation
  networks. Areas requiring work include
   –   Architecture and Protocols
   –   End to end QoS
   –   Service platforms
   –   Network management
   –   Lawful interception
   –   Security
• This work is being carried out in the IETF, the ITU
  and regional telco standards bodies such as ETSI


                                                        421
                  Ongoing Work

• Given the proposed use of Internet Protocols,
  much current IETF work is directly relevant
• Work is needed to define inter-network (inter-
  domain) interconnection and operation
• The following slides summarise some of the
  current work at the international level. Other
  bodies working on NGN include fora and
  consortia such as the Multi-service Switching
  Forum and the MPLS Forum.




                                                   422
             International Work - IETF


• RTP (Real Time Protocol)
   – Carries VoIP audio media
   – Used by H.323, SIP, Megaco/H.248, others.
• SDP (Session Description Protocol)
   – Describes multimedia sessions
   – Used widely as well, see above.
• SIP (Session Initiation Protocol)
   – Rendezvous protocol, discovery and session management
   – Commonly used as VoIP signalling protocol
   – Associated with MMUSIC, SIP, SIPPING, SIMPLE WGs



                                                             423
             International Work - IETF

• ENUM (E.164 Number Mapping)
   – Transforms E.164 telephone numbers into URLs
   – used for SIP, HTTP, SMTP, etc.
   – Interim operation plan for e164.arpa is a collaboration
     between IETF (Internet Architecture Board) and ITU-T (Study
     Group-2)
• SIP-T (Interworking SIP & ISUP)
   – Defines encapsulation of ISUP in SIP and mapping between
     SIP & ISUP fields
   – SIP-T architecture is approved document
   – SIP-ISUP mapping is close to approval
   – Current ITU-T SG 11 work on application for NGN-legacy
     network interworking

                                                                   424
             International Work - IETF


• Interworking SIP & H.323
   – Requirements almost complete
• Security and VoIP
   –   TLS, Digest, S/MIME, IPSEC IETF protocols from Security Area
   –   Used to secure SIP and SDP
   –   SRTP
   –   SIP Privacy/Identity work
   –   MIDCOM (firewall control)




                                                                      425
            International Work - IETF

• Media Gateway Control
   – Megaco
   – MGCP
• Transports for VoIP
   – SCTP
      • Signalling transport
   – New work begun on DCCP, unreliable protocol with
     congestion control properties
• Service development
   – CPL (Call Processing Language)
   – SIP CGI (Applying HTTP service creation to SIP)
   – New work underway on Speech Services Control



                                                        426
          International Work - IETF


• Accounting and Management
  – DIAMETER
     • AAA protocol
• Signalling Compression
  – Robust Header Compression
     • Specifications for IP/UDP/RTP headers and the SIP/SDP
       messages to be compressed, especially for wireless VoIP
       uses.




                                                                 427
             International Work - ITU

• ITU-T Study Group 13
  – Overall responsibility for IP work
  – Recommendations/areas of work include
     • Rec. Y.1541: Quantifying User QoS Needs in IP Terms
     • Rec. Y.1221: Traffic and Congestion Control in IP Based
       Networks
  – Leading ITU’s “NGN 2004” project




                                                                 428
            International Work - ITU

• ITU-T Study Group 11
  – Responsible for signalling and interworking. Current work
    includes
     • Interactions between IN and IP-based networks
     • IP-related signalling protocols
     • Bearer (ATM, IP) Independent Call Control (BICC)
     • Signalling transport over IP
     • Use of SIP for user access and network-to-network
       interfacing
  – Has just initiated new projects on signalling control
     • between session control functions (across networks),
     • between session, resource and bearer control, and
     • between session control and user profile management.
  – Other new work on control architecture and signalling
    requirements about to commence.

                                                                429
            International Work - ITU

• ITU-T Study Group 12
  – Lead group for end-to-end transmission performance. Areas of
    work include
     • Transm. Req’ts for IP gateways and terminals
     • E-Model (model for speech quality incl.VoIP)
     • Transm. Plan. for VB, Data and Multimedia
     • Transm. of multiple interconnected networks
     • Voiceband services via IP networks
     • Multimedia QOS and perf. requirements
     • Effects of multiple IP domains on VoIP
     • QOS coordination in the ITU (as Lead SG)
     • In-service non-intrusive assessment of VoIP



                                                                   430
         International Work - ITU

• ITU-T Study Group 16
  – Lead group for multimedia and convergence. Work includes
     • Voice Coding
     • Video Coding
     • Multimedia Signalling; including Data Conferencing,
       Modems, Facsimile, Call control and conference control
       and Media gateway control (H.248)
     • Security
     • Multimedia Architecture (H.323)
     • Mobility
     • Emergency Telecommunication Services




                                                                431
           International Work - ETSI

• Considerable NGN work in all areas, including
   – TIPHON (VoIP and Multimedia)
   – SPAN (Signalling and interworking)
   – Security and Coding work
• Co-ordinated by ETSI Board “NGN
  Implementation Group”
• Major input to 3GPP IP work




                                                  432
              ACIF NGN Project

• ACIF’s Strategic Plan in early 2001 identified
  need to work on “Next Generation Networks”
• Meetings with ACA, ACCC and SPAN confirmed
  they had a similar interest.
• ACCC sponsored an initial consultancy in
  second half of 2001 “to raise issues”
• ACIF held an NGN seminar in May 2002 to scope
  the issues
• Attendees proposed a continuing industry
  “conversation” on NGN matters.


                                                   433
                  ACIF NGN Project

• The ACIF Board agreed to support an ACIF
  NGN Project, working through the ACIF NGN
  Framework Options Group (“NGN FOG”).
• The aim of the ACIF NGN Project is to help all
  involved discuss issues that cross current
  boundaries, including
   –   Internet/telco divisions
   –   Regulatory issues (ACA and ACCC)
   –   Industry issues (including self-regulation requirements)
   –   Policy issues
• An early agreement was that user requirements
  must be the main driver of this work.
                                                                  434
                      NGN FOG Work

• The main task of the NGN FOG is to assist
  understanding of the transition to next
  generation network equipment. The NGN FOG
  work involves consideration of issues including
   –   Technical standards
   –   End-user issues
   –   End-to-end services
   –   Interconnection across networks
   –   Regulatory issues (both self-regulation and government
       regulation




                                                                435
SIP-Based Telephony




                      436
           Basic Telephony Services

• Basic Telephony invloves the establishment of
  sessions between endpoints.
• This chapter will focus on SIP and PSTN
  interworking for basic telephony services




                                                  437
                  SIP and PSTN interworking

• SIP and PSTN interworking occurs whenever a call
  originates in one network and terminates in the other
  network.
• Gateways are the network elements bridging the two
  networks.
• There are two basic approaches to building these
  gateways:
   – Complete protocol interworking
   – Protocol encapsulation (SIP Telephony(SIP-T))




                                                          438
                   SIP servers                SIP servers

                                                            SIP enabled
                                                              devices
                                 IP Network

                       Media:RTP       Signaling:SIP
      SIP phones

                                          Gateways

Telephons     Media:TDM PCM           Signaling:ISUP,Q.931,CAS,etc



                                   PSTN                        Telephons
              PBX
            Gateway Location and Routing

• The problem of gateway location and routing has
  been tackled in the IETF IP Telephony Working
  Group(IPTEL WG)with the development of the
  Telephony Routing over IP (TRIP) protocol
• This gateway location to the location server
  protocol,based on Border Gateway Protocol(BGP) ---
  - allows a gateway to advertise what PSTN number
  range it supports.




                                                       440
             SIP/PSTN Protocol Interworking

• SIP and PSTN prtocol interworking has two levels:
   – The media
   – The signaling
• The media interworking in a gateway involves
  terminating a PCM trunk on the PSTN side and
  bridging the media with an IP port that sends and
  receives RTP packets.
• The signaling interworking is much more
  complex.The PSTN uses many different signaling
  protocols to complete a call.



                                                      441
                SIP and Early Media

• In the PSTN,call progress indicators are often
  provided in-band in the media path,such as ring
  tone,busy signal,etc.
• In SIP,the media path is not established until the
  called party answers(200 OK).
• This is not a problem in a call from the PSTN to SIP --
  - the gateway simply takes the SIP response code
  and generates any tones or signals in the PSTN
  media path.




                                                            442
PSTN switch            Gateway                 SIP user agent
              1 IAM
                                     2 INVITE
                                     3 100 Trying
                                    4 180 Ringing
            5 ACM
          One-way speech
                                    6 200 OK
              7 ANM                  8 ACK
          Two-way speech         RTP Media Session
              9 REL
                                     10 BYE
              11 RLC
                                    12 200 OK
        No Media Session
                                   No Speech path
              SIP and Early Media

• For SIP-to-PSTN calling,the SIP phone’s local ring
  tone generated by the receipt of a 180 Ringing
  response from the gateway masks the in-band
  progress indicators being received by the gateway.
   The result is that the call may fail and the SIP caller
  will never hear any indictation,just the locally
  generated ring tone
• The solution was to add a response code to
  SIP,called 183 Session Progress,which is used to
  indicate that the call is progressing.



                                                             444
SIP user agent             Gateway           PSTN switch Telephone
         1 INVITE
                                     2 IAM
         3 100 Trying
                                     4 ACM          ACM maps to SIP
       5 183 Session Progress                       183 session progress
                                                    so that SIP caller
        One way RTP Media        One-way speech     hears ringtone,
                                     6 ANM          busytone
            7 200 OK
            8 ACK
        RTP Meddia Session       Two-way speech
             9 BYE
                                    10 REL
           11 200 OK
                                     12 RLC
       No Media Session          No Speech path
          SIP Telephony and ISUP Tunneling


   IAM                       Initial Address Message
   CgPN=314-555-1111,         Calling Party Number, Numbering Plan Indicator,
   NPI=E.164, NOA=National    Nature of Address
   CdPN=972-555-2222,         Calling Party Number, Numbering Plan Indicator,
   NPI=E.164,NOA=National     Nature of Address

   USI=Speech                 User Service Information
   FCI=Normal                 Forward call Indicator
   CPC=Normal                 Calling Party’s Category
   CCI=Not Required           Call Charge Indicator

Table9.1 ISUP IAM Message and Field Description


                                                                                446
       SIP Telephony and ISUP Tunneling

• Some field mapping is obvious,such as Calling Party
  Number to From,but others are not so obvious.
• The call routes over the SIP network to the
  destination,there is no net effect on the call
  completion,since all information usable in the SIP
  network has been mapped.
• Mapping from SIP to ISUP does not cause a loss in
  fuctionality.Some ISUP parameters that have no
  counterpart in SIP will need to be created for the
  mapped IAM.



                                                        447
      SIP Telephony and ISUP Tunneling

• A call be routed from the PSTN to SIP then back to
  the PSTN,some of the lost parameters from the first
  PSTN leg could be useful in routing in the second
  PSTN leg.
    To slove this problem for networks designed to do
  this,the encapsulation of PSTN signaling messages.




                                                        448
PSTN switch SIP-T gateway SIP-T gateway PSTN switch
        1 IAM
                   2 INVITE (IAM)
                                     3 IAM
                         4 100 Trying
                         6 183 Session       5 ACM
                         Progress (ACM)
        7 ACM             One-way
      One-way speech      RTP Media       One-way speech
                                            8 ANM
                       9 200 OK(ANM)
       10 ANM
                          11 ACK
     Two-way speech    RTP Media Session Two-way speech
          12 REL
                        13 BYE(REL)
                                              14 REL
                        16 200 OK(RLC)        15 RLC
        17 RLC
                       No Media Session   No Speech Path
     No Speech Path
           SIP Telephony and ISUP Tunneling
SIP MESSAGE           ISUP              ISDN
OR RESPONSE         MESSAGE            MESSAGE
INVITE             IAM or SAM            Setup
INFO                  USR                 User
BYE                   REL               Release
CANCEL                REL               Release
ACK
REGISTER
18x                 ACM or CPG           Alerting
200(to INVITE)      ANM or CON           Connect
4xx, 5xx, 6xx          REL               Release
200(to BYE)            RLC       Release Complete

                                                  450
             Enhanced Telephony Services

 Call transfer
 Call waiting
 Call hold
 Call park and pickup
 Calling line identification
 Incoming and outgoing call screening
 Automatic callback and recall
 Speed dial
 Interactive voice response (IVR) system



                                            451
                Enhanced Telephony Services

• Call transfer:There are three types of call transfer services
(1)blind:The transferor sends a REFER then immediately sends a BYE
    and terminates the existing session without waiting for the outcome of
    the transfer.
(2)unattended :The transferor may keep the transferee on hold pending
    the outcome of the REFER request.
(3)attended:The attended transfer involves a temporary conference call
    between the three parties in which the transferor knows the exact
    progress of the transfer.




                                                                             452
           Enhanced Telephony Services

• Call waiting:A SIP phone that offers multiple “line”
  behavior would return a 180 Ringing response and
  initiate alerting when there is an active session
  established.
• Call hold:In a SIP network,a call is placed on hold by
  sending a re-IVINTE with a connection IP address of
  0.0.0.0 in the SDP .
• Call park and pickup:A call is placed on hold at one
  location,and then retrieved at another location.
  There are a number of proposals to implement these
  features in SIP.
  Some of them use a REGISTER request and a re-
  INVITE,while others use a REFER,and then a redirect.
                                                           453
            Enhanced Telephony Services

• Calling line identification:The basic functionality is built
  into SIP to accomplish this using the From header.and
  not by a trusted source.An extesion header to SIP has
  been proposed called Remote-Party-Id,which would be
  inserted and verified by a trusted source.
• Incoming and outgoing call screening : Incoming and
  outgoing call screening can be implemented in either a
  proxy or user agent.




                                                                 454
         Enhanced Telephony Services

• Automatic callback and recall:In a SIP network, a
  SUBSCRIBE is sent to request notification when the
  called user agent is no longer busy.The NOTIFY response
  would then automatically generate a new INVITE to
  complete the call.
• Speed dial:Speed dial allows a call by dialing a shorter
  digit string.




                                                             455
       Enhanced Telephone Services

• Interactive voice response (IVR) system:
  These “voice menu” or “auto prompt” systems allow
  an automated attendant to answer a call.




                                                  456
     Call Control Services and Third-Party
                  Call Control


• The REFER Method
  – The Refer method allows a third party,such as a
    controller,to request the caller set up a call with a resource.
  – The resource is identified in a new SIP header called Refer-
    To.
  – A REFER request must contain a Referred-By header which
    identifies the referring party.




                                                                      457
                The REFER Method

• Examples of the Refer-To header are
1.To request a party to call John Doe,a REFER request is
   sent containing:
  Refer-To: sip:john.doe@isp.com
2.The same request containing the header
  Refer-To: sip:john.doe@isp.com?
  Accept-Contact=sip:jdoe@100.101.102.103;only=true
• Examples of the Referred-By header are:
   Referred-By: sip:manager@isp.com;
   ref=http://headhunters
  .com provides the reference source for
   “headhunters,”which is
   a Web page.
                                                           458
            Basic Third-Party Call Control


Controller                Party A                       Party B
     1 INVITE with no SDP
      200 OK with SDP from A
        3 ACK with hold SDP
                                     4 INVITE no SDP
                                    200 OK with SDP from B
      6 INVITE with SDP from B
       7 200 OK with SDP from A
                                   8 ACK with SDP from A
            9 ACK
                                  10 RTP Session from A to B
                11 BYE
                                             12 BYE
                   13 200 OK
            14 200 OK
               Secretary PC     Secretary Phone     Boss       Customer
1. Setup call to bass   REFER/200 OK  INVITE/200/ACK
                        NOTIFY/200 OK
                        REFER/200 OK   INVITE/200/ACK
2. Put boss on hold
                        NOTIFY/200 OK
3. Setup call to        REFER/200 OK   INVITE/200/ACK
   customer             NOTIFY/200 OK
4. Put customer on      REFER/200 OK  INVITE/200/ACK
   hold                 NOTIFY/200 OK
                        REFER/200 OK                 INVITE/200/ACK
5. Setup boss to
   call customer        NOTIFY/200 OK                  RTP audio
6. Send BYE to boss REFER/200 OK        BYE/200 OK
                      NOTIFY/200 OK
7. Send BYE to         REFER/200 OK               BYE/200 OK
   customer            NOTIFY/200 OK
8. Boss and customer talk
               Typical Configurations

•   T-1:   1.544 Mbps
•   T-2:   6.312 Mbps
•   T-3:   44.736 Mbps
•   T-4:   274.176 Mbps




                                        461
                 LAN Standards - 802

•   802.1: internetworking
•   802.2: logical link control
•   802.3: CSMA/CD
•   802.4: token passing bus
•   802.5: token ring
•   802.6: MAN-CATV
•   802.7: broadband LAN operating specifications
•   802.8: fibre optic LAN standards
•   802.9: integrating voice/data on a LAN
•   802.11: wireless LANs



                                                    462

				
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