Voice over IP VoIP

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Voice over IP (VoIP)
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Introduction (1)

• A recent application of Internet technology
  Voice over IP (VoIP) Transmission of voice over
  Internet
• How VoIP works
• Continuously sample audio
• Convert each sample to digital form
• Send digitized stream across Internet in packets
• Convert the stream back to analog for playback
• Why VoIP
• IP telephony is economic High costs for
  traditional telephone switching equipments.

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Introduction (2)

• Challenge
• Voice transmission delay
• Call setup call establishment, call termination,
  etc.
• Backward compatibility with existing PSTN (Public
  Switched Telephone Network)
• IP Telephony Standards
• ITU (International Telecommunication Union)
  controls telephony standards.
• IETF (Internet Engineering Task Force) controls
  TCP/IP standards.

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Encoding, Transmission, Playback (1)

• Both groups agree on the basics for encoding and
  transmission of audio
• Audio is encoded using a well-known standard such
  as Pulse Code Modulation (PCM).
• Audio is transferred using the Real-time
  Transport Protocol (RTP).
• RTP message is encapsulated in a UDP datagram
  that is further encapsulated in an IP datagram
  for transmission.

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Encoding, Transmission, Playback (2)

• UDP is used for transport because
• lower overhead audio must be played as it
  arrives.
• Playback cannot be stopped to wait for a
  retransmitted packet.
• Two independent RTP sessions exist, because an IP
  phone call involves transfer in two directions
• IP phone acts as sender for outgoing data, and
• IP phone acts as receiver for incoming data.

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Signaling Systems Protocols

• Main complexity of VoIP Call setup and call
  management.
• The process of establishing and terminating a
  call is called Signaling.
• In traditional telephone system, signaling
  protocol is SS7 (signaling System 7).
• In VoIP, signaling protocols are
• SIP (Session Initiation Protocol), by IETF
• H.323, by ITU
• Megaco MGCP, jointly by IETF and IUT.
• VoIP signaling protocols should be able to
  interact with SS7.

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A Basic IP Telephone System

• The simplest IP telephone system uses two basic
  components
• IP telephone end device allowing humans to
  place and receive calls.
• Media Gateway Controller providing overall
  control and coordination between IP phones
  allowing a caller to locate a callee (e.g. call
  forwarding)

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Interconnection with Others (1)

• IP telephone system needs to interoperate with
  PSTN or another IP telephone system.
• Two additional components needed for such
  interconnection
• Media Gateway
• Signaling Gateway

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Interconnection with Others (2)

• Media gateway translates audio between IP
  network and PSTN.
• Signaling Gateway translates signaling
  operations.

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Signaling Protocols

• Two major protocols H.323, SIP
• H.323, invented by ITU, defines four elements
  that comprising a signaling system
• Terminal IP phone
• Gatekeeper provides location and signaling
  functions coordinates operation of Gateway.
• Gateway used to interconnect IP telephone system
  with PSTN, handling both signaling and media
  translation.
• Multipoint Control Unit provides services such
  as multipoint conferencing.

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Signaling Protocols

• SIP Session Initiation Protocol. Invented by
  IETF.
• SIP defines three main elements that comprise a
  signaling system
• User Agent IP phone or applications
• Location servers stores information about users
  location or IP address
• Support servers
• Proxy Server forwards requests from user agents
  to another location.
• Redirect Server provides an alternate called
  partys location for the user agent to contact.
• Registrar Server receives users registration
  requests and updates the database that location
  server consults.

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H.323 Characteristics

• H.323 consists of a set of protocols that work
  together to handle all aspects of communication,
  including
• Transmission of a digital audio phone call
• Signaling to set up and manage phone call
• Allows transmission of video and data while a
  phone call is in progress
• Sends binary message
• Incorporates protocols for security
• Uses a special hardware Multipoint Control Unit
  for conferencing calls
• Defines servers for address resolution,
  authentication, accounting, features, etc.

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H.323 Layering

• H.323 uses both UDP and TCP over IP.
• Audio travels over UDP
• Data travels over TCP

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SIP Characteristics

• Operates at the application layer.
• Encompasses all aspects of signaling, e.g.
  location of called party, ringing a phone,
  accepting a call, and terminating a call.
• Provides services such as call forwarding.
• Relies on multicast for conference calls.
• Allows two sides to negotiate capabilities and
  choose the media and parameters to be used.
• SIP URI is similar to email address. (with prefix
  sip) E.g. sipbob_at_somewhere.com

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SIP Methods

• Six basic message types, known as methods

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An Example SIP Session

• User agent A contacts DNS server to map domain
  name in SIP request to IP address.
• User agent A sends a INVITE message to proxy
  server that uses location server to find the
  location of user agent B.
• Call is established between A and B. Then media
  session begins.
• Finally, B terminates the call by sending a BYE
  request.

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Telephone Number Mapping Routing (1)

• How should users be named?
• PSTN follows ITU standard E.164 for phone
  numbers. E.g. 1-613-123-4567
• SIP uses IP addresses. E.g. sipsmith_at_uottawa.ca
• In an integrated network (PSTN IP), two
  problems defined
• Locate a user
• Find a efficient route to the user
• IETF proposed two protocols
• ENUM E.164 NUMbers
• TRIP Telephone Routing over IP

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Telephone Number Mapping Routing (2)

• ENUM
• Converting E.164 phone number into a Uniform
  Resource Identifier (URI)
• Using Domain Name System to store mapping
• A phone number is converted into a special domain
  name e164.arpa
• E.g. 1-800-555-1234 ? 4.3.2.1.5.5.5.0.0.8.e164.arp
  a

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Telephone Number Mapping Routing (3)

• TRIP
• Finding a user in an integrated network
• Used by location server or other NEs to advertise
  routes
• Independent of signaling protocols
• Dividing the world into a set of IP Telephone
  Administrative Domains (ITADs)

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IP Telephones and Electrical Power

• Analog telephone system continues to work when
  electrical power are unavailable
• The wires that connect a telephone to the central
  office supply the power
• Currently, IP telephones have to depend on an
  external source of power
• IP phones must have both network connection and
  power connection.
• Several mechanism proposed to integrate power
  with network connections.

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Summary (1)

• IP telephony or VoIP refers to the transmission
  of voice telephone calls over IP networks.
• Hot area both in research and market because of
  low cost
• Challenge in backward compatibility with PSTN
• The complexity of IP telephony is on signaling.
  Both ITU and IETF propose signaling standards.
• H.323, by IUT
• SIP, by IETF, offering similar functions to
  H.323, but simpler than H.323.
• Both are competing to be recognized as 1
  signaling protocol

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Summary (2)

• H.323 uses a set of protocols for call setup and
  management
• SIP uses a set of servers to handle various
  aspects of signaling
• ENUM maps an E.164 telephone number into a URI
  (usually SIP URI)
• TRIP provides routing among IP telephone
  administrative domains
• IP telephones depends on external power, while
  analog phones dont.