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IP Telephony (VoIP). CSI4118 Fall 2005. 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

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IP Telephony (VoIP)

CSI4118

Fall 2005


Introduction 1 l.jpg
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 user’s location or IP address

    • Support servers:

      • Proxy Server: forwards requests from user agents to another location.

      • Redirect Server: provides an alternate called party’s location for the user agent to contact.

      • Registrar Server: receives user’s 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. sip:[email protected]


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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. sip:[email protected]

  • 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.arpa


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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 don’t.


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