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VII VoIP

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Transcript

  • 1. IP Telephony (VoIP)
  • 2. 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.
  • 3. 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.
  • 4. 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.
  • 5. 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.
  • 6. 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.
  • 7. 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)
  • 8. 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
  • 9. Interconnection with Others (2)
    • Media gateway: translates audio between IP network and PSTN.
    • Signaling Gateway: translates signaling operations.
  • 10. 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.
  • 11. 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.
  • 12. 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.
  • 13. H.323 Layering
    • H.323 uses both UDP and TCP over IP.
      • Audio travels over UDP
      • Data travels over TCP
  • 14. 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:bob@somewhere.com
  • 15. SIP Methods
    • Six basic message types, known as methods :
  • 16. 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.
  • 17. 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:smith@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
  • 18. 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
  • 19. 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)
  • 20. 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.
  • 21. 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
  • 22. 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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