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Voice over Internet Protocol transports voice data over a digital, packetized Internet Protocol connection. Voice over Internet Protocol (VoIP) was created to allow voice communications devices (terminals) to communicate over the Internet. Analog voice data is encoded as digital data and the resulting voice data is inserted into IP datagrams. VoIP is actually a suite of protocols created to support this function and there is more than one suite that can be used to provide this type of service. Voice communication is highly delay sensitive (that means any delay in transmitting part of the conversation is immediately detected by the user); therefore, quality of service protocols are also used to improve the quality of the communications path by prioritizing traffic and managing the available bandwidth.

VoIP, as a service, runs on top of Internet Protocol. VoIP clients (terminals) communicate with VoIP gateways, calls are routed between VoIP gateways and back to other remote terminals. VoIP is dependent upon working end-to-end IP connections between gateway and all clients involved in a session.

Protocols Used in Voice over IP

There are several protocol stacks used to provide VoIP services, SIP, H.323, MGCP and MEGACO/H.248.

  • JOINT STANDARD: Media Gateway Control Protocol (MGCP / Megaco / H.248)
    • IETF RFC 3015 - Megaco Protocol v1.0
  • IETF Standard:  Session Initiation Protocol (SIP)
    • IETF RFC 3261 - Session Inititiation Protocol (SIP)
  • ITU-T Standard:  H.323 (Multimedia conferencing over LAN) Packet-based multimedia communications (VoIP) architecture
    • Call Control and Signalling
      • H.225.0 - Call Signaling and RAS in H.323 VOIP Architecture
      • H.225.0/RAS -
      • H.245 - Control Protocol for Multimedia Communication
    • Audio Processing
      • G.711
      • G.722
      • G.723.1
      • G.728
      • G.729
    • Video Processing
      • H.261
      • H.263
      • H.264 / MPEG-4
    • Data Conferencing
      • T.120
      • T.121
      • T.125
      • T.126
      • T.127
    • Media Transport
      • Real-time Transport Protocol (RTP)
      • RTP Control Protocol (RTCP)
    • Security
      • H.235 - Endpoint Security and Encryption
    • Other Supplementary Services
      • H.450.1 - Generic control
      • H.450.2 - Call transfer
      • H.450.3 - Call diversion
      • H.450.4 - Call hold
      • H.450.5 - Call park and pick up
      • H.450.6 - Call waiting
      • H.450.7 - Message waiting indicator
      • H.450.8 - Names Identification Services
      • H.450.9 - Call completion services


Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP) is an IETF VoIP standard that traces its origin to Cisco and Telcordia proposals. Session Initiation Protocol (SIP) is a transaction based protocol that operates at the application layer and controls the creation, modification and termination of multimedia and VoIP sessions with one or more participants. SIP clients initiate requests to the server, and the server receives, then processes the client request, and then sends a response, completing a transaction.

ITU-T H.323

The H.323 protocol

Media Gateway Control Protocol (MGCP / MEGACO / H.248)

The Megaco protocol supports multimedia communications used between elements of a "physically decomposed multimedia gateway", which is a gateway in which the gateway's components distributed across multiple computer systems.




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