Protocols

Voice over IP has been implemented with proprietary protocols and protocols based on open standards in applications such as VoIP phones, mobile applications, and web-based communications.

A variety of functions are needed to implement VoIP communication. Some protocols perform multiple functions, while others perform only a few and must be used in concert. These functions include:

• Network and transport – Creating reliable transmission over unreliable protocols, which may involve acknowledging receipt of data and retransmitting data that wasn't received.
• Session management – Creating and managing a session (sometimes glossed as simply a "call"), which is a connection between two or more peers that provides a context for further communication.
• Signaling – Performing registration (advertising one's presence and contact information) and discovery (locating someone and obtaining their contact information), dialing (including reporting call progress), negotiating capabilities, and call control (such as hold, mute, transfer/forwarding, dialing DTMF keys during a call [e.g. to interact with an automated attendant or IVR], etc.).
• Media description – Determining what type of media to send (audio, video, etc.), how to encode/decode it, and how to send/receive it (IP addresses, ports, etc.).
• Media – Transferring the actual media in the call, such as audio, video, text messages, files, etc.
• Quality of service – Providing out-of-band content or feedback about the media such as synchronization, statistics, etc.
• Security – Implementing access control, verifying the identity of other participants (computers or people), and encrypting data to protect the privacy and integrity of the media contents and/or the control messages.

VoIP protocols include:

• Session Initiation Protocol (SIP),^[4] connection management protocol developed by the IETF
• H.323, one of the first VoIP call signaling and control protocols that found widespread implementation.^[5] Since the development of newer, less complex protocols such as MGCP and SIP, H.323 deployments are increasingly limited to carrying existing long-haul network traffic.^[6]
• Media Gateway Control Protocol (MGCP), connection management for media gateways
• H.248, control protocol for media gateways across a converged internetwork consisting of the traditional PSTN and modern packet networks
• Real-time Transport Protocol (RTP), transport protocol for real-time audio and video data
• Real-time Transport Control Protocol (RTCP), sister protocol for RTP providing stream statistics and status information
• Secure Real-time Transport Protocol (SRTP), encrypted version of RTP
• Session Description Protocol (SDP), a syntax for session initiation and announcement for multi-media communications and WebSocket transports.
• Inter-Asterisk eXchange (IAX), protocol used between Asterisk PBX instances
• Extensible Messaging and Presence Protocol (XMPP), instant messaging, presence information, and contact list maintenance
• Jingle, for peer-to-peer session control in XMPP
• Skype protocol, proprietary Internet telephony protocol suite based on peer-to-peer architecture

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