VoIP Protocols

H.323 An ITU standard protocol for interactive conferencing. H.323 was originally designed for multimedia in a connectionless environment, such as a LAN. H.323 serves as an umbrella of standards that define all aspects of synchronized voice, video, and data transmission. H.323 defines end-to-end call signaling.

Media Gateway Control Protocol (MGCP) A method for PSTN gateway control or thin device control. Specified in RFC 2705, MGCP defines a protocol to control VoIP gateways that are connected to external call-control devices, referred to as call agents. MGCP provides the signaling capability for less-expensive edge devices, such as gateways, that might not have implemented a full voice-signaling protocol such as H.323. For example, any time an event such as an off-hook condition occurs at the voice port of a gateway, the voice port reports that event to the call agent. The call agent then signals that device to provide a service, such as dial-tone signaling.

Megaco/H.248 A joint Internet Engineering Task Force (IETF) and ITU standard that is based on the original MGCP standard. Megaco defines a single gateway control approach that works with multiple gateway applications including PSTN gateways, ATM interfaces, analog-like and telephone interfaces, interactive voice response (IVR) servers, and others. Megaco provides full call control intelligence and implements call level features such as transfer, conference, call forward, and hold. The basic operation of Megaco is very similar in nature to MGCP. However, Megaco provides more flexibility by interfacing with a wider variety of applications and gateways.

Session Initiation Protocol (SIP) A detailed protocol that specifies the commands and responses to set up and tear down calls. SIP also details features such as security, proxy, and transport (TCP or User Datagram Protocol [UDP]) services. SIP and its partner protocols, Session Announcement Protocol (SAP) and Session Description Protocol (SDP), can provide announcements and information about multicast sessions to users on a network. SIP defines end-to-end call signaling between devices. SIP is a text-based protocol that borrows many elements of HTTP, using the same transaction request and response model, and similar header and response codes. It also adopts a modified form of the URL-addressing scheme used within e-mail that is based on Simple Mail Transfer Protocol (SMTP).

Real-Time Transport Protocol (RTP) An IETF standard media-streaming protocol. RTP carries the voice payload across the network. RTP provides sequence numbers and time stamps for the orderly processing of voice packets. In addition to voice packets, RTP can also carry streaming video packets.