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Repair Windows errors before they cause bigger problemsFix Now →Fix the driver behind crashes, sound loss and screen glitchesFind Drivers →An IP telephony signaling protocol is the set of rules devices and services use to set up, change and end calls over an IP network. It carries the control messages around a call: who is being contacted, whether they can take the call, and which session parameters both sides support. The best-known example is the Session Initiation Protocol (SIP). The other established standards family is ITU-T H.323.
What “IP telephony” means
ITU-T describes VoIP, also called IP telephony, as providing services traditionally offered by the circuit-switched Public Switched Telephone Network (PSTN) over an IP network. The services are primarily voice, and may include video and data. They come with familiar calling features such as conferencing, call forwarding, call waiting and call transfer.
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So IP telephony is a broad service category. A signaling protocol is one component that makes it work, not a synonym for the whole thing.
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What a signaling protocol does
Signaling handles the control side of a communication session:
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- identifying and locating the party being called;
- checking whether that party is available and able to take part;
- agreeing on compatible session parameters, such as media types;
- marking when the session is established, modified or terminated.
Signaling is separate from the media stream, which is the audio or video itself. The media may use other protocols, so a signaling protocol should not be described as the thing that carries the voice.
SIP: the session-control protocol most people mean
The IETF’s RFC 3261 defines SIP this way: “This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants.” The sessions it covers include Internet telephone calls, multimedia distribution and conferences.
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The five facets of SIP
RFC 3261 identifies five facets of establishing and ending communication:
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- User availability: determining whether the called party is willing to communicate.
- User capabilities: determining the media and media parameters to use.
- Session setup: establishing the session between the parties.
- Session management: including transferring and terminating sessions and changing their parameters.
SIP also supports name mapping and redirection, and it lets media be added to or removed from a session that is already running. RFC 3261 uses a SIP application on a PC calling a SIP phone over the Internet as an example of endpoints, which is why a SIP desk phone is a typical physical endpoint.
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- 3-way voice conferencing per port
H.323: a broader multimedia framework
ITU-T Recommendation H.323, “Packet-based multimedia communications systems,” covers terminals and other entities that support real-time audio, video and data over packet networks that may not guarantee quality of service. Audio support is mandatory; video and data are optional. The ITU-T database lists Version 8, approved 2022-03-16, as in force. Status can change, so check the ITU-T entry for the current version.
H.323 is an umbrella recommendation, not just a signaling message set. Its call-signaling procedures include call setup, capability exchange, establishing audiovisual communication, call services and call termination. ITU-T says H.225.0 call-control messages are used for setup. H.225.0 is separately described as covering call signaling protocols and media stream packetization for packet-based multimedia systems. That makes H.225.0 the piece with a call-signaling role inside H.323.
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SIP and H.323 compared
| Aspect | SIP | H.323 |
|---|---|---|
| Defined by | IETF (RFC 3261) | ITU-T (Recommendation H.323; H.225.0 for call signaling) |
| Nature | Application-layer control (signaling) protocol for sessions | Recommendation describing a whole packet-based multimedia communications system |
| Session types | Telephone calls, multimedia distribution, conferences | Real-time audio (mandatory), video and data (optional) |
| Call lifecycle | Locate, check availability and capabilities, set up, manage, terminate | Setup, capability exchange, audiovisual communication, call services, termination |
Neither is universally better. The choice depends on system requirements and what the equipment on both ends supports. No reliable adoption or market-share figure is available to say which is more widely deployed today, so treat any such claim with caution.
When the two must work together
A network with SIP devices on one side and H.323 devices on the other needs an interworking function. RFC 4123 (informational, 2005) lists requirements for a SIP–H.323 one. Its example functions include:
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- mapping call setup and teardown sequences between the two;
- registering and resolving endpoints;
- maintaining SIP and H.323 state;
- negotiating terminal capabilities;
- opening and closing media channels.
The RFC shows how the two can be bridged. It does not show how common such gateways are in current networks.
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