SIP (Session Initiation Protocol) is a network signaling protocol used to establish, modify, and terminate communication sessions over IP networks. SIP is most widely used in IP telephony (VoIP), where it is used to set up voice and video calls between users.
SIP is primarily responsible for managing communication sessions: locating the called party, negotiating connection parameters, establishing the call, and terminating it. The voice or video itself is usually transmitted using other protocols, such as RTP, rather than SIP.
How SIP Works
SIP can be compared to the process of setting up a telephone conversation. Before users can exchange voice data, the system needs to determine where the called party is located, establish a connection, and negotiate the parameters of the upcoming session.
When a user makes a SIP call, their device sends an INVITE request. The request may pass through one or more SIP servers that help locate the recipient and route the call to them.
If the called party accepts the call, the parties confirm establishment of the session. Voice traffic is then typically transmitted separately using RTP (Real-time Transport Protocol). When the conversation ends, a SIP BYE message terminates the session.
SIP therefore manages the communication session itself, while media data can be transmitted using other protocols.
Main Components of a SIP Network
A SIP infrastructure may include the following components:
- User Agent (UA) — an end-user device or application, such as an IP phone or softphone.
- SIP Proxy Server — receives SIP requests and forwards them to other participants in the network.
- Registrar Server — registers SIP clients and stores information about their current location.
- SIP Gateway — enables communication between a SIP network and other communication networks.
- Session Border Controller (SBC) — operates at the network boundary and can provide security, control, and routing functions for SIP sessions.
In a small system, some of these functions may be handled by a single server or software platform.
SIP Addresses and Registration
SIP users can be identified by addresses that resemble email addresses, for example, sip@example.com.
A SIP client typically registers with a server and informs it of the network address at which it can currently be reached. This allows the server to determine where to route an incoming call for a particular user.
This approach makes it possible to use the same account on different compatible devices and enables communication between users regardless of their physical location.
SIP and Voice Transmission
A common misconception is that SIP directly carries voice during a telephone call. In a typical VoIP system, its role is different.
SDP (Session Description Protocol) is commonly used together with SIP to describe the parameters of the media session. It allows the parties to negotiate IP addresses, ports, codecs, and other parameters.
Once the connection has been established, the audio or video stream is typically transmitted using RTP (Real-time Transport Protocol). SIP, SDP, and RTP therefore perform different but interconnected functions.
Where SIP Is Used
SIP is primarily used in enterprise and carrier IP telephony. The protocol is used in IP PBX systems, softphones, contact centers, unified communications systems, and telecommunications provider infrastructure.
Using SIP, a company can connect employees’ IP phones to an on-premises or cloud-based PBX, set up internal extensions, and connect its telephony infrastructure to a telecommunications provider.
A SIP trunk is also commonly used to connect a corporate telephone system to a telecom provider. It allows multiple simultaneous calls to be carried over an IP connection and can be used instead of traditional telephone lines.
SIP is not limited to voice communications. The protocol is designed to manage multimedia sessions and can also be used for video communication, conferencing, and other communication services.