Our use of cookies 
What is Session Initiation Protocol (SIP) and how it works
Session Initiation Protocol (SIP) is the foundation behind VoIP calls, controlling how sessions are established, managed, and ended separately from voice and video data.
It orchestrates calls behind the scenes, ensuring one-to-one calls, meetings and video conferences can be carried out over a phone system.
This guide explains what SIP is, how it works across a session lifecycle, and where it fits within real-world business communication environments.
Contents:
- What is Session Initiation Protocol (SIP)?
- How Session Initiation Protocol works
- What types of communication can SIP support?
- Issues with SIP-based communication
What is Session Initiation Protocol (SIP)?
Session Initiation Protocol is a signalling protocol (i.e. a set of rules) that orchestrates IP-based video and voice call sessions (VoIP).
Devices joining a one-to-one call, meeting, or video presentation exchange SIP-based call control messages to initiate, manage (e.g., invite another participant, share screen), and terminate the call.
These SIP messages operate separately from the data streams that carry the actual media (voice, video, or messaging content). SIP does not transmit the media itself, but controls three key aspects of the session:
- Session initiation: Establishing a connection between participants.
- Session management: Handling changes during a session, such as holds, transfers, or user additions.
- Session termination: Ending the session and releasing resources.
SIP is widely used in structured business VoIP phone systems, where each endpoint is assigned a UK phone number (+44) and an extension to make and receive regular phone calls over the internet.
Communication and messaging platforms that operate independently of traditional telephony, such as Google Meet, Zoom, Microsoft Teams, and WhatsApp, typically do not use SIP as their primary signalling protocol.
Instead, these rely on web-based communication technologies optimised for browser and mobile environments and designed to work reliably across modern internet connections.
How Session Initiation Protocol works
SIP serves as the signalling protocol that controls the full lifecycle of a real-time communication session, from the moment a user initiates a call to its termination.
It operates at the application layer of the network stack and exchanges structured signalling messages between call endpoints (e.g., IP phones or softphone applications) over an IP network.
These messages coordinate how a session is created, how participants connect, and how the session behaves, while the media itself is transmitted separately.
1. SIP session initiation
A session begins when a user initiates a VoIP call from a device or application.
The endpoint generates a SIP request that includes details about the intended recipient and the session type (e.g., voice or video).
The request may pass through one or more intermediary systems, depending on the [network topology] and design, including:
- Session Border Controllers (SBCs): Secure and control SIP traffic at network boundaries (especially between organisations or carriers).
- SIP proxy servers: Route SIP requests between endpoints and networks.
- Registrar servers: Maintain records of where users/devices are currently located (e.g. IP address, availability), which other systems use for routing.
- Redirect servers: Instruct the sender where to forward a request next, allowing requests to be redirected to moving endpoints such as remote VoIP devices.
- Business VoIP provider platforms: Provide the underlying call-control environment (such as a hosted PBX or carrier infrastructure), handling routing, numbering, and external connectivity.
So when a user dials a number on a VoIP phone, the device sends a SIP request, which is routed across one or more intermediary systems before reaching the recipient to establish the session.
2. SIP session setup
Once the request reaches the destination, the receiving endpoint is alerted and decides whether to accept the session.
During this stage, SIP messages are exchanged to negotiate how the session will be established, including basic parameters such as the media type and device compatibility.
This exchange is commonly referred to as the SIP handshake, where both endpoints confirm they are ready to communicate.
If the call is accepted, both endpoints complete the handshake and prepare to exchange media.
3. SIP session management
During an active session, the actual media (voice or video) is transmitted directly between endpoints using separate protocols (typically, the Real-time Transport Protocol (RTP).
At the same time, SIP remains available to manage changes and control behaviour.
It allows endpoints and systems to modify the session dynamically without interrupting it, such as placing a call on hold, transferring it, or adding additional participants.
SIP also enables more advanced call handling by coordinating how sessions move between systems and services within a communication environment.
Here are some examples:
- Hold: When a user places a call on hold, a SIP message updates the session state without terminating the connection
- Call transfers: A call can be redirected from one user to another, such as transferring a customer to a different department
- Conference calling: Additional participants can be added to an existing session without restarting the call
- Interactive Voice Response (IVR) and call routing: Calls can be directed through automated menus or routing systems, such as IVR, before reaching a final recipient
- Virtual assistants and automation: Calls can be passed to AI-driven systems for tasks such as answering queries or triaging requests
In each case, SIP coordinates the session changes and routing between endpoints and systems, ensuring the communication remains continuous while its behaviour is modified.
4. SIP session termination
When either participant ends the call, SIP is used to terminate the session cleanly.
A termination message is sent to signal that the session has ended, ensuring that both endpoints release resources and stop transmitting media.
What types of communication can SIP support?
SIP is used as the signalling framework in a wide range of real-time communication sessions over IP networks, supporting different media types depending on the application and platform.
The most common types of communication include:
- Voice calls (VoIP): Real-time audio communication between two or more participants, forming the foundation of modern commercial phone communication.
- Video calls: Point-to-point or multi-party video communication, normally combined with audio.
- Multimedia sessions: Sessions that combine multiple media types, such as voice, video, screen sharing, and file transfer within a single interaction.
- Messaging and presence: Instant messaging and user availability status, commonly used in unified communications platforms (UCaaS).
- Conference calls: Multi-participant sessions where several users join the same call, either for voice or video communication.
Issues with SIP-based communication
While SIP is a universal protocol considered highly stable, it can lead to issues with signalling, compatibility, and security when interacting with network components.
The most common issues include:
- NAT and firewall traversal: SIP signalling can be disrupted when passing through firewalls or Network Address Translation (NAT) environments. This is addressed by using SIP-aware network devices (routers, switches) or traversal techniques that allow signalling and media streams to pass correctly between endpoints.
- Security vulnerabilities: SIP systems can be exposed to threats such as spoofing, registration hijacking, and toll fraud if authentication and encryption are not properly implemented. Enforcing strong authentication (passwords and multi-factor authentication), encrypting signalling and media, and monitoring for abnormal call behaviour.
- Interoperability issues: Variations in how vendors implement SIP can lead to compatibility problems between devices, platforms, and providers. This issue is rare when using industry-standard equipment and validating systems during a VoIP system installation.
- Signalling complexity: SIP messaging and session handling can become complex in multi-line phone system environments, or when multiple providers are involved. Simplifying call flows, reducing unnecessary routing layers, and centralising control help limit complexity.
- Session routing and infrastructure dependencies: SIP often relies on intermediary systems (such as provider platforms), which can introduce points of failure or misconfiguration. Resilience can be improved through redundancy, failover routing, and careful configuration of provider dependencies.
Session Initiation Protocol – FAQs
Our business VoIP experts answer the following commonly asked questions regarding SIP:
Is SIP the same as SIP trunking?
No, SIP is the signalling protocol used to control communication sessions. SIP trunking is a service that uses SIP to adapt a traditional business phone line (built for traditional PSTN or ISDN telephony) into a VoIP phone system.
What is the difference between SIP and VoIP?
VoIP is the overall technology that enables voice communication over IP networks, while SIP is one of the protocols used within VoIP systems to establish and manage those calls. SIP manages calls, while Real-time Transport Protocol (RTP) transmits voice data.
What is a SIP session?
A SIP session is a real-time communication exchange, such as a voice or video call or meeting, that is established, managed, and terminated using SIP signalling between participants.
What layer does SIP operate on?
SIP operates at the application layer of the network stack (Layer 7 in the OSI model), where it handles signalling and session control between devices and systems.
Back to blog
