What is SIP Protocol? A Simple Guide 2025
Publicado por JiangDavid en
The SIP(Session Initiation Protocol) protocol is a signaling protocol used to establish, modify, and terminate multimedia sessions. It primarily enables session establishment, user location, session management, and session termination. It is commonly applied in VoIP, IP broadcasting and intercom, video conferencing, and IoT communications.
What is SIP Protocol?
SIP (Session Initiation Protocol) is a way for devices to start, manage, change, and end live communication over the internet. It is used for voice calls, video meetings, text messages, and other media. SIP works at the application level of internet protocols and is common in VoIP and Unified Communications (UC) systems.
Why choose SIP Protocol?
Session Initiation Protocol (SIP) is a powerful communication technology that enables businesses to streamline their voice, video, and messaging services over the internet. Here’s why organizations should consider adopting SIP:
Cost Savings
SIP reduces the need for traditional phone lines, cutting down long-distance and international call costs.
Scalability
Easily add or remove lines as your business grows without the need for extensive infrastructure changes.
Flexibility
Works with a variety of devices, including IP phones, softphones, and mobile applications, ensuring seamless communication.
Reliability
Built-in redundancy and failover capabilities help maintain business continuity even during network issues.
Enhanced Features
Supports advanced functionalities like call forwarding, video conferencing, and instant messaging, improving overall communication efficiency.
Integration with Existing Systems
SIP can work alongside VoIP, cloud-based PBX, and other communication platforms, making it a versatile choice.
Components of SIP
User Agent (UA)
A device used to create and manage SIP protocol sessions.
Location Server
Records and manages the location information of the UA.
Registrar Server
Determines the user's location. The UA terminal registers with a registrar server, which assigns a specific address to the UA. This address has a lifecycle, and the UA must periodically refresh its registration status to stay active.
Proxy Server
Accepts session requests from the UA and queries the registrar server to obtain the recipient UA's address information. It then forwards the session invitation directly to the recipient UA (if within the same domain) or to another proxy server (if the UA is in a different domain).
How Does SIP Work?
SIP works by exchanging request and response messages between clients and servers to establish communication sessions. The process involves the following steps:
① User Registration
A terminal device (such as an IP phone or softphone) sends a REGISTER request to a SIP server (Registrar) to register its IP address and identity.
The server records the device’s address for future SIP call routing.
② Call Setup
The caller sends an INVITE request to a SIP proxy server, containing the caller's and callee's information, along with media parameters (such as codec details).
The callee acknowledges receipt of the INVITE and replies with 100 Trying (processing).
The callee's device rings and sends a 180 Ringing response.
When the callee answers, a 200 OK response is sent, confirming acceptance of the call and providing media details.
The caller responds with an ACK, confirming successful connection, and the call is established.
③ Media Transmission
SIP itself does not transmit audio or video data; media streams are typically handled by RTP (Real-time Transport Protocol).
SIP uses SDP (Session Description Protocol) to negotiate media details, such as IP addresses, ports, and codecs.
④ Call Termination
When either party wants to end the call, a BYE request is sent, and the other party responds with 200 OK, terminating the session.
Below is the entire process of SIP Calling:
Description of the common codes:
♦ ACK:Acknowledgment of receipt of the previous request.
♦ 100 Trying: Indicates that the request has been received and processing is continuing. It’s a provisional response, meaning the process is still ongoing.
♦ 100 Ringing: Sent when the called party’s phone is ringing, but the call hasn’t yet been answered. It’s another provisional response.
♦ 200 OK: This means the request was successful, and the call has been accepted. In response to an INVITE request, it indicates that the call is being established.
♦ 202 Accepted: Indicates that the request has been accepted for processing, but the processing is not yet complete. It’s more of an acknowledgment than an immediate result.
♦ 403 Forbidden: The server understood the request but refuses to authorize it. This could happen if the client lacks the necessary permissions or credentials.
♦ 404 Not Found: This response means the server could not find the requested resource, typically indicating that the requested URI is not valid.
♦ 407 Proxy Authentication Required: The client needs to authenticate itself with a proxy server. The server is asking for credentials before the request can proceed.
The Difference Between SIP and Traditional Telephony
Traditional telephone systems (such as PSTN, public switched telephone network) are based on circuit switching technology, relying on physical lines (copper cables, optical fibers) and dedicated hardware (such as PBX switches) to achieve voice communication. The core features are stability, high reliability, but high cost and single function.
The SIP protocol (representative of VoIP technology) is based on IP packet switching, transmits multimedia data such as voice and video through the Internet, and is software-based, with the advantages of flexible expansion, low cost and converged communication. The essential difference between the two lies in the technical architecture (circuit switching vs packet switching) and business model (closed hardware system vs open software ecosystem).
| Feature | SIP(VOIP) | Traditional Telephony(PSTN) |
| Connection Type | Internet-based | Circuit-switched |
| Scalability | Highly Scalable | Limited expansion capability |
| Cost | Lower | Higher |
| Multimedia Support | Voice , Video , Messaging | Primarily voice-only |
| Flexibility | Works on various devices | Limited to telephone lines |
SIP vs Other VoIP Protocols
Compared with other protocols (such as H.323, RTP/RTCP, WebRTC, etc.), the core role of SIP is to establish and manage sessions rather than directly transmit media data. It works with multiple protocols (such as RTP to transmit media streams, SDP to describe media parameters), and has become mainstream in Internet communications due to its lightweight and standardized advantages.
| Protocol | Purpose | Key Features | Typical Use Cases |
| SIP | Call signaling and session management | Text-based, lightweight, flexible, supports voice, video, and messaging | VoIP, IP telephony, video conferencing, WebRTC |
| RTP | Media transport | Transmits real-time audio and video streams, works with SIP and H.323 | VoIP, video streaming, real-time communication |
| MGCP | Gateway control | Centralized control of VoIP gateways, designed for integration with PSTN | Carrier networks, VoIP-PSTN interconnection |
| H.323 | Call signaling and multimedia communication | Early VoIP standard, complex, binary-based, requires more resources | Video conferencing, enterprise communication |
| WebRTC | Real-time communication in browsers | Low-latency voice and video communication, peer-to-peer, supports NAT traversal | Browser-based VoIP, video calls, online meetings |
SIP Application With SPON
Application 1
SPON IP Intercom and IP PA devices support the SIP protocol, allowing them to be used independently with third-party VoIP telephone system to receive two-way or one-way SIP calls.
Application 2
The SPON ICPAS Platform supports third-party SIP telephone access, enabling intercom and paging functions.
Application 3
The SPON ICPAS Platform supports integration with third-party VoIP telephone systems through SIP Trunking to enable two-way communication and one-way paging.
Conclusion
SIP is a powerful and versatile protocol that underpins modern IP-based communication systems. Its ability to handle voice, video, and messaging makes it an essential technology in VoIP, conferencing, and unified communications. While it comes with certain challenges, the benefits of SIP far outweigh its drawbacks, making it a preferred choice for businesses and individuals seeking efficient and cost-effective communication solutions.
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