What Is SIP and How Does It Work?

Session Initiation Protocol, or SIP, is a text-based signaling protocol for initiating, managing, and terminating multimedia sessions over the Internet. It is extensively used in telecommunication networks like voice conferencing and VoIP business phone systems to enable voice, video, and messaging between users.

While SIP itself doesn’t carry the actual conversations, it handles all the behind-the-scenes work to connect your call. The actual voice and video data is carried by RTP (Real-time Transport Protocol).

In SIP, distinct request and response messages serve various purposes in the signaling process of media sessions. Request messages are used to initiate the action or ask for information, while response codes are used to provide feedback on the status of a request, including failure, success, or redirection.

Here are some popular terminologies and codes used for request and response messages during SIP workflow:
 

SIP sets up, manages, and terminates conversations on online calls using different terminologies, such as INVITE, 100 Trying, 180 Ringing, 200 OK, BYE, and more. However, as discussed earlier, SIP only manages the session and does not handle audio/video transmission.

Here’s the Step-by-Step Process of How the SIP Protocol Works:
 

1. Call Initiation (Starting the Call): The caller’s device, whether a VoIP phone, app, or softphone, sends an INVITE request to the recipient through an SIP server. This request contains detailed information regarding caller IDs, the recipient’s SIP address, and the type of session (voice, video, or instant messages).
    
2. Finding the Receiver (SIP Register and Proxy): Once the INVITE is sent, the SIP server determines the receiver’s availability. If available, the server checks whether the device is registered with the SIP server and forwards the call request to the recipient.
    
3. Call Setup (Recipient Responds): The recipient’s device accepts the request by sending SIP response codes, such as 100 Trying, 180 Ringing, and 200 OK, to indicate the status of the call. “100 Trying” means processing the call, “180 Ringing” indicates the phone ringing status, and “200 OK” confirms that the call is accepted.
    
4. Call Connection (Session Establishment): After receiving the 200 OK response from the receiver, the caller’s device sends an ACK (Acknowledgement) message. This establishes a direct media path between the two devices using RTP for voice and video transmission, while SIP handles the signaling.
    
5. Call Termination (Call End): When the call ends, either party sends a BYE message to terminate the session, and the other confirms with a 200 OK message. The SIP server processes this request, and the call is officially ended.

SIP’s core features are call setup and signaling. Other essential features include user registration, call routing, session management, tracking user availability, SIP trunking, protocol independence, and security features.

SIP routes call based on user availability, priority, and predefined rules. If the receiver is unavailable, the call can be forwarded to another device, voicemail, or even the next person.

For this, SIP initiates the call by sending an invite request to the server. The server checks the recipient’s location and routes the call accordingly, using a SIP Trunk Gateway if needed. If the recipient has multiple devices, SIP can ring them all or in sequence.

SIP is not only a call setup and termination protocol; it also provides advanced session management capabilities. Once the call is placed, SIP permits modifications throughout the session, such as adding people to a conference or changing media settings (such as switching from audio-only to video).

Moreover, you can place calls on hold or transfer them to another person/device with the help of SIP.

SIP facilitates real-time tracking by allowing devices to register their location and status (online, busy, or offline) with the SIP server. Using SUBSCRIBE request (real-time updates about the recipient’s status) and NOTIFY response (notify subscribers about changes in the requested user’s status) messages, you can receive status updates about your contacts.

If a VoIP user wants to call a regular landline or mobile phone, a Session Initiation Protocol can facilitate this with SIP Trunking. It replaces traditional phone lines by connecting the VoIP system to the Public Switched Telephone Network (PSTN).

With SIP Trunking, businesses can make and receive calls over carrier networks using a virtual phone system. Plus, you can easily add or remove phone lines on your existing system without extra hardware.

SIP is not limited to a single transport protocol; it works with other protocols, such as User Datagram Protocol (UDP) and Transmission Control Protocol(TCP). It uses UDP for real-time applications like voice and video calls, where minor data losses are acceptable. Likewise, it works with TDP for tasks requiring accurate data delivery, such as call setups and messaging.

SIP integrates strong encryption protocols and techniques to protect both signaling and media transmission. For this, it uses Transport Layer Security (TLS), which encrypts transmission messages during the call setup phase, preventing illegal access, eavesdropping, and data manipulation.

Additionally, it uses Secure Real-Time Transport Protocol (SRTP) to encrypt voice and video streams during the call, providing an extra layer of protection against potential threats.

SIP-based phone lines are cost-effective as they rely on the Internet, and they are more scalable, flexible, and reliable than traditional phone systems.

• Reduce Costs: SIP replaces traditional landlines with internet-based calls, reducing phone bills. Moreover, you do not need to invest in expensive hardware, and you only need a strong internet connection and a SIP phone system.
• Scalability: You can easily add/remove phone lines and services based on your preference without needing extra physical installations. To add or remove lines, simply access your SIP provider’s admin panel, navigate to SIP trunks, and add new phone numbers or remove existing ones.
• Flexibility: By using a SIP-enabled device, you can make and receive calls from anywhere, using multiple devices such as a VoIP phone, mobile app, or softphone. This mainly benefits businesses that have remote workers in multiple locations.
• Security: SIP supports data encryption with TLS and SRTP protocols to facilitate secure communications and protect your privacy and sensitive information.
• Reliability: When one server is down, SIP Session Initiation Protocol reroutes the calls to a backup server. This ensures continuous service availability and operations without major disruptions.

SIP is a signaling protocol that enables smooth multimedia communication over an IP network.By effectively managing the signaling for the internet-based phone system, it offers more flexible and cost-effective telephony solutions.

Modern phone systems like VoIP use SIP to offer additional telephony features and advanced security. Switch to SIP-based VoIP today for an advanced business phone system.

An example of SIP in action is a VoIP call made using a softphone or VoIP phone system. When you call these systems, SIP starts, manages, and ends the sessions.

Yes, you can have VoIP without SIP. Some VoIP systems use alternative VoIP protocols like H.323, MGCP (Media Gateway Control Protocol), or proprietary protocols.