Ever wonder how your voice travels seamlessly across the internet? The answer lies in a hidden hero: Session Initiation Protocol, or SIP.
SIP protocol orchestrates real-time audio and video, whether a phone call, video chat, or even a conference bridge.
Imagine it as the digital handshake that initiates and ends your online interactions, setting the stage for smooth and reliable communication. You’ll soon learn all about SIP protocol, including how it works at a technical level, its impact on businesses, and how it relates to Voice over Internet Protocol (VoIP).
What Does SIP Stand For?
SIP is short for Session Initiation Protocol (SIP), which is a set of online standards that facilitate real-time voice, video, and other media to work over any high-speed internet connection. SIP protocol is the foundation of Voice over Internet Protocol (VoIP) as well as other meeting and collaboration tools.
Let’s break down the SIP protocol:
- Session: This refers to a specific instance of communication between two parties, like a phone call or a video chat.
- Initiation: This refers to the start of the session. SIP is responsible for establishing the connection between participants and negotiating the parameters of the communication, such as audio and video codecs.
- Protocol: This is a set of rules and procedures that govern how devices communicate with each other. SIP defines the messages that are exchanged to establish, maintain, and terminate communication sessions.
Related: What is SIP Trunking? How it Works, Benefits & How to Get It
Brief history of SIP
In the 1990s, tech experts at the Internet Engineering Task Force (IETF) saw the need for a standardized way to manage online conversations. Drawing inspiration from existing protocols like HTTP and SMTP, they developed SIP, a simple and efficient framework for initiating, managing, and terminating real-time communication sessions.
Initially, SIP was primarily used for VoIP (Voice over IP) calls, allowing users to make voice calls over the internet. However, it quickly became apparent that SIP’s potential stretched far beyond phone calls. Soon, video conferencing, instant messaging, and even online gaming adopted SIP as their communication backbone.
The Internet Engineering Task Force (IETF) standardized SIP in 1999 in RFC 3261. On a technical level, SIP carries VoIP traffic over either UDP or TCP on ports 5060 or 5061. By comparison, browsing the web typically occurs over ports 80 and 443.
SIP use today
You’re probably already using SIP without even realizing it. Those crystal-clear VoIP calls you make? That’s SIP in action. But it’s not just for phone calls. Video conferencing, instant messaging, and even online games rely on SIP to keep things running smoothly.
To use SIP, you need a SIP phone, which can be a physical device or softphone on your computer or mobile device. Then, it’s just like making any other call or video chat.
Related: What Is PSTN and How Does It Work?
SIP Protocol Benefits for Businesses
Companies that are evaluating whether they should replace their old PBX or retrofit it with SIP trunking should consider the advantages that SIP-based communications offer.
- Cost-effective: Leverages existing internet infrastructure, reducing communication costs.
- Scalable: Easily adapts to growing user bases and demands.
- Unified communication: Integrates voice, video, chat, and other modes into one platform.
- Mobility: Seamlessly switch between devices and locations without interruption.
- Security: Follows standardized protocols and supports various security features.
Noteworthy SIP Protocol features
Media Negotiation
SDP (Session Description Protocol) ensures compatible communication by negotiating media formats like audio codecs and video resolution between participants. This allows devices with different capabilities to communicate smoothly, guaranteeing seamless audio and video quality.
Security
SIP uses SRTP (Secure Real-time Transport Protocol) and TLS (Transport Layer Security) to encrypt media content (voice, video) and signaling messages. This provides end-to-end security, protecting against eavesdropping, tampering, and unauthorized access, ensuring confidential and trustworthy communication.
Presence
SIP provides real-time information about users’ online status and availability. This enables features like call forwarding and chat notifications based on presence, allowing for efficient communication management.
Conferencing
SIP facilitates multi-party audio, video, and screen sharing sessions. This allows for effective collaboration and communication in meetings, webinars, and online classrooms.
Telephone calls made over SIP are relayed over to the traditional phone network, from a SIP service provider like Nextiva.
Related: Hosted PBX vs. SIP Trunking: Top 7 Differences & Why It Matters
How Does the SIP Protocol Work?
Session Initiation Protocol works with bidirectional communication. For every SIP message, one device sends a request, and the other device receives and later responds.
Responses are coded based on their message. Different preceding numbers in a three-digit sequence have different meanings.
For example, 1xx response codes mean the device received and is processing the message. Codes starting with 2xx mean completion, 3xx is used for redirections, 4xx is for authentication errors, etc.
The most common code is 200, meaning the action was completed successfully without further details.
A SIP request or reply is relatively short, with just a few lines explaining the details of the call.
What is the role of a SIP registrar?
While SIP messages can contact another party directly, they usually go through a SIP proxy server—kind of like a switchboard.
The SIP server handles SIP requests and directs them to individual users. From there, devices establish trusted communication with each other. Where does a SIP request come from? Most likely, these requests originate from a SIP phone or a VoIP app.
A SIP registrar is similar to an address book. It associates the various users with the access points on the IP network where one can reach them.
Notably, a user’s address is not an IP address, but a separate SIP address that resembles an email. This identifier is what allows multiple devices such as a smartphone or a desk phone to ring at the same time.
A related but different type of server is a redirect server. It works similar to the post office’s change of address function, where it forwards mail to a new location.
Like a registrar, a redirect server has a list of locations. But instead of making connections, the server sends a 3xx redirect message that indicates the site has moved.
Related: What Is IP PBX? A-Z Guide to VoIP Servers & IP PBXs
Does SIP use TCP or UDP?
Transmission Control Protocol (TCP) and User Datagram Protocol (UDP) are different ways to send data packets. Both methods are called transport protocols. This isn’t a trick question — it uses both!
SIP often uses UDP for most communication, relying on TCP only when reliability is critical. Some SIP configurations may also allow specifying which protocol to use for specific types of messages.
It’s important to note that the underlying media streams for audio and video calls typically use separate protocols, like RTP (Real-time Transport Protocol), which ensures timely delivery of media data.
UDP
Used for most basic SIP signaling messages to set up and end calls. This allows faster transmission so calls can start quicker. However, UDP does not guarantee message delivery, so some messages (packets) may be lost.
TCP
Used for larger SIP messages or when reliability is critical. For example, TCP ensures the delivery of important setup information by resending lost packets. But TCP has more communication overhead than UDP, so that it may cause slower call setup times.
Differences between SIP and VoIP
If you’re researching how VoIP phone service works, you might be confused with all the acronyms.
Voice over Internet Protocol, or VoIP, is a set of protocols used for voice calling over the internet, called internet telephony.
Internet phone service uses VoIP, which also utilizes the SIP protocol.
Built into the core of VoIP is the SIP protocol, along with other open standards. You don’t have to choose between VoIP and SIP protocol, as they often come together. They function similar to HTTP and TLS; they are used in conjunction with each other to provide fast, reliable, and secure communication.
A more direct comparison would be between PRI and SIP, with PRI representing an older form of establishing a communication channel as SIP does.
Related: What is PRI?
Congrats, You’ve Now Mastered the SIP Protocol
Just like that, now you have a much better understanding of the SIP protocol and how it works. While technical, the concept is easy to digest.
Session Initiation Protocol is a critical set of standards to establish real-time digital communication. It helps two or more parties have a successful, productive interaction.
Unlike many of the complex protocols in telecommunications, SIP protocol is easy. It controls the beginning, end, channels, and users during the call. With this deeper understanding, you can fix common VoIP problems and prevent them from happening.
One of the most valuable features of the SIP protocol is trunking. Trunking allows you to provide SIP-based phone service to your PBX instead of completely overhauling your company’s phone system. It lowers costs and improves service. What could be better than that?
If you’re looking for SIP trunking in your business, consider Nextiva as your next SIP trunking provider.
U.S. News & World Report rated Nextiva as the best business phone service, making it the top choice for businesses across the United States.
