What is Session Initiation Protocol (SIP)? A Complete Guide

As businesses expand, so do their communication requirements. Traditional phone systems frequently fail to scale with this growth, resulting in inefficiencies and elevated expenses. Session Initiation Protocol (SIP) provides a contemporary alternative, facilitating smooth integration of voice, video, and messaging services over the Internet.

This article will delve into the reasons why adopting SIP is a wise move for any organization aiming to improve connectivity and cut costs.

What is SIP Protocol?

The Session Initiation Protocol (SIP) is a protocol consisting of rules that enable devices such as phones and computers to initiate voice and video calls over the Internet. It functions as a common language that allows devices to locate each other, establish the call, manage the interaction, and terminate the call once finished.

Think of SIP as an Operator

Let’s use an analogy to help you better understand how SIP works.

SIP operates in a manner akin to a telephone operator. The role of this operator is to initiate calls, establish connections, oversee the call, and terminate the connection once the conversation concludes.

The SIP protocol utilizes a SIP server to initiate, manage, and terminate calls. Additionally, a SIP proxy server can manage call routing and serve as an intermediary for handling SIP requests and responses.

Using a SIP server, the SIP protocol is responsible for initiating, controlling, and ending calls.

Specifically, SIP handles the following tasks:

1. Answering Calls: The Session Initiation Protocol (SIP) processes the initial request when a call is initiated, similar to how an operator would respond to a phone call.
2. Establishing Connections: SIP locates and links you to the individual you wish to call, akin to how an operator connects your call to the desired party. This process may involve a redirect server that assists in routing the call to the proper destination using the IP address.
3. Call Management: Throughout the call, SIP manages various control signals including mute, hold, and transfer, akin to an operator overseeing different facets of the call behind the scenes.
4. Hanging Up: Upon completion of the call, SIP ends the connection in a manner akin to an operator hanging up the telephone.

Thus, SIP serves as the orchestrator and overseer of the entire calling process, while RTP (Real-time Transport Protocol) manages the transmission of the actual audio and video data.

How SIP Protocol Works

Understand how SIP works in under 2 minutes with this short video:

To understand the operation of the SIP internet protocol, consider the following example of a SIP protocol call flow:

Example of a SIP call flow.

• INVITE: Alice's phone issues an INVITE request to Bob's phone, signaling the intention to initiate a call.
• 100 Trying: Bob's phone receives the INVITE request and issues a "100 Trying" provisional response, signifying that the call setup process is underway.
• 180 Ringing: When Bob's phone begins to ring, it sends a "180 Ringing" provisional response back to Alice's phone, signaling that the call is alerting the user.
• 200 OK: When Bob picks up the call, his phone dispatches a "200 OK" signal to Alice's phone, signifying that he has accepted the call.
• ACK: Alice's phone sends an ACK (acknowledgment) to confirm the receipt of the "200 OK" response, thus completing the call setup process.
• RTP Media: The media data, such as the conversation between Alice and Bob, is transmitted using the Real-Time Transport Protocol (RTP).
• BYE: When Alice chooses to end the call, her phone issues a BYE request to Bob's phone, signaling the intention to conclude the session.

• 200 OK: Bob's phone sends a "200 OK" response to Alice's BYE request, confirming the successful termination of the call.

Common Uses of SIP Communication Protocol

SIP Trunking

SIP trunking replaces traditional telephone lines with a SIP-based connection, allowing businesses to make and receive calls over the Internet, often at reduced costs. It provides scalability and flexibility by enabling the use of existing internet connections for voice communication.

SIP trunking can also be connected to the Public Switched Telephone Network (PSTN) for broader reach.

Voice over IP (VoIP)

VoIP leverages the SIP protocol to facilitate internet telephony and calls over IP networks. SIP functions on ports like 5060 and 5061 to ensure secure communication. Devices enabled with SIP, such as SIP phones and softphones, register with SIP servers to initiate and control calls. VoIP offers an economical and high-quality voice communication solution, utilizing the internet instead of conventional telephone lines.

SIP Forking

SIP forking enables the distribution of a single SIP call request to several endpoints at once. It proves beneficial in situations where various devices, such as a desk phone, mobile phone, and softphone, are required to ring concurrently, thereby guaranteeing that the call can be picked up on any one of the devices.

Unified Communications (UC)

Unified Communications (UC) combines different communication methods such as instant messaging, video conferencing, and phone calls through the SIP protocol. SIP enables smooth interaction among these services, typically via SIP-enabled platforms.

Unified Communications (UC) systems employ Session Initiation Protocol (SIP) to manage internet sessions, thereby boosting collaboration and productivity. Typical UC applications encompass real-time presence information, desktop sharing, and integrated messaging, all of which operate on SIP protocol ports such as 5060.

SIP Paging

SIP paging facilitates the broadcast of voice messages to numerous recipients at once, akin to a public address system. It is widely utilized in settings such as schools, hospitals, and expansive office complexes for delivering announcements or emergency alerts.

Benefits of SIP and Key Differences from Other Protocols

Benefits of SIP

• Cost Savings: Utilizes internet calling to reduce expenses associated with traditional phone lines.
• Scalability: Easily adjusts lines and services as needed.
• Flexibility: Supports voice, video, and messaging.
• Mobility: Enables calls from any internet-connected location.
• Interoperability: It ensures compatibility and communication across various devices and services.
• Advanced Features: Includes call forwarding, voicemail, and more.

SIP vs. VoIP

VoIP refers to the wide-ranging technologies that deliver voice communications via the internet. SIP, a particular protocol within VoIP, is utilized to start, manage, and end sessions of multimedia communication, which can include voice, video, and messaging services.

VoIP stands for the broader technology enabling communication over the internet, whereas SIP is a protocol used to facilitate this communication. SIP provides advanced functionalities such as video conferencing and instant messaging, in addition to voice calls, rendering it a versatile and essential component of contemporary VoIP systems.

SIP vs. H.323 & RTP

When discussing internet calls or video chats, protocols such as SIP, H.323, and RTP frequently come into play.

H.323 is a suite of protocols established by the International Telecommunication Union (ITU) for voice, video, and data conferencing across packet-switched networks. It offers an extensive framework for multimedia communications, although its usage has diminished in modern times.

In contrast, SIP is a simpler, more flexible protocol, which has led to its wide adoption in modern VoIP systems. SIP's ease of implementation and integration with other internet technologies stands in stark contrast to the older and more complex H.323 protocol.

RTP is fundamentally distinct from both SIP and H.323. Unlike signaling protocols, RTP is responsible for managing the delivery of audio and video data during a call.

To summarize, SIP and H.323 protocols manage the setup and control of calls, while RTP is responsible for the real-time transmission of media streams.

SIP vs. SDP

The Session Description Protocol (SDP) is a format used to describe the multimedia elements of communication sessions, detailing the media type, codec, and network information. SDP is specified in the standard document RFC 4566.

SIP and SDP frequently collaborate to set up and control multimedia communication sessions. SIP manages the signaling, while SDP describes the multimedia content within these sessions.

Wrap-Up

SIP technology is increasingly adopted across diverse industries due to its flexibility, cost-effectiveness, and advanced features.

VoIP phone systems utilize SIP technology to provide crystal-clear voice communication, advanced call features, and seamless integration with various communication tools.

Businesses looking to modernize their communication infrastructure will find that Connect3 offers a range of advanced SIP-based PBX and VoIP solutions. Contact Connect3 today to discover how their innovative products can enhance your business communications.