Session Initiation Protocol (SIP) — Signalling Protocol for Internet Telephony

What is Voice Over Internet Protocol (VoIP)?

VoIP is an abbreviation for the Voice over Internet Protocol, which specifies how to make and receive phone calls over the internet. Since the late 1990s, telephony has depended on digital lines to carry phone calls. VoIP is a low-cost method of handling an infinite number of calls. It is considered a tried-and-true technology that allows anybody to make phone calls over the internet. With the growth of broadband and has emerged as the clear choice for phone service for both individuals and companies. VoIP services have the ability to transform the voice of the users into a digital signal that can be transmitted over the Internet. If the user dials a standard phone number, the signal is transformed into a regular phone signal before reaching its destination. It also allows them to make a call immediately from the computer, a VoIP phone, or a standard phone linked directly to a specific adapter.

What is Session Initiation Protocol (SIP)?

The Session Initiation Protocol (SIP) is one of the main protocols in VoIP technology. It is an application layer protocol that controls multimedia communication sessions over the Internet in combination with other application layer protocols. It is essentially a signalling system used to create, modify, and end a multimedia session over the Internet Protocol. The endpoint can be a smartphone, a laptop, or any device that can receive and distribute multimedia information over the Internet. It employs a request/response model to establish communications between network components and, eventually, construct a call or session between two or more endpoints. Several clients and servers may be involved in a single session. It transparently provides name mapping and redirection services, allowing users to keep a single externally visible identity independent of network location.

Why SIP is Essential?

A protocol is a set of rules for communicating between pieces of hardware. Many protocols, such as SIP, TCP, and HTTP, are used in modern digital services. A protocol defines the syntax and semantics of communication. A protocol can only be useful if various vendors agree to utilize it on their devices and apps. Many protocols grow over time to become industry standards when more developers begin to use them.

SIP Sessions

The basic SIP specification includes a method for establishing and managing sessions between two user agents. SIP sessions, sometimes known colloquially as “calls” and more technically as dialogues, are initiated by invitations from one User-Agent (User Agent Client or UAC) to another (User Agent Server or UAS). This invitation transaction is essentially a three-way handshake between the UAC and the UAS. To establish a call session between different users, the SIP Sessions are used with the VoIP and Voice and Video over IP (VVoIP or V2oIP).

SIP Architecture

  • User Agent Server (UAS)
  • Proxy Server
  • SIP Terminal
  • Location server
  • Redirect Server

SIP Call Flow

  • Basic SIP session setup involves a SIP UA client sending a request to the SIP URL of the called endpoint (UAS), inviting it to a session.
  • The request can be sent if the UAC knows the UAS’s IP address. If the user cannot be found, the UAC redirects the request to a proxy or redirect server.
  • The request may be sent to various servers until the user is found.
  • The request is made to the UAS after the SIP address is resolved to an IP address.
  • If the user accepts the call, capabilities are discussed and the conversation begins.
  • If the user does not answer the phone, the call might be redirected to voice mail or another number.

Why SIP Should be Preferred?

The primary goal of SIP is to start and then finish the session. SIP answers inform you of the other party’s existence, establish a connection, and allow you to do anything you need via the connection. However, it has no understanding of what is going on over the connection, which allows SIP to be used for video conferencing, instant messaging, and making calls over the internet. This is how SIP works in a corporate VoIP call: before voice data can be delivered over the internet, it must be encoded with codecs that convert audio impulses into data.

SIP Gateway

In VoIP communication solutions, a SIP gateway, also known as a SIP Server, is required. It is a device that handles device registrations and connects devices such as desk phones, conference phones, and softphones. SIP gateways allow audio and video connections to be made over the internet at the same time. A SIP gateway is required if you wish to use conference calls, voicemail, or video calls.

SIP Trunking

SIP trunking is a means of providing phone services to businesses that have their own IP PBX. It takes the place of the classic PRI line between organisations and regular phone companies. It acts as a link between VoIP and the public telephone network (PSTN). A SIP trunk is a virtual link between your company and your ITSP rather than a physical line (Internet Telephony Service Provider). The seller is only responsible for the connection and supplying you with a dial tone. You must handle the PBX on your own. That is, your team will determine which features to activate and administer. You have complete control over when and how your hardware and software are upgraded. You are in charge of data security and privacy.

Advantages of SIP Trunking Over VoIP:

  • SIP is very scalable, and it is not only restricted to speech; it can also be used for video, messaging, and other purposes.
  • For better efficiency, it frequently contains built-in interface with regularly used applications.
  • Can be used with PRI lines to provide the greatest combination phone system for your company.
  • Pricing is quite flexible, allowing for more features and lines as needed.


A straight comparison is not always possible when it comes to SIP versus VoIP technologies. Whereas VoIP is used to describe any internet-based phone service, SIP is a set of communication protocols used in most VoIP implementations. VoIP refers to any phone call made through the internet rather than traditional phone lines. VoIP systems rely on data connectivity rather than the public switched telephone network (PSTN) to transport voice packets. Unlike SIP, which is used to support and expand Voice over IP, not all VoIP is supported by SIP technology. While SIP is only one protocol that may be used in a corporation VoIP to expand communications beyond voice-only calling to allow video conferencing, text, instant messaging, and other multimedia communications, it is not the only one.

Patent Analysis

Future scope of SIP Trunking

SIP, being a cloud technology, is open to continual enhancement as connection, PBX, and network technologies advance. When improvements are made at any level, you should reasonably expect them to flow through to your current situation. SIP is already the most popular protocol for handling audio, video, and instant messaging sessions. As video continues to gain traction in business communications, stronger codecs will be used to guarantee video conferencing quality is maintained. If the accompanying bandwidth and network constraints are met, SIP will continue to be the communications protocol for business interactions. Thus, it is quite evident that SIP trunking is future proof and will be utilized by more and more organizations to come.




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