What is SIP?

 It’s time to get to the bottom of SIP.

Today, SIP isn’t just a consideration for modern businesses. Companies rely on the Session Initiation Protocol to power conversations all around the world.

Unfortunately, there are still many people who don’t understand what SIP means, or what it stands for. After all, there are many complicated terms to understand and learn in today’s communication and collaboration landscape. From PSTN to CPaaS and DX, the industry has a language all of its own.

As the digital transformation of the communication landscape continues, pushing out old-fashioned concepts like ISDN and PSTN, no one can afford to underestimate the power and importance of SIP.

Defining SIP: Session Initiation Protocol

SIP, or Session Initiation Protocol, is a valuable tool within the telephony and communication landscape.

SIP or Session Initiation Protocol is a standard used in real-time communication sessions with video, voice, and even messaging component. Approved by the IETF (Internet Engineering Task Force) in 1996, and standardised by 1999, SIP promised to address the evolving expectations of IP communication.

SIP is one of the more flexible protocols created in history. It was designed originally as a general-purpose solution for setting up multimedia communication sessions in real-time between various participants. Notably, SIP doesn’t just transmit voice, although this is commonly an important part of its role. SIP can also transmit text and video data.

The Session Initiation Protocol is the technology that initiates and terminates a communication session within the modern world. For instance, when you start a video conference with members of your team, you’re using SIP to send that video and audio content between yourself and your colleagues. As a central protocol, SIP also requires endpoints to complete the transfer of data from one environment to another.

You can use SIP with standard telephones, softphones, and applications.

Where Did SIP Come From?

SIP was born in response to the demand for multimedia, mobility, and even interoperability in the communication environment. The SIP protocol complements other methodologies, like the Real-Time Transport Protocol or RTP used in IP sessions. When establishing sessions, SIP protocols establish:

  • Session management
  • Session setup
  • User capabilities
  • User availability
  • User location

SIP, as a protocol can be used to invited participants to multicast or unicast sessions that might not involve the initiator.

Importantly, the SIP protocol doesn’t provide communication services by itself. Instead, the protocol defines the implementation of interoperable SIP features that facilitate certain services.

SIP supports things like redirection and name napping services and can deliver mobility through a user resource identifier or URI. SIP can also be used for asynchronous solutions like message-waiting indicators, presence, and automatic call-back.

SIP and SIP trunking are rapidly emerging as must-have technologies in the current digital landscape. Companies that already have an on-premises PBX system can access SIP to take their PBX into the cloud and unlock new levels of flexibility and scalability. SIP allows for better collaboration, communication, and even system management in the business world.

However, many larger enterprises that have huge investments in existing phone technology have found it difficult to make the transition into SIP. This began to change recently when the 2020 pandemic pushed more companies to shift into the cloud for business communications, whether they were ready to do so or not.

How Does SIP Work?

The SIP protocol is very similar to SMTP and HTTP. It works in the application environment of a system’s open environment to create server/client architecture.

SIP is essentially a response and request protocol. When the protocol receives requests from clients and responses from servers, it can connect the right information on the backend to help with hosting important communication sessions. Devices that use SIP can communicate either through a SIP proxy or directly. Proxies work by offloading some of the tasks typically managed by SIP protocol.

SIP is also valuable for determining the end system to be used for the session, the media parameters, and the communication space. Once those parameters are assured SIP establishes the parameters at both ends of the communication and handles things like call termination and transfer.

SIP is an important component of SIP Trunking, which involves using a bundle of connections within a virtual environment to facilitate internal and external communication for a team. SIP trunks are the digital alternative to PSTN and ISDN, where you would connect through a PBX (private branch exchange), to the telephone network.

SIP trunks are made up of various SIP lines (individual branches) that use the SIP protocol to transfer data from one person in a conversation, to another. This information moves through the internet in a matter of milliseconds, allowing for high-quality communication clarity.

SIP trunking creates the connection between two domains in a Unified Communications network. By creating connections, SIP trunking allows users to segment networks into private and public domains. Private domains are connected to a personal server, where public domains are linked to an internet telephone service provider or ITSP.

How Does SIP Work in a VoIP Call?

SIP Calling is the process of sending voice call information through a SIP trunk or SIP line. Usually, the term “SIP calling” is used interchangeably with VoIP calling. However, it’s worth noting that there are some differences between SIP and VoIP.

In a traditional call, a phone system will feature a Private Branch Exchange (PBX), which is the on-premises system that manages calls. There will also be a Primary Rate Interface, which are lines that connect calls to the PSTN. The PSTN is the Public Switched Telephone Network, responsible for routing calls to a specific destination.

With SIP, you remove the need for PRI lines. SIP trunks use your internet connection to link the PBX to the PSTN, without any PRI lines at all. SIP also opens the door to various features like call recording, call attendees, and voicemail.

In a VoIP (Voice Over Internet Protocol) discussion, SIP works between two devices like a phone and a computer in the application layer. The SDP payloads in the conversation are transferred through a SIP message, while SIP works with the various protocols in VoIP.

Before voice is transported over your networks via SIP, it’s usually encoded using specific codecs. These codecs allow for voice information to be translated into binary data, so it can travel through the internet. Common codecs include the G.729 codec for compressed voice, or the G.711 codec for uncompressed voice.

SIP is just one tool for deploying VoIP. Another commonly referenced solution for managing VoIP conversations is RTP, the Real-Time Transport Protocol.

Encoded packets of audio data in a VoIP call are carried by the real-time transport protocol, otherwise known as RTP. The RTP is a special application layer protocol that allows for real-time audio and visual data. The RTP sessions work independently but parallel to SIP.

What are the Benefits of SIP for Communication?

So, why are so many companies turning to SIP for their communication stack?

Essentially, SIP works with VoIP to help people around the world communicate using computers, mobile devices, and more. SIP is an essential component of the internet telephony environment, supporting a rich and cost-effective global comms strategy. SIP also has the following benefits to offer today’s companies and communication environments:

Flexibility: Aside from just helping to set up a call, a SIP solution can also perform various other important call functions, like identifying the location of a user, determining the availability of a user, and more. SIP trunks can also support various forms of communication at once. Rather than just relying on voice, SIP can also allow businesses to access things like multimedia conferences, messaging with user presence information, and more. The enhanced flexibility of SIP cuts down the need for extra devices

Affordability: Switching to SIP communication can support enormous savings for companies, particularly in a global or growing environment. SIP supports cost savings in a host of crucial ways. For instance, with SIP, you can use VoIP and cloud calling to eliminate the demands of things like hardware purchases and maintenance costs. SIP also makes it easier to access a variety of different communication environments at once, reducing the number of subscription fees that you have. With SIP, you can also ensure that you’re future-proofed, reducing the risk of additional purchases in the short term

Scalability: Supporting a growing business can be an expensive and daunting prospect. Adding typical phone lines and waiting for those lines to be provisioned can really eat into your bottom line. Through SIP calling, it’s possible to buy a single channel at a time, and you can upgrade in a more convenient fashion. For businesses that are scaling rapidly, SIP offers a far more cost-effective strategy for evolution and growth

User-friendliness: While a lot of new technologies, including SIP, can seem complicated at first glance, the truth is that they’re actually easier to embrace than you might think. SIP, for instance, is extremely easy to implement as part of a technology stack. Maintenance is usually managed through an online control panel in the cloud. Additionally, when you’re ready to start using SIP, everyone only needs to learn how to use a single system instead of several

Better, more consistent calls: SIP systems can deliver exceptional call quality through HD voice and wideband audio, supporting the excellent conversations you need in your business. More importantly, when your office is outfitted with SIP calling, it’s easy to automatically reroute calls from an office phone to a mobile phone, making you more available to your customers. With SIP, you never have to miss an essential conversation

What’s the Difference Between SIP and VoIP?

As mentioned above, SIP and VoIP are often connected, but they’re not exactly the same.

VoIP is the general term for voice communication over the internet, SIP is a technical standard for voice and video communications over IP networks.

With SIP, users can access and send multimedia messages and send them to more than one person at once. SIP allows people across the globe to communicate through the internet using the computers and mobile devices that they prefer. Key elements of SIP include:

  • Capable of establishing, modifying, and controlling multimedia communications
  • Processes requests via a SIP proxy, ensuring that messages can be transmitted in their original format, such as video, voice, or instant message
  • Sends packets of information that can be encoded, containing video, data, and voice
  • Can perform independently without access to a computer
  • Uses SIP servers to manage the registration of SIP devices, including VoIP phones

Alternatively, VoIP:

  • Uses the internet in your company to make and manage calls
  • Uses a variety of technologies, including SIP and SIP trunking
  • Requires a connection to a computer from a VoIP phone
  • Can only send voice content via the internet
  • Can organize and relay phone calls on a network

Updating to SIP Technology

SIP trunking is the solution that many companies will use to replace outdated technology like PRI solutions. Staying on old-fashioned environments for your calling and communication strategy can be a dangerous decision in today’s digitally transforming landscape. Increasingly, companies of all sizes are beginning to switch to SIP as a way of preparing for the future of cloud flexibility.

The good news is that making the transition into the SIP environment is usually simple enough. If you have a PBX today, then you can build an entire voice service for your company using SIP Trunking. SIP trunking providers can set up entire ecosystems with the SIP protocol for your business, which support not only voice interactions but chat and video too.

SIP trunking and calling will reduce the overall costs of running your communication stack and make it easier to handle call information and communication data in a combined space. If you’re looking for the benefits of a cloud phone system that comes connected to a simple interface, then SIP could be the perfect transition for you.

 



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