Everything You Need to Know About SIP Protocol

What is SIP Protocol?

SIP Protocol can deliver more reliable business communications, but what is it, and how does it work? First off, it is worth mentioning that SIP stands for Session Initiation Protocol. It is a signaling protocol used to set up, connect and disconnect communication sessions (such as calls) between two or more parties. It is a well-known concept in IP telephony.

When a call takes place with SIP, it uses the internet to deliver communication sessions. This makes it a more reliable form of connecting. SIP calling uses SIP trunks or SIP channels to transmit a voice or video call.

While people often interchange the term with VoIP calls, the difference between SIP and VoIP is also worth noting: SIP calling uses VoIP to move analog call traffic over an internet connection.

Traditional business phone systems usually have two components:

1. The Private Branch eXchange (PBX), for call management, auto attendants, and voicemail.
2. Primary Rate Interface (PRI) lines to connect calls to the Public Switched Telephone Network (PSTN), where they travel to the destination telephone.

When the company switches to SIP trunks, the IP-enabled PBX connects to the data network instead of the PRI lines. That way, the voice traffic moves through the internet to connect to the PSTN. 

SIP Protocol for calling purposes is straightforward to use because it operates from a relatively basic interface that lets you manage the entire phone system (incorporate extensions, extra phone lines, and call routing). Businesses choose it as a way to enhance the powers of a conference call, for example. 

There are more benefits for companies that embrace SIP Protocol to communicate, particularly with the increase in remote workers. Let’s explore them in the following section.

RELATED: HOW DOES A SIP TRUNK WORK?

Why do Businesses choose SIP Protocol?

Communication is key to any business, so if you haven’t done your research already, you might want to consider the following benefits of installing SIP trunks in your company.

Availability and Reliability

This technology can ensure that your organization is always available for your customers and other external parties to contact you. Having responsive customer service and excellent communications within your business is crucial. A recent study found that 92% of consumers would stop purchasing from a company after three or fewer bad customer service experiences. 

SIP Protocol isn’t affected by factors that can break traditional phone lines’ service, such as bad weather interruptions (which usually take a long time to fix) or power outages. SIP trunks use the internet and can have backups available to ensure that you aren’t offline for too long, even if you lose service or connection. 

Cost 

SIP trunking eliminates the need for PRI lines and the associated cost. PRI lines contain 23 channels, but companies can purchase SIP trunks in increments as small as one channel (which equals one concurrent call). Businesses of all sizes can buy and pay for only what they need and quickly scale as capacity requirements change.

SIP can use analog adapters or “SIP-to-T1 gateways” in case your company wants to keep legacy PBX equipment and take advantage of lower telecom costs.

Variety of features

With SIP Protocol-based calls, users can make national (US) and international calls (to Canada and Europe, for instance). Premium providers will offer features such as call forwarding, email voicemail, and caller ID at no extra cost.

How does SIP Protocol Work?

If you’re interested in the more technical aspect of how this technology works, pay close attention to this brief breakdown.

SIP is an application layer protocol and is the foundation of interactive communication sessions over the internet.

It is important to understand that a protocol is a set of rules that defines how two (or more) devices (such as laptops, phones, routers, and network switches) communicate. Several protocols take place on the internet. 

For example, when you’re making voice calls online, many protocols work together to make the call possible. These multiple protocols build on top of each other in layers, which create a protocol stack.

Protocol stacks come in many shapes and sizes. The easiest one to understand is the Open Systems Interconnection model which consists of:

1. Application (SIP)
2. Presentation
3. Session
4. Transport
5. Network
6. Datalink
7. Physical 

The transport and the application layer of the protocol stack are the essential parts: 

• The transport layer controls the speed, reliability, and order of the data exchange. The data during a voice call is broken into packets and then transported over the internet. 
• The application layer specifies the protocols that the software applications need to communicate over a SIP network connection.

What Does Exactly SIP Do?

A common mistake that people make is that they believe that SIP trunking provides all services of their communication sessions. However, it doesn’t. SIP has two main functions, as we briefly implied above:

1. It sets up a call, conference, or any other form of communication sessions.
2. It terminates the session once it is finished. 

At the same time, there are five criteria involved in these SIP functions:

1. User location. This criterion tells the protocol where the end system is for the call.
2. User availability. It tells the protocol whether the called party is available to take the call.
3. User capabilities: This determines the media that will be used for the call.
4. Session setup: This involves the ringing from one user to the other.
5. Session management: Which runs the communication session, for example, the transfer and termination of the call.

RELATED: DO I NEED A SIP TRUNK?

SIP in Action: How Does SIP Work During a Voice Call?

Let’s break down SIP on this step-by-step call example:

Step 1: Voice information needs encoding with codecs before going out to the internet. This process translates the audio signals into data. There are two common types of codecs: G.711 codec and G.729. Encoded packets of audio data travel using a real-time transport protocol (RTP), which has its own protocol application layer to carry audio and video data in real-time. 

Step 2: Another protocol, called RTP control, then works with RTP to ensure that the information about the RTP packet delivery is being used and the quality of the voice service is good. 

Step 3: Protocols then transport the RTP and the SIP packets at the transport layer. This can either be done using a Transmission Control Protocol or User Datagram Protocol. User Datagram Protocol is better for transporting VoIP calls. 

Step 4: And finally, another application layer protocol works with SIP (because SIP is a media-independent application) known as the Session Description Protocol. This last step signals what type of media the SIP client is sending to the call, and the client receiving the call can support it.

Learn About SIP Protocol for Your Business

Now that you’ve learned more about SIP Protocol and how it works, why not introduce this technology to help your business communications? Installing SIP trunks into your company is easy to do and, with SIP.us, you can try it for free. Get in touch with us to learn more.