Ultimate Guide To Understanding VoIP Protocols And Standards
As a business owner or manager, you probably came across the term VoIP every time you looked up anything related to communications solutions. And you probably got the idea that VoIP protocols and standards convert your regular communication system into a highly efficient one.
So you probably know that VoIP protocols enable you to make and receive calls over the Internet instead of traditional lines. But you’ve probably wondered how a protocol can do that. And what a protocol is.
Well, even though VoIP communication is a complex process, it doesn’t have to be complicated. Let’s take a look at VoIP protocols and standards and see what they are and how they influence your communications.
What Are Protocols?
Protocols are essential for voice over internet protocol (VoIP) communication services. Simply put, VoIP is a method of transmitting audio and/or video data across the Internet.
But sending data with VoIP over the Internet is not as simple as attaching a file to your email. In fact, even that is made possible thanks to protocols.
A protocol is, simply speaking, a set of rules that computers use to define how they interact or communicate with each other. If you’ve ever witnessed a dial-up tone, you might remember the series of beeps and buzzes a computer would make before it connected to the Internet. Well, those buzzes and beeps were the computer’s way of “communicating” with the Internet, if you would.
Then, computers started communicating with each other using TCP/IP. You might have stumbled upon this term throughout your time using devices that are connected to the Internet. Just about every device that can go online relies on TCP/IP protocols to do so.
TCP/IP is a suite of protocols that enables computers to consistently send and receive messages over the Internet. To do that, the TCP protocol breaks down the content into smaller parts called packets and the IP protocol identifies your device, similar to how a specific address identifies your house.
And while the TCP/IP protocols are essential for online communications, VoIP uses other signaling protocols to establish connections between two or more devices and transfer audio and/or video data more efficiently than the standard protocol suite.
What Are VoIP Protocols?
Now that you understand what a protocol is and why it’s essential for data transmission, let’s take a look at some of the protocols used for VoIP communication. Even though there are a lot of protocols suites out there, there are only a handful of protocols that are vital for the vast majority of users.
Nowadays, the most popular protocols used for VoIP are Session Initiation Protocol (SIP) and H.323, which is why we’re going to examine these two in more detail.
SIP and H.323 are known as Intelligent End Protocols because they contain all the “intelligence” needed to find and establish a connection between your device (the local host) and the device you’re calling (the remote device). Both these protocols have been around for more than 20 years and they’ll likely be used for years to come because they’re very good at their job.
In recent years, H.323 has grown more popular than SIP, but that’s not to say that it’s objectively better. It’s just that more businesses use this suite. Let’s take a look at what these protocols do.
SIP - Session Initiation Protocol
SIP is the Internet Engineering Task Force (IETF) standard for multimedia communication. The IETF is a large international community made up of everyone who is someone in the networking industry, including researchers, operators, designers, vendors, and more. They are responsible for maintaining the functionality of the Internet. And this community selected SIP as the standard protocol suite for their audio, video, and messaging solutions.
It’s important to note that SIP is modular, which means you can change it around. You can designate your SIP deployment specifically for the type of data you want to transmit, whether the data is video, audio, or text-based.
As a result, VoIP and Instant Messaging (IM) service providers will use SIP for different aspects of their services, which is one of SIP’s strengths. SIP is not a one-size-fits-all solution, it’s a communication solution that’s tailored to specific problems.
SIP can transfer messages and files instantly because it indicates who is and isn’t online. It offers a wide range of functionalities and features that enable providers to offer top-notch services.
Lastly, it’s worth noting that SIP provides trunking capabilities that enable it to work great with a private branch exchange (PBX) system.
H.323 is the international standard for “packet-switched networks” (PSN) multimedia communication. PSN is a communication network that breaks down the data in small packets, groups it, and transfers it from a source to its destination. This type of communication includes Wide Area Networks (WANs), Local Area Networks (LANs), and the general Internet we all use.
It’s important to note that the H.323 is actually a suite of protocols comprising several standards: H.323, H.245, H.225.0, and H.460. This set of standards allows traditional phones to communicate over the public switched network (PTSN).
The H.323 protocol was specially designed to operate over IP networks. It focuses on real-time communication such as voice, video, and text-based messaging, so it’s perfect for service providers that offer voice and video conferencing software.
Nowadays, H.323 is the most famous protocol for voice and video over IP communication - basically your VoIP system - and it’s used by multiple companies that offer video conferencing solutions.
How Is SIP Different From H.323?
The H.323 protocol was based on the binary language, so it’s essentially a complicated series of 1’s and 0’s. On the other hand, SIP has a text-based format, similar to the HTTP format pretty much all websites use to show up on the Internet. A lot of the technology behind HTTP was used in the development of SIP.
But the differences between SIP and H.323 are bigger than that.
SIP was developed by IETF to improve the overall functionality of the Internet. The protocol suite was designed to add a modular layer to the Internet, to improve its flexibility.
H.323 was developed by the same organization responsible for building the PSTN, the networks you use for landline phones. The protocol suite was designed with video conferencing in mind, so it’s great at transferring voice and video data.
SIP is a flexible protocol suite. Most SIP phones will operate on any SIP network.
H.323 is generally a proprietary solution at this point. That’s why providers using this suite will require users to purchase specific phones. If you don’t use the right phone, you can’t access all the functions and features the protocol offers.
SIP is modular, so you can change it to suit your needs. That’s why SIP doesn’t require a specific type of data to be transferred. It can be used for anything from instant messages, presence indicators (allows you to see who is online and who isn’t), file transfers, voice, video, and more.
H.323 is based on the original PTSN protocol so it excels at voice and video communication. You can expect the kind of reliability you would get from a landline when using it. However, the protocol suite hasn’t expanded much beyond voice and video at this point.
SIP is flexible, so it offers a wide array of features and functionalities. While it’s less focused on voice and video than H.323, it’s still completely capable to provide top-notch VoIP services.
H.323 was designed with video conferences in mind, and it’s great at it. However, the protocol suite hasn’t been updated in more than 10 years now, and it doesn’t offer one of the most in-demand features - team messaging.
Other Popular VoIP Protocols
SIP and H.323 are the most common VoIP protocols, but they are not the only ones. There are a lot of standards and protocols out there, but the following are more popular than others.
The Media Gateway Control Protocol (MGCP) is a signaling protocol, also known as a call control protocol. MGCP mirrors the structure of PTSN and it’s used in some VoIP systems.
Telephony gateways are network elements that convert audio signals transmitted on the PTSN into data packets transferred over a LAN network or the Internet.
A call agent is an element used in VoIP to control the signaling communication between devices and deliver specific services to users. These are the elements that instruct your phone to provide dial tone and do the heavy lifting with functions like call control, endpoint registration, phone number switching logic, and more.
Alternatively called SKINNY, this protocol suite offers the same core elements as SIP or H.323 but without so many features and functions. SCCP is a Cisco-specific protocol developed for IP telephony but with video capabilities. SCCP employs a call agent, which adds to the complexity of its features. However, since SKINNY requires a call agent to access its features, it’s less suitable for implementations that work with using call agents.
This protocol suite was developed by Cisco as an alternative to H.323 and it uses the MGCP protocol to provide telecommunication functions and features across both the PTSN and modern packet solutions like your LAN or Internet.
Are New VoIP Protocols Coming Soon?
As we mentioned earlier, both SIP and H.323 have been around for more than 20 years. Now, that might not seem like a lot, but it’s the equivalent of millennia in the telecommunication industry.
H.323 came out in 1996, the same year as the first flip mobile phone, the Motorola StarTAC. You could say that the telecommunication industry changed a bit since then, so it’s only natural for the protocols it uses to transfer data to change alongside it.
And there’s a new protocol suite coming soon. The WebRTC, which stands for Web Real-Time Communication, is the latest collection of protocols and APIs that make real-time communications between your browser, your phone, and your computer apps possible.
WebRTC uses peer-to-peer connections, allowing you to establish the most direct connection with the receiver to date. In simple terms, WebRTC will allow you to make video calls and host video conferences directly from your browser or mobile phone, without having to install a third-party application.
The new protocol suite will also enable you to send voice and video data over IP networks with fewer restrictions and faster speeds. And that’s before we take into account the speed boosts provided by 5G connections that will take the capabilities of this protocol to a whole other level.
VoIP Protocols Represent The Future Of Communication
VoIP protocols are the future of communication. There’s no VoIP without protocols and regular telephones are becoming expensive for both the end-users and the service providers, so most will switch to VoIP sooner rather than later.
Old telephone lines and networks cost providers 13.5 billion per year to maintain, despite the fact that subscription numbers are dropping by the year. And you can already see the change.
In Europe, Estonia and Germany completed their transition to 100% VoIP communications in 2018, the Netherlands completed it in 2019, and Portugal is bound to complete its transition by 2020.
And that trend can be seen across the Atlantic as well. More and more businesses and individuals switch to VoIP every day. VoIP allows you to talk to anyone at a fraction of the cost of a traditional phone call, no matter where the receiver is located.
VoIP is scalable and reliable, so it’s an effective communication solution for businesses. And VoIP makes your business communications flexible and mobile. As long as you have a stable Internet connection, you can receive a call or join a conference, whether you’re at the office, in your car, or even skiing.
VoIP is the future of business communications, and VoIP protocols make it possible. So in some way, VoIP protocols and standards make the future a reality.