WebRTC and SIP: What's Best for My Business?

Multichannel communications reached buzzword status years ago, but it's never been old news thanks to the ever-expanding list of channels. Today, those channels are increasingly real-time. This trend is partly driven by need — during the COVID-19 pandemic, companies adopted on average 3.5 new communication channels, with one in three rolling out interactive voice response (IVR) and live chat for the first time. Multichannel real-time communications (RTC) has also become more popular because it's become more accessible with the help of WebRTC technology and session initiation protocol (SIP).

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Intro

As your company's list of RTC channels continues to grow, so does your need to integrate the technology that powers them. Unless you want to invest in a robust, cloud-based unified communications platform that has already integrated everything you need, you'll need a backbone for your communications tech stack. That's where SIP and WebRTC come in. 

What is WebRTC? How does it differ from SIP? And which is the best fit for your business?

What Is SIP?

Let's start with the older form of technology first. SIP is a text-based signaling protocol that enables real-time communication online. The technology first emerged in the 1990s as a way for people to communicate using computers and later helped make voice over internet protocol (VoIP) technology possible. Today, SIP facilitates conversations across different types of devices and different apps. 

In its simplest form, SIP is computer code that establishes communication sessions, manages the signal throughout the conversation, and terminates the connection when your session ends. At its highest level, a SIP trunking provider gives you a multiuser, media-agnostic platform that you can use to build and integrate internal and external communications tools. Modern SIP trunking solutions play nicely with newer IP networks and legacy systems, letting you make and receive internet-based phone calls via traditional desk phones. In a nutshell, SIP trunking gives you the power to bring voice, data, and video into a single digital location.

How Does SIP Work?

SIP initiates, maintains, and terminates internet-based RTC by sending data packets between two devices, explaining who is trying to communicate with whom, what technology will be used, and what requirements the recipient's device must meet to receive the call. 

If the recipient answers the call, an SIP response message is triggered, sending back a data packet explaining their communication capabilities. If both parties send enough of the right information, a connection can be established and communication can begin.

Pros and Cons of SIP

SIP is format-agnostic, so it can facilitate any type of RTC via any internet-connected device or (with the use of SIP trunking) even legacy hard phones. Because SIP can identify different devices and link their communications capabilities, it enables multiple media types to coexist in a seamless communications channel. That's why it's commonly used in VoIP solutions, including unified communications and contact center platforms. 

SIP-based VoIP and video conferencing solutions typically cost less than traditional phone service, but it does require an investment in specialized equipment and software. It also requires a strong internet connection for true real-time interactions.

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What Is WebRTC?

SIP has long been the most common mechanism for establishing RTC, but WebRTC technology has become an increasingly popular alternative.

WebRTC is an open-source protocol developed by Google that facilitates RTC between web browsers and devices. It's what lets voice and video communications work inside web pages without the use of plug-ins, and you can integrate it into applications without the need for a browser.

This technology has been around for a decade and was quickly adopted by web developers. In 2021, it was deemed a web standard by both the World Wide Web Consortium and Internet Engineering Task, and it is currently available on all popular web browsers.

How Does WebRTC Work?

WebRTC technology is a set of APIs that allow browsers to access devices, including the microphone and camera. It also lets you send various types of data, including audio and video signals, text, images, and files. 

WebRTC is an open-source platform, meaning it's free to use the technology for your own website or app. For example, you can add click-to-call functionality on your website or social media pages, enabling customers to initiate a voice or video call with you from their browser to your device.

Not only can WebRTC power communications, but it also enables other real-time data sharing possibilities, from online gaming to virtual events to internet-of-things operations.

Pros and Cons of WebRTC

Because WebRTC is already built into browsers and business applications, there's no upfront investment, no software to download, and no hardware for your IT department to manage. Business apps and communications platforms that leverage WebRTC technology tend to be less expensive, more flexible and scalable, and easier to use than SIP-based solutions, making them more accessible to smaller businesses with few IT resources.

But WebRTC does have limitations. Applications that leverage it tend to use a lot of memory, which can cause problems on older, slower devices. It also doesn't play nicely with analog legacy technology.

When Should Companies Adopt Each Option?

When WebRTC first started gaining momentum, some experts predicted it would eventually replace SIP, but markets for both solutions continue to grow. That's because each technology has advantages and disadvantages, and some companies use both in their communications tech stacks. 

Which is right for your business? That depends on your current setup and future needs. If you have already invested in SIP and it's meeting your needs, WebRTC-based communications might not be necessary. With the right communications platform, you'll still be able to use new and emerging communications features and integrate other business software. If you want to continue using legacy technology such as desk phones and private branch exchange (PBX) systems, you'll need SIP trunking to integrate that technology with digital channels.

If you're building your tech stack from scratch, your team only communicates through digital channels, and everyone has relatively new devices, WebRTC can handle your communication needs at a lower cost and with less complexity than SIP. WebRTC can also do things that SIP can't — for example, enable you to embed communication channels in your website or applications. 

That's what industry-leading education tech provider, Top Hat, did. During the pandemic, the need for an internet-based education platform skyrocketed, and TopHat implemented Vonage Video APIs to help meet this demand.

"By building Vonage's Video APIs into our digital courseware platform, Top Hat has effectively supported educators to deepen student engagement and enable human connection during a time when student isolation, anxiety, and motivation has become a top concern," said Nick Stein, the chief marketing officer at Top Hat. "Whether being used in large-class settings or one-to-one office hours, the ability to connect with students using Vonage's video capabilities has helped students to feel more connected with their learning, with their instructors, and with each other."

Next Steps

Both SIP and WebRTC can provide a backbone for RTC, but either way, you need more than a way to open the lines of communication. You also need a way to have conversations across an ever-growing list of channels — from voice and video to live chat and social media. You can piece together third-party apps and integrate them into a communications stack or invest in a robust unified communications platform that includes them all and lets you leverage SIP or WebRTC depending on the needs of your business.

To learn more about RTC and Vonage Communications APIs, contact a Vonage representative.

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