These apps showcase the power of WebRTC & SIP trunking when combined, enabling real-time comms across browsers and phone systems.
Sometimes it can be hard to understand what differentiates one method of voice communication over another. Here, we'll try and break down the main differences between two of the most popular protocols- SIP Trunking, and WebRTC (Web real time communications).
WebRTC is an open-source protocol specification that allows for real-time video and audio communications between web browsers and mobile applications.
is a means of operating phone systems over the internet, instead of using a traditional phone line, based on SIP for establishing and managing connections between users.
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The subtle differences between these two technologies mean that together, they unlock a staggering range of applications for the next generation of communications.
| Feature | WebRTC | SIP Trunking |
|---|---|---|
| Primary use | Browser-based voice, video, and data | Phone system connectivity over IP |
| Setup requirements | No plugins or downloads needed | Requires SIP-enabled devices or PBX |
| Best for | In-app calling, video chat, peer-to-peer comms | Business phone systems, PSTN connectivity |
| Media types | Voice, video, screen sharing, data channels | Voice, video, fax, messaging |
| Connection type | Peer-to-peer (browser to browser) | Client-server (device to carrier network) |
| Infrastructure | Works in any modern browser | Requires SIP trunk provider and configuration |
Most experts predict that WebRTC won't replace legacy VoIP infrastructure, but WebRTC applications offer easy peer-to-peer voice and video communication in situations where a standard phone call isn't optimal. To give you a sense of what WebRTC is capable of, and how it can be used, here are some applications that leverage WebRTC technology to deliver some awesome user functionality:
UPDATED 09/17/2020: Since this article was originally published way back in 2018, WebRTC has come a long way. With that in mind, we've updated our run-down with five NEW awesome apps that leverage WebRTC and SIP Trunking.
Want to know more about WebRTC SIP applications, and learn how you can use WebRTC in your application? Ask our experts how the best WebRTC apps are built on Telnyx.
WhatsApp was born as a simple messaging service, but, since being acquired by Facebook in 2014, has grown into one of the world's most popular platforms for real-time voice and video communications. The technology underpinning WhatsApp's native Android and iOS applications borrows heavily from WebRTC (such as the use of acoustic echo cancellation and active gain control from the WebRTC voice engine), but it also leverages SIP and associated technologies for faster call setup and more reliable communications.
WhatsApp launched a web client in January 2015, accessible through a limited number of modern web browsers. It was later discovered that this selective browser support mirrored the set of browsers supported by WebRTC, confirming suspicions that WhatsApp leverages WebRTC in its web client.
Aside from building on a WebRTC foundation with WhatsApp, Facebook has found no shortage of other uses for the technology in its broad range of app offerings. Indeed, Facebook's early rollout of WebRTC-based communications in 2015 caused a stir among the tech community as one of the largest bets to date on the emerging technology.
Facebook subsequently launched real-time voice and video calls - all powered by WebRTC - under its newly-standalone Messenger app in both mobile and web client form. Since that point, Facebook has continued to bet on WebRTC, rolling it out for co-broadcasting on Facebook Live (usurping the higher-latency RTMP protocol preferred by Facebook for single-host live streaming).
In 2010, Google purchased many of the codecs and echo cancellation components for WebRTC from Global IP Solutions. Google made modifications and took the technology to the IETF and W3C to get industry consensus. In 2011, Google released WebRTC as an open-sourced project. The Google Chrome team still maintains the WebRTC website.
Google Hangouts offers phone calls, SMS, video conferencing, and messaging capability all within the browser. There are other applications that are more popular than Google Hangouts, but Google's software is still a solid benchmark for demonstrating the scope and capabilities of WebRTC.
Google's landscape of video communications apps is unwieldy, to say the least. Google Meet, the Mountain View company's newest offering, is largely an extension of Google Hangouts, and as such, is based heavily on WebRTC. (In fact, it was originally named Google Hangouts Meet). Meet is Google's premium enterprise video conferencing tool, supporting more participants and real-time speech-to-text transcription, features not available in the consumer-focused Hangouts.
As if two Google video communications apps weren't enough, Google's stable also plays host to Duo, a native Android and iOS app for video calling. The app was launched in 2016 as a WebRTC-based competitor to Apple's ubiquitous FaceTime, but has seen low adoption in the intervening years. Nonetheless, the app's WebRTC foundations allow for more reliable peer-to-peer connectivity and default-on end-to-end encryption, garnering it a loyal user base.
Discord was originally created to serve the online gaming community and illustrates how WebRTC can easily accomplish the same tasks that were once performed using VoIP applications (see more WebRTC use cases). Discord is centered around group voice calls and uses WebRTC to support in-app messaging and enable users to add a supposedly unlimited number of people to calls.
Discord's engineers are not shy about revealing the technical details of their WebRTC implementation, either, writing an in-depth engineering blog on how their cross-platform voice architecture serves two and a half million concurrent users. Discord built signaling servers to establish and manage sessions using Elixir, a low-latency language widely used in telephony and communications systems (including right here at Telnyx).
GoToMeeting historically leveraged various different VoIP technologies, and integrated WebRTC into their application in the form of GoToMeeting Free, their browser client for joining GoToMeeting conferences. The browser option for GoToMeeting has historically been underutilized due to a large portion of their legacy userbase still using the non-WebRTC desktop client, but with the growth in videoconferencing popularity seen an uptick.
Group video chat app HouseParty flew largely under-the-radar for almost four years after its initial launch on iOS and Android. That is, until the coronavirus pandemic confined swathes of the world's population to their homes, limiting their social interactions and inspiring them to turn to new avenues for socializing with their friends and families. The app's informal user experience made it a viral hit and helped it to [top app store charts] across the globe.
Before the app experienced a surge in demand in early 2020, they were remarkably open about the technology underpinning their services - namely, WebRTC. The company published an in-depth engineering blog in mid 2018 discussing how they leveraged WebRTC to build cross-platform, real-time peer-to-peer video chat.
In early 2014, when it seemed that nothing could stop Snapchat's meteoric rise among younger generations, the company quietly acquired AddLive, a small WebRTC voice and video chat company. Snapchat almost immediately rolled out a video chat feature in its native mobile app, powered by WebRTC.
Amazon started out using Chime for their internal video conferencing and eventually rolled it out to their customers. Interestingly, Amazon Chime originally started with legacy video conferencing systems, so the speculation is that they made some in-house modifications to the WebRTC technology for their own purposes. Additionally, Amazon uses WebRTC to power parts of their wide-ranging services ecosystem, including Amazon Kinesis Video Streams used to securely stream video to AWS, and their Alexa home assistant's integration with smart home cameras and doorbells.
Some of these applications leveraged the flexibility of WebRTC to integrate real-time communications into existing infrastructure (like SIP trunks and other VoIP setups), whereas others built from the ground-up on browser-based real-time communications. Either way, all of these examples illustrate the immense progress that has already been made in real-time communications and hint that the WebRTC revolution is just beginning.
Telnyx WebRTC enables easier-than-ever integration of real-time browser-based communications into your existing SIP infrastructure. Our intuitive Javascript SDK, ready-to-run examples in vanilla Javascript, React and React Native, and quick-start guide leveraging JSON Web Tokens (JWTs) give you everything you need to get up and running with WebRTC in minutes.
What's more, our global, purpose-built IP communications network gives you the foundation to build WebRTC into the next generation of SIP and telephony applications, powering everything from Click-to-Call buttons all the way to complex Contact Center solutions.
With Telnyx WebRTC, it's incredibly easy to build an app like Google Duo with WebRTC, or imitate WhatsApp video calling with WebRTC. Talk to our experts about WebRTC SIP solutions for your WebRTC app today.
