WebRTC began as an open source gift to the industry from Google in 2011. Now almost ten years later, WebRTC has gone through the hype cycle and is powering a new era in web communications. WebRTC’s value and potential have taken center stage during 2020, as web video communications became critical during the unprecedented global lock downs. From a general adoption perspective, WebRTC enables most if not all of the OTT web video conferencing services everyone around the world from elementary school students to senior executives now rely on for daily existence.
Developers can leverage WebRTC in a range of ways. One approach might involve working purely from standards documents and implementing home grown applications. On the other end of the spectrum one can sign up for a CPaaS service with pre-baked SDKs for web and mobile apps. Carriers and service providers perhaps stand in the middle, with many seeking to enable WebRTC web applications to interwork with their existing SIP based UC services.
From a standards perspective, the latest “Candidate Recommendation”” was published (as of the time of writing this blog) on September 24, 2020. It’s important to note that standards experts are still, after all these years and increasing adoption, discussing what will be version 1.0. The W3C working group presented an overview at their annual conference in May 2020 of where things stand. It is good to see that while there are still some challenges, WebRTC 1.0 is getting very close to finally finished.
The pandemic related, increased in video communications and now essential nature of WebRTC, has put the focus on two key requirements: the need for it to “just work” while at the same time adding neat features such as superimposing “funny hats“ and filters on people’s faces during video conferences.
Work continues on in:
- Interoperability: On the browser side interoperability about 75%, with good progress across Chrome, Firefox, Edge, and Safari.
- Security: Challenges remain with IP address and port scanning
- New use cases: Low level access to encoded and raw audio and video, and low latency streaming for gaming and audio conferencing.
WebRTC is now mature in adoption, even if still evolving as a standard, and without question essential. Its popularly requires deployments to be scalable like never before.
At Enghouse Networks, we help scale WebRTC applications and have done so for large enterprises, service provider and carriers since the beginning with our media servers and SBCs, such as the Dialogic PowerMedia XMS and Dialogic BorderNet Session Border Controller (BN-SBC). Media servers help WebRTC to scale by facilitating multi-point sessions, and applications such as Transcoding, Interworking, Broadcasting, Recording, Processing, and Machine-Interaction (For a deeper dive please check out a Dialogic sponsored white paper on the seven reasons why you would want to use server side media processing on this topic).
WebRTC reduces development barriers and allows almost anyone to develop “telecom services” using only browsers (and HTML5). But some carriers and service providers want to combine “traditional” telecom services with the new WebRTC services. For that the Dialogic BN-SBC can act also as WebRTC gateway, allowing interworking functionality (IWF) between WebRTC and SIP, or even H.323
Learn More
If you enjoyed my musings and would like to read more about our thoughts on the future of the Session Border Controller, we invite you to download our new Enghouse whitepaper “The SBC, From Here to Eternity.”
Let’s start a discussion on enabling WebRTC at scale for your real-time-communications applications!
