What began as an open source gift to the industry from Google ten years ago is now powering a post-pandemic era in global web communications. Just over a year ago few people could predict that web video communications would come to play such a key role in work, education and socialising across so much of the planet.
During this rapid and essential shift, WebRTC has been a key enabler. It facilitates most web video conferencing services around the world, used by schools, universities and across working environments. Without webRTC, the shift towards remote working would have been more challenging.
How do developers leverage WebRTC?
WebRTC can be leveraged by developers in a number of different ways. One option is to work purely from standards documents and then implementing home grown documents. Alternatively, it’s possible to sign up for a CPaaS service which comes complete with ready-to-go software development kits (SDKs) for both web and mobile apps.
Between these two options, carriers and service providers might be to enable WebRTC based web applications to integrate with session initial protocol (SIP) based UC services.
WebRTC 1.0
While a number of challenges remain, WebRTC 1.0 is close to being finalised. The increase in video communications has brought a number of different issues to the fore, and the W3C working group presented a working overview at their last conference in May 2020.
The priority at the conference is that WebRTC 1.0 should “just work”. The much greater dependence on video conferencing services means that any glitches are likely to have a much greater impact than they previously would have done, so there’s an urgency in ensuring that they don’t happen.
Work is currently continuing to ensure reliable interoperability across the major browsers including Chrome, Firefox, Edge and Safari and some security challenges remain in relation to IP address and Port Scanning.
Development is still taking place to ensure low level access to encoded and raw audio and video, and low latency streaming for gaming and audio conferencing.
Making WebRTC scalable
WebRTC has now reached a mature stage of uptake, if not quite being completely essential. The key to the next stage of its deployment will be scalability, particularly going forward as temporary remote working and studying strategies develop into permanence.
Enghouse Networks has helped scale WebRTC applications since it was first launched. We’ve worked with large enterprises, carriers and services providers with our media servers and SBCs including Dialogic BorderNet Session Border Controller (BN-SBC) and Dialogic PowerMedia XMS.
Because media servers facilitate multi-point sessions they help make WebRTC scalable. They also facilitate applications such as Transcoding, Interworking, Broadcasting, Processing, Machine-interaction and Recording.
Because WebRTC reduces the barriers to development it makes it significantly easier 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
To find out more about our thinking on the future of the Session Border Controller,
we invite you to download our new Enghouse Executive Summary “Making the Connection: WebRTC During the Pandemic.”
