Like just about any other technology, WebRTC has followed the hype cycle pretty closely. (Were you surprised to see a headline on this topic in 2018?) WebRTC, an open source project, aims to boost mobile apps and browsers to real-time speed using APIs. After reaching peak hype around 2013, we slipped down into the trough of disillusionment. However, at times it may feel like we’ve reached an even deeper level of disappointment with WebRTC.
After years of vendor and developer misalignment, it’s become clear that the promise of direct endpoint-to-endpoint communication (eliminating the need for complex backend infrastructure) is more of a dream than a destination, even as cloud and SaaS collaboration tools evolve and gain enterprise adoption.
That is, unless you take a closer look at the actual WebRTC situation, according to Irwin Lazar and Nemertes Research. In fact, 28% of survey participants have WebRTC either supported or planned for web-based voice and video chat.
While plugin-free, browser-based peer-to-peer communication will be a boon for your end users, it may cause problems in your network.
Changing the WebRTC Vision
“No WebRTC-based services thus far challenged the dominance in the consumer space of Microsoft Skype, Google Hangouts and Apple FaceTime for voice and video calling, or have disrupted traditional UC vendors in the enterprise. WebRTC hasn’t eliminate dedicated softphone apps, nor has it yet led to widespread click-to-call implementations enabling website visitors to speak with companies without having to pick up the phone.” —Irwin Lazar, Nemertes Research
Needless to say, WebRTC hasn’t come to fruition the way you may have expected. And as a result, IT teams have optimized their network management and monitoring to accommodate traditional VoIP and video services.
At first glance, this may seem like it sets you up for success with WebRTC-based communications, too. But if you thought communications services seemed bandwidth-hungry and unwieldy before, just wait until WebRTC comes in and enables unlimited access to communication between employees, partners and customers across your network.
WebRTC isn’t coming in to replace UC/UCaaS vendors as people may have once believed. However, that doesn’t mean you can sit back and wait for it to burden your network. WebRTC demands a closer look at end-user experience monitoring.
The New(ish) World of WebRTC Visibility
Think about how you monitor your existing VoIP and video traffic. You deploy performance monitoring solutions on the wire of your backend communications infrastructure, set thresholds for MOS scores, voice loss, jitter and other metrics, and proactively address any quality concerns.
But what happens when that backend infrastructure is bypassed? Those reliable communications metrics become more of a mystery and you’re left with server monitoring as opposed to service monitoring.
The key to actually monitoring WebRTC services is to maximize visibility into the voice/video traffic being sent over this specific set of protocols. This means reevaluating your management, development and monitoring strategies to include QoS policy management as well as performance metrics specifically geared toward WebRTC.
The bottom line is that you can’t just sit back and let your existing VoIP and video planning account for WebRTC. That’s just asking for trouble, which you’ll get as soon as you see metrics that say connections are great and users are still complaining about call quality issues.
Make sure you have the means to not just monitor communications infrastructure, but individual VoIP/video packets as well. If you want to see what this kind of visibility looks like in action, check out this case study with a leading UCaaS provider.