Managing WebRTC voice and video services is slightly different than managing other Voice over IP services. WebRTC and VoIP differ in two key ways: WebRTC integrates with websites and web apps, while VoIP is usually a standalone service. Additionally, WebRTC services can be developed in-house or by developers in specific verticals, as opposed to being developed by vendors specializing in VoIP.
As such, keep these best practices in mind when managing WebRTC voice and video services:
Maintain browser compatibility. Browser updates will affect how your WebRTC services operate. These updates can disrupt services due to changes in the browser's behavior or its APIs. Make sure you test your server against beta versions of browsers and be prepared to upgrade your WebRTC deployment whenever problems occur. Browser vendors won't stall their upgrade cycles to avoid disruptions to your service.
Monitor your service end to end. Test the CPU load, memory, storage and network use of your servers. WebRTC services for voice and video usually contain several moving parts. Validating that each part works independently is nice, but if you aren't running real sessions through your service as a monitoring measure, then you aren't monitoring your service end to end. This may cause extended downtimes for your users.
Ensure your network's quality. WebRTC services can be deployed on premises, but they are usually deployed and consumed from the cloud. As a result, connectivity and geographic location of your users will affect the quality of the media experienced. Make sure to test and validate the quality of your service at your users' location and at the level of load you expect from them.
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