This site is operated by a business or businesses owned by Informa PLC and all copyright resides with them. Informa PLC's registered office is 5 Howick Place, London SW1P 1WG. Registered in England and Wales. Number 8860726.
The Business Value of Voice Quality Using WebRTC
WebRTC adds a new dimension to the voice quality mission. It is appearing in new IVR applications, omnichannel contact center desktops, and mobile applications. While WebRTC features new technologies for delivering high-quality experiences over a wide range of network conditions, voice communications remain sensitive to the devices, codecs, and networks in use.
- Conversation Dynamics -- Having a large number of participants in an interactive dialog can make it difficult to prevent people from talking over one another. How many times in a call does someone have to say, “No, you go ahead?” A good quality conference produces natural conversation dynamics. But, if one person on a multiparty conference is attached to a poor network, they can degrade the experience for all participants. This is why the next few items on this list are important.
- Timeliness -- Keep the round-trip time (RTT) low, as I wrote about in a November 2018 post, by setting a design goal of having users within 25ms of the communications platform that they’re using. Callstats.io can monitor the metrics that are important to timely conversation streams, including RTT, loss, and jitter.
- Annoyances -- Setting voice levels based on how loud or soft someone speaks, echo or feedback, not recognizing the person’s voice, and voice clipping can all result in people having to repeat themselves and listeners having to work really hard to follow the conversation.
- Effects of Transcoding -- While WebRTC supports the Opus codec, this isn’t always the best codec to use despite its robustness. For example, when calls are coming in on a SIP trunk using G.711, the additional transcoding to Opus results in call degradation.
- Troubleshooting Tools -- These passively capture real-time call metrics from browser endpoints and network devices to monitor quality and detect problems. Callstats.io correlates data from each participant in a call and visualizes it in a way that enables the user to identify problems quickly. Users can monitor top-level service metrics and easily drill down to individual calls. Artificial intelligence and machine learning automatically identify the root cause of problems and long-term trends.
- Synthetic Media Generation -- It uses the WebRTC API to send a media file to the far end and measure how it compares against the original. The callstats.io Smart Connectivity Test is a much more robust and accurate way to measure anticipated quality for a given network connection than just a mean opinion score. With this data, you can tune the voice systems for codec in use, length of jitter buffer, TURN server selection, network pipe, and so on.
- Real-Time API -- When incorporated into agent desktops and network management platforms, these APIs provide the ability to monitor voice quality in real-time. The API can be invoked immediately prior to initiating/answering a call so user expectations can be set for voice quality. This is useful for mobile and remote workers who may be connecting to wireless networks.
- Global Call Quality Baseline -- Inter-geography WebRTC test calling can help inform what a user is getting and whether that is above or below the mean score.
- Benchmarking -- Global maximums and minimums enable customers to leverage the wealth of data collected by callstats.io for all its customers and use it to compare with the performance of their own service.