I’m a network engineer optimizing voice quality for a pool of remote agents. Many of them live in rural areas and rely on 5G cellular modems or even Starlink, which means our latency is acceptable but our packet jitter is all over the place (sometimes spiking to 150ms).
We are experiencing severe audio clipping and dropouts on their WebRTC phones. I know that increasing the jitter buffer on the endpoint can smooth this out, at the cost of adding a bit of delay to the conversation (which we are fine with).
Is there any way via the API, the WebRTC Phone base settings, or the Edge Trunk configuration to statically increase the jitter buffer for a specific group of users? The default dynamic buffer seems to be too aggressive and drops packets instead of buffering them during spikes.
Unfortunately, you can’t manually configure the jitter buffer for the Genesys Cloud WebRTC client. The WebRTC implementation in modern browsers (Chrome/Edge) handles the jitter buffer entirely within the browser engine itself. The browser’s audio pipeline dynamically adjusts the buffer based on network conditions, and Genesys doesn’t expose a dial in the Admin UI to override Chrome’s internal logic.
I’ve been dealing with this in healthcare where we have mobile clinics on cellular connections.
Since you can’t change the buffer, the best mitigation is to change the codec payload size. Go into your WebRTC Phone Trunk settings under Telephony. By default, the Opus codec uses a 20ms packet time (ptime). If you change the ptime to 40ms or 60ms, you are sending fewer, larger packets. This reduces the overall packet rate, which can significantly help cellular networks process the stream more reliably and reduces the burden on the browser’s dynamic jitter buffer.
I inherited an org with exactly this problem 6 months ago. Another thing to check: Disable ‘Comfort Noise’ and ‘VAD’ (Voice Activity Detection) on the WebRTC trunk if you haven’t already.
When VAD is enabled, the browser stops sending audio during silence. When the agent speaks again, the stream has to ‘ramp up’, and on high-jitter networks, the first few syllables get eaten by the browser’s buffer while it figures out the new latency baseline. Forcing a continuous audio stream keeps the browser’s dynamic buffer ‘warmed up’ and prevents those initial dropouts.