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Method for carrying call audio and video with timing so playback stays smooth
An agent uses a browser based phone to make a call. The browser sends the call audio to the cloud contact center gateway and tags it for higher priority on the network. If office Wi Fi gets shaky, the app briefly buffers a little more audio to keep speech clear and logs any missing packets for quality reports. When the agent presses keypad numbers, those tones are sent as small data events so menus still work.
Real time Transport Protocol is the standard way contact centers move live call audio and video. It adds simple numbering and timing to each packet so the listener can play sound in the right order and at the right speed. A partner channel sends back quality hints like delay and loss, which platforms use for dashboards and alerts.
In production it is commonly encrypted to protect conversations. Edge gateways in the provider network receive and forward the media, help with home and office routers, and keep the right paths open. Typical audio formats include Opus and G.711. Small buffers and error smoothing hide brief network hiccups. Many teams mark these packets for high priority so they get through busy links first.
For browser based agents, connection setup tools find workable paths on the internet and can fall back to a relay when needed. Correct handling reduces one way audio, clipping, and dropped calls in cloud contact center deployments.
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