WebRTC audio VS standard VoIP?

When you use webRTC audio, you are using the highest quality, lowest latency audio option. Audio from your mic is going directly to Oritor Conferencing switch, supporting opus48. If you are using standard VoIP, you may experience a delay, depending on the network latency from the speaker to the Oritor server, when using the built-in VoIP in Oritor (clicking the headset icon).

Here’s a breakdown of the path for the audio packets, from your Oritor client, when using VoIP. When you speak, your audio is transmitted by the Oritor client to the FMS server using speex codec. It is then transmitted by the FMS server to the Oriotr switch where it mixes in the audio, then the resulting audio stream is sent back the FMS server, and transmitted back to the client.

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