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  • 1.  Audio quality degradation in softphones using the WebRTC SDK

    Posted yesterday

    We have developed a softphone utilizing the `genesys-cloud-webrtc-sdk`.
    When making an external call using this softphone, we encountered a specific issue: if the external party's environment contained significant background noise, the audio from our operator (OP) side during the initial 10 seconds of the call would intermittently drop in volume or cut out, making it difficult to hear.
    After the initial 10-second period had elapsed, the audio from the OP side returned to normal; furthermore, the audio from the external party remained clear and problem-free throughout the entire call.
    Playback of the corresponding Genesys Cloud call recording files reveals the exact same audio anomalies.

    We tested the SDK's audio-related parameters using the following configurations:
    ・Automatic Gain Control: ON, Echo Cancellation: OFF, Noise Suppression: OFF
    ・Automatic Gain Control: OFF, Echo Cancellation: ON, Noise Suppression: OFF

    Notably, this issue does not occur when using the standard Genesys Cloud softphone-which operates directly within a web browser-as opposed to the custom softphone we developed.
    Furthermore, if we make a call using the standard Genesys Cloud softphone immediately before making a call with our custom softphone, the issue does not manifest. However, if we then proceed to make a second consecutive call using our custom softphone, the issue reappears.

    Is anyone familiar with similar occurrences or symptoms? Additionally, does anyone know of a potential solution to this problem?


    #EmbeddableFramework
    #PlatformSDK

    ------------------------------
    Masahiro Shioi
    なし
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  • 2.  RE: Audio quality degradation in softphones using the WebRTC SDK

    Posted 4 hours ago

    Hi Masahiro,

    This behavior looks like a WebRTC audio processing "warm-up" issue, especially related to AGC/VAD reacting to strong background noise at the start of the call.

    Key clues:

    • Only happens during the first ~10 seconds
    • Recorded audio shows the same issue → it's client-side processing
    • Doesn't happen after using the Genesys softphone → likely better initialization

    Some things you can try:

    • Force explicit audio constraints in getUserMedia (enable/disable AGC/NS/EC explicitly)
    • Always create a new media stream per call and fully stop tracks after each call
    • Reset any AudioContext if used
    • Add a short mic "warm-up" (2–3s) before starting the call

    Most likely cause is audio processing ramp-up or incomplete media reinitialization between calls.

    Hope this helps 👍



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    Cesar Padilla
    INDRA COLOMBIA
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