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Troubleshooting Call Quality
Choppy Audio

Choppy Audio

Choppy, distorted, or robotic audio during calls often points to packet loss or network jitter. These problems occur when audio data packets are delayed or lost while traveling across the network, causing gaps, static, or uneven playback.

Understanding the Issue

Good audio quality depends on a steady flow of RTP (Real-time Transport Protocol) packets. When packets arrive late, out of order, or not at all, your browser struggles to reconstruct smooth sound. Even small disruptions can make speech sound clipped, robotic, or intermittent.

Symptoms

  • Voices sound robotic or garbled
  • Short dropouts or silence during conversations
  • Audio lag or echo when multiple participants speak
  • Call quality fluctuates, especially during long sessions

Common Causes

  • Network congestion: Competing traffic from streaming, downloads, or other users.
  • Wi-Fi interference: Weak or unstable signal, often from distance or crowded frequencies.
  • Packet loss: Data packets are dropped between your device and the server.
  • High jitter: Packet arrival times vary too much for the audio stream to stay consistent.
  • VPNs or proxies: These can reroute or delay RTP packets, reducing quality.

Troubleshooting Steps

Follow these steps to improve audio stability:

  1. Run a WebRTC diagnostics test Go to the WebRTC Diagnostics Test Page.

    • Complete the reCAPTCHA.
    • Allow microphone access.
    • When prompted, record a short message and listen to playback quality.
    • You may skip camera access—video is not used by Revenue.io.

    As the test completes, a Log Output panel will appear on the right. Take a screenshot or save the results to a text file for future reference.

  2. Review Key Metrics:

    MetricRecommended ValueImpact
    Packet Loss< 3%Above this, expect dropouts or missing words.
    Jitter< 30 msHigher jitter creates uneven, robotic audio.
    Roundtrip Time (RTT)< 200 msDelays or lag appear when RTT is high.

  3. Switch networks: Move to a different Wi-Fi network or try a mobile hotspot. This helps isolate whether the issue is specific to your office or home network.

  4. Disable VPNs or proxies: These can block or reroute RTP audio packets, increasing latency or packet loss.

  5. Minimize other network traffic: Pause large downloads, streaming services, or cloud backups during calls.

  6. Use a wired headset: Bluetooth can introduce latency or intermittent connections.

  7. Check signal strength: If using Wi-Fi, move closer to your router or access point.

Getting Help

If you continue to experience choppy or robotic audio:

  • Take screenshots or save logs from your diagnostics test.

  • Share them with your Network or IT Administrator to check for network congestion or firewall restrictions.

  • If the issue persists despite good network metrics:

    1. Contact our Support Team
    2. Include your network test logs and describe when the problem occurs (time of day, call length, Wi-Fi vs. Ethernet, etc.).

Technical Insight

Packet loss and jitter usually stem from unstable or congested networks. WebRTC uses jitter buffers to smooth playback, but when disruption exceeds threshold limits, gaps become audible. Common root causes include:

  • Router QoS settings not prioritizing real-time traffic.
  • Overloaded Wi-Fi channels (especially on 2.4GHz networks).
  • ISP-level congestion or packet shaping.

If you experience consistent issues in one environment but not another, the problem likely lies with local network conditions rather than your device or the Revenue.io application.

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