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Audio/video synchronization issues

lisa 0 Reputation points
2025-11-20T10:14:58.6066667+00:00

Hello Microsoft Support Team,

We are developing a real-time interactive avatar using Azure. Everything works, but we consistently experience audio/video desynchronization, sometimes the audio starts before the avatar animation, and other times the video leads.

The WebRTC stream arrives correctly, but sync varies. We tried client-side buffering, but the results remain inconsistent.

Could you please provide guidance on how to correctly ensure AV synchronization, recommended buffering strategies, or any known limitations?

Best regards

Azure AI Speech
Azure AI Speech
An Azure service that integrates speech processing into apps and services.
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  1. Q&A Assist
    2025-11-20T10:15:19.0666667+00:00
    AI generated content. This question contains an answer created with AI from Q&A Assist. Learn more

    To address audio/video synchronization issues in your real-time interactive avatar using Azure, consider the following strategies:

    1. Buffering Strategies: While you mentioned trying client-side buffering, ensure that the buffering is dynamically adjusted based on network conditions. This can help mitigate variations in latency that might lead to desynchronization.
    2. Jitter Buffer: Implement a jitter buffer to handle variations in packet arrival times. The jitter buffer can help smooth out the playback by temporarily storing incoming packets and releasing them at a steady rate. However, be cautious with the buffer size, as too large a buffer can introduce additional latency.
    3. Synchronization Mechanism: Develop a synchronization mechanism that aligns audio and video streams based on timestamps. This can involve marking the start of audio and video frames and adjusting playback accordingly.
    4. Monitoring Network Conditions: Continuously monitor network conditions and adapt the audio and video quality in real-time. Use diagnostics to identify any fluctuations in network performance that could impact synchronization.
    5. Testing Across Different Environments: Test your application in various network conditions to understand how it behaves under different scenarios. This can help you identify specific issues related to audio/video sync.
    6. WebRTC Capabilities: Since you are using WebRTC, leverage its built-in capabilities for error correction and media handling to optimize your streams. WebRTC is designed to minimize latency, which is crucial for maintaining synchronization.

    By implementing these strategies, you should be able to improve audio/video synchronization in your application. Keep in mind that achieving perfect sync can be challenging due to varying network conditions, so continuous monitoring and adjustments may be necessary.

    References:

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