> ## Documentation Index
> Fetch the complete documentation index at: https://docs.sayvyai.com/llms.txt
> Use this file to discover all available pages before exploring further.

# Web Calls

> Browser-based WebRTC voice calling and low-latency audio

# Web Calls

*This page covers the technical implementation and architecture for low-latency WebRTC browser audio streaming in Sayvy AI.*

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## Latency Budget Breakdown (Sub-500ms)

*Content coming soon.*

* Audio packetization (20ms frames)
* Streaming Speech-to-Text (STT) pipeline
* LLM time-to-first-token (TTFT) streaming
* Incremental Text-to-Speech (TTS) chunking at sentence boundaries

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## WebRTC Transport & Audio Codecs

*Content coming soon.*

* Opus audio codec (24kHz / 48kHz)
* WebRTC Data & Media Channels configuration
* Jitter buffer management and packet loss concealment (PLC)

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## Browser Client Audio Lifecycle

*Content coming soon.*

* Microphone capture constraints (`echoCancellation`, `noiseSuppression`)
* Audio track initialization and handshake
* Network drop recovery and automatic session reconnection


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