Real-time Voice
Supports ultra-low latency and ultra-high quality audio calls globally, providing developers with easy-to-integrate, highly reliable, and multi-platform interoperable audio/video services.
Product Introduction
Overview
Product Features
Real-time Voice Pricing
Server-side Stream Mixing Pricing
CDN Live Streaming Pricing
Restrictions
Differences between Real-time Audio/Video SDK and Real-time Voice SDK
Quick Start
Run Example Source Code
Integrate SDK
Implement Voice Call
Scenario-based Audio/Video Configuration
Communication Capabilities
Using Token Authentication
Call Quality Monitoring
Network Speed Test
Multi-Source Capture
Publishing Multiple Streams Simultaneously
Supplemental Enhancement Information (SEI)
Traffic Control
Cloud Proxy
Geo-fencing
Game Voice
Mass-Scale Range Audio/Video
Real-time Synchronization of Multi-User Status
Room Capabilities
Room Connection Status Description
Real-time Messaging and Signaling
Login to Multiple Rooms
Audio Capabilities
Audio Spectrum and Sound Level
Audio 3A Processing
Voice Changer/Reverb/Stereo
Audio Mixing
Scenario-Based AI Noise Reduction
Custom Audio Capture and Rendering
Custom Audio Processing
Getting Original Audio Data
Live Streaming Capabilities
Stream Mixing
Using CDN for Live Streaming
CDN Stream Publishing Authentication
Playing Stream by URL
Ultra-low Latency Live Streaming
Direct to CDN
Other Capabilities
Media Player
Audio Effect Player
Audio/Video Recording
Reference Documentation
Client API
Server API
Common Error Codes
FAQ