Give your mobile and web apps a voice by using these building blocks.
WebRTC audio is made seamless with the use of signaling, media routing, and STUN/TURN services. Create a call session to PSTN carriers, SIP endpoints, networks, mobile clients, and browsers using only one line of code.
Automatic Gain Control (AGC) and Echo Cancellation lets you add voice calling to native apps, with voice web apps features that can leverage conferencing to the intelligent wait queue.
Ensure the fastest voice transmission between callers while maintaining low audio latency across 7 data centers globally with our dynamic call-routing optimization, where media transport is automatically re-routed as the call dynamically changes and connects.
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Add music or play pre-recorded voice prompts
Prompt for voicemail or have the ability to record & store calls
Communicate recorded calls into written words
Voice out text to callers in 20 + languages and accents
Create & utilize smart hold experiences for callers
Track live usage & trigger custom webhooks
Connect and bridge any combination of client, SIP, & PSTN calls