Stream bi-directional RTP audio over secure WebSockets for real-time speech access. Ideal for AI assistants and real-time coaching platforms that analyze speech on the fly.
Take full control of every leg of a call with one-line API commands. Route callers, move them between endpoints, or cleanly end sessions in real time.
Protect caller privacy by replacing real numbers with temporary, anonymized proxies—all handled transparently by the Telnyx platform.
Play dynamic, natural-sounding voices in 40 + languages and accents. Select normal, neural, or HD output from Telnyx, Polly, Azure, or ElevenLabs to power lifelike prompts and responses.
Convert voice to text in real-time with low-latency, high accuracy engines from Telnyx or Google and automatic speaker separation for clear multi-party transcripts.
Add voice bots to any call in minutes by chaining TTS, STT, and LLM logic via simple AIGather and AIAssistant commands, with no extra middleware or SIP scripting needed.
Start or stop single- or dual-channel recordings at any moment for compliance, training, or playback, with secure storage.
Add carrier-grade voice to any web or mobile app. Telnyx WebRTC handles signaling, NAT traversal, and encrypted media so users can place and receive calls right in the browser or app.
Bridge multiple participants worldwide into a single, high-quality audio stream, complete with mute, record, and participant management via API.
Identify voicemail, fax tones, or live humans in milliseconds. Premium mode can identify silence and separate business business greeting from personal greeting.
Inject MP3 or WAV prompts (think announcements, ads, or hold music) directly into a live call.
Gather keypad input to drive IVRs, menu navigation, or PIN entry, with flexible timeouts and validation rules.
Pass an active call to another SIP endpoint without dropping audio, enabling warm transfers and carrier-grade failover.
Remove background hiss, hum, and chatter so voices stay front-and-center, even from busy call-center floors or noisy cafés.
Use 16 kHz wideband audio on eligible networks for richer, more natural-sounding conversations.
Track and optimize every call in real time. Get real-time call status and sentiment scores to your stack so you can monitor, troubleshoot, and optimize every conversation in real time.
Receive instant JSON callbacks for call status, media fork handshakes, and transcription results so your app stays perfectly in sync.
Using a legacy system but want to take advantage of AI? No problem, we also support SIPREC. Client and server support for standards-based call recording that streams dual-channel RTP.
Get a clear breakdown of how programmable voice works, what it costs, and the features to look for when building voice into your app.
Explore how real-time transcription supports natural interactions, improves responsiveness, and helps power multilingual AI conversations.
