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Telnyx Conversation Relay streams transcribed conversation text over WebSockets, so you can integrate voice calls with text-based AI workflows without handling raw audio.
Conversation Relay sits between Telnyx telephony and your AI application. Telnyx manages the voice transport, real-time STT, and TTS, while your application handles the language model and business logic. Your existing AI engine, knowledge base, and tools stay where they are, and the audio layer underneath stops being something your team has to maintain. The result is a faster path to a production voice agent without giving up control over the parts that make your product yours.
Telnyx already supports real-time media streaming over WebSockets for bidirectional raw audio. Conversation Relay builds on this by streaming text instead of media, so your application receives transcribed conversation in real time without needing to process audio payloads yourself.
This suits teams focused on text-based AI workflows where raw audio handling is unnecessary, such as CRM logging, chatbot handoff, and any pipeline that works with transcribed text rather than media frames.
Processing raw audio requires dedicated infrastructure and adds complexity. Conversation Relay removes that overhead by delivering transcribed text directly, so you can plug voice calls into existing text-based workflows without building a media pipeline, and use the Telnyx STT and TTS with best-of-breed vendors, low latency, and native integration.
Read our developer documents for Conversation Relay Voice API and Conversation Relay TeXML to learn more.
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