Speech-to-text sounds simple until you actually build it.

You need to handle RTP packet assembly, choose the right audio codec (G.711? G.722? Opus?), manage jitter buffers, stream audio chunks to a transcription API with low enough latency that the conversation doesn't feel broken, and then pipe that text into your AI agent — all in real time, while keeping the call alive.

Most developers who try this spend weeks on audio infrastructure before writing a single line of AI logic.

There's a better path.

The Real Problem: Audio Is Hostile Territory for Most Developers

Voice calls operate at the network layer — RTP streams, SIP signaling, DTMF tones. These are protocols that telecom engineers have spent decades specializing in.…