TTS latency explained: time to first audio vs. time to first byte

TTS latency in voice agents should be evaluated primarily through time to first audio (TTFA), the interval from sending text to a TTS service until the caller hears an audible sound, rather than time to first byte (TTFB), which only measures when initial audio data arrives and may include headers, empty frames, or silence. TTFA incorporates server processing, network transit, decoding, resampling, buffering, and device or telephony output, making it a more accurate measure of perceived responsiveness within the broader voice-to-voice latency budget that also includes turn detection, speech-to-text, and LLM generation. Streaming TTS can reduce delay by transmitting audio before an entire response is synthesized, while bidirectional streaming allows text to be sent as an LLM generates it; however, chunk size involves a trade-off between fast playback and natural prosody. Recommended practices include prewarming persistent connections, aligning TTS audio formats and regions with telephony infrastructure, reducing playback buffers carefully, prompting for short opening sentences, and measuring TTFA under production-like conditions using realistic text, concurrency, and percentile metrics such as P50 and P95. The discussion also emphasizes that accurate transcription and effective turn detection may improve conversational speed more than TTS optimization alone, since misheard speech and poorly timed interruptions can create costly additional turns.

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