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The right parameter configuration can make a significant difference in transcription quality and latency for realtime use cases. This guide covers recommended starting points for common scenarios and highlights pitfalls that frequently trip up new integrations.
These recommendations apply to the Realtime API and are passed during session initialization. They are starting points — tune them to match your specific needs.

Language Configuration

One of the most common configuration mistakes is misunderstanding how language_config works. Choosing the right setup avoids unnecessary detection overhead and improves accuracy. When to set an explicit language:
  • You know the language of the audio ahead of time.
  • The audio is monolingual (single language throughout).
  • You want the fastest, most accurate results.
When to use auto-detection:
  • You process audio in many different languages and don’t know which one beforehand.
  • You want Gladia to pick the language automatically.
When code_switching is false and no language is set, the language is detected on the first utterance and reused for the rest of the session or file. If the beginning of your audio contains silence, music, or a different language than the main content, this can lead to incorrect detection for the whole transcription.
Even when using auto-detection, pass a small list of likely languages in languages to constrain the search. This improves both accuracy and processing time.

Code Switching

Code switching (language_config.code_switching: true) lets Gladia detect and transcribe multiple languages within the same audio, re-evaluating the language on each utterance. When to enable it:
  • Speakers switch languages mid-conversation (e.g. bilingual meetings, multilingual customer support).
  • You need the detected language returned per utterance.
When NOT to enable it:
  • The audio is in a single language — code switching adds unnecessary processing and can introduce misdetections.
  • You’ve set exactly one language in languages — in that case code_switching is ignored anyway.
Do not enable code_switching with an empty languages list. When no languages are specified, the language detector evaluates every utterance against 100+ supported languages, which leads to frequent misdetections — especially between similar-sounding languages. Always provide a short list of languages you actually expect in the audio.

Custom Vocabulary

Custom vocabulary is a post-transcription replacement based on phoneme similarity. It’s essential for domain-specific terms that speech models frequently mis-transcribe. Best practices:
  • Always provide both the custom_vocabulary flag and a custom_vocabulary_config.
  • Add pronunciations to provide all the close spelling variants. You can use Automatic Phonemic Transcriber (IPA) in order to check if all the different spellings are covered.
  • Keep intensity moderate (0.4-0.6). High values increase false positives where unrelated words get replaced.
  • Set language on individual vocabulary entries when your audio is multilingual and a term is pronounced differently depending on the language.

Voice Agents

For callbots, customer-service assistants, or voice-driven chatbots the top priority is low latency. The agent must react quickly to user speech, even if sentence boundaries are not perfectly formed.
This setup is optimized for fast turn-taking. If utterances get cut off mid-sentence, raise endpointing slightly.

Meeting Recorders

For apps that record and transcribe meetings in real time — team stand-ups, board sessions, 1-on-1s — the goal is to produce a structured, speaker-attributed live transcript that can feed downstream features like summarization or live note-taking.
Diarization vs. multi-channel: if each speaker is on a separate audio channel (e.g. a, use the channel field on each utterance to identify who is speaking — diarization is not needed. See Multiple channelsIf all speakers share a single audio channel, enable diarization to separate the speakers. See Speaker diarization.

Call Centers

For live phone calls the priorities are speaker identification and fast, accurate transcription despite variable audio quality (telephony codecs, background noise, cross-talk).
Diarization vs. multi-channel: if each speaker is on a separate audio channel (e.g. a, use the channel field on each utterance to identify who is speaking — diarization is not needed. See Multiple channelsIf all speakers share a single audio channel, enable diarization to separate the speakers. See Speaker diarization.
For calls with more than two participants (e.g. conference bridges), use diarization_config.min_speakers / max_speakers instead of number_of_speakers to give the model a flexible range.

Subtitles & Captioning

When providing live subtitles, the goal is to sync text with the speaker in real time. The right balance between speed and segment quality depends on whether captions are displayed live or post-produced.
For post-production subtitles generated from a recording, consider using the Pre-recorded API with the dedicated subtitles feature instead — it produces SRT/VTT files with fine-grained timing controls.