Voice Bot Implementation in Mobile Apps
Imagine: you launch a voice assistant in a navigation app. The user says 'Route to Minsk, avoiding toll roads,' but the response comes after 4 seconds, and half the words are misrecognized—'Minsk' turns into 'mink.' This isn't a hypothesis; it's a typical scenario when integrating a voice bot without considering latency, ambient noise, and the quirks of Russian speech. We've encountered such cases in projects for logistics companies and retail: losing 30% of users on first interaction due to slow response.
To avoid this, we build a pipeline: Speech-to-Text → NLU/LLM → Text-to-Speech with total response time under 1.5 seconds. The key technique is streaming processing at each stage: start transcription before the phrase ends, generate response from the first tokens, and synthesize speech concurrently with output. We also tune configurations per device (iOS/Android) and acoustic environment (quiet office vs. noisy street).
Below is a breakdown of the main components and their impact on dialogue quality: STT engine, TTS synthesizer, audio session management, and wake word. We optimized each block on real projects handling up to 10,000 requests per day.
Why Latency Is the Main Enemy of a Voice Bot
A typical latency breakdown shows where time is lost:
| Stage | Cloud variant | Optimized (streaming) |
|---|---|---|
| STT (transcription) | 400–800ms | 200–400ms |
| NLP / LLM response | 500–2000ms | 150–400ms (streaming + cache) |
| TTS (synthesis) | 300–600ms | 100–200ms |
| Network (2x) | 100–300ms | — |
| Total | 1.3–3.7s | ~1s |
The optimized variant is 2–3 times faster. The key is not waiting for each stage to finish:
- STT with
shouldReportPartialResults = true—start processing before the phrase ends. - LLM streaming—start synthesis as soon as the first tokens arrive.
- TTS streaming—start playback while the rest of the phrase is still being synthesized.
Choosing an STT Engine for Russian
| Engine | Russian Quality | Latency | Offline | Cost |
|---|---|---|---|---|
| Native (SFSpeechRecognizer / SpeechRecognizer) | Fair (short commands) | 200–300ms | Yes | Free |
| Yandex SpeechKit | Excellent | 200–400ms (streaming) | No | Per minute |
| Whisper API (OpenAI) | Very high | 200–500ms | No | Per request |
| Google Cloud Speech-to-Text | High | 200–400ms (gRPC streaming) | No | Per minute |
For conversational dialogues with specialized vocabulary (medicine, law), we recommend Yandex SpeechKit—it's trained on a large Russian corpus and gives 15–20% fewer errors than native STT.
// iOS: AVAudioEngine → Yandex SpeechKit streaming
class VoiceBotRecorder {
private let audioEngine = AVAudioEngine()
private var recognitionStream: RecognitionStream?
func startRecording() throws {
let inputNode = audioEngine.inputNode
let format = AVAudioFormat(commonFormat: .pcmFormatInt16,
sampleRate: 16000,
channels: 1,
interleaved: false)!
recognitionStream = speechKitClient.createStream(config: streamConfig)
inputNode.installTap(onBus: 0, bufferSize: 4096, format: format) { [weak self] buffer, _ in
guard let pcmData = buffer.int16ChannelData?[0] else { return }
let bytes = Data(bytes: pcmData, count: Int(buffer.frameLength) * 2)
try? self?.recognitionStream?.send(audio: bytes)
}
audioEngine.prepare()
try audioEngine.start()
}
}
TTS: Speech Synthesis That Doesn’t Annoy
- ElevenLabs—best quality, voice cloning for branding, streaming via WebSocket.
- OpenAI TTS—models
tts-1(fast) andtts-1-hd(high quality); for Russian, thenovavoice sounds most natural. - Yandex SpeechKit TTS—voices
alena,filipp,jane; streaming via gRPC, naturalness on par with ElevenLabs for Russian-speaking audience. - Native synthesis—
AVSpeechSynthesizer(iOS) andTextToSpeech(Android)—free, offline, but robotic.
Choosing a TTS depends on budget and naturalness requirements: for simple responses, native is enough; for a 'consultant assistant' service, only cloud solutions will do. Contact us for help selecting the optimal TTS for your scenario.
Audio Management: Avoiding Echo and Interruptions
iOS. The AVAudioSession category must be .playAndRecord with the .defaultToSpeaker option. When playing TTS, temporarily deactivate the microphone: AVAudioSession.sharedInstance().setActive(false) before synthesis, true after.
try AVAudioSession.sharedInstance().setCategory(
.playAndRecord,
options: [.defaultToSpeaker, .allowBluetooth]
)
Android. Use AudioManager.requestAudioFocus() during playback, abandonAudioFocus() after. Bluetooth headsets require separate handling via BluetoothHeadset.
Barge-in (interruption). If the user starts speaking while the bot is still responding, detect speech → stop TTS → start recording. Voice Activity Detection can be implemented via AudioRecord.getMaxAmplitude() or using WebRTC VAD for greater accuracy.
Wake Word and Hands-Free Mode
For scenarios like 'driving navigation' or 'smart glasses,' we add a wake word, e.g., 'Hey Assistant,' that activates the bot without a button press. We use Picovoice Porcupine—supports custom wake words and runs entirely on-device. An alternative is OpenWakeWord (open source). Resource consumption: ~50 MB RAM, active only during listening.
What’s Included in the Work
- Audit of current architecture—assessing requirements for language, latency, speaker diarization.
- Engine selection—STT/TTS per task (budget, accuracy, offline mode).
- Audio pipeline development—microphone capture, PCM/WAV encoding, streaming to cloud APIs.
- NLU/LLM integration—from simple intent routing to response generation via OpenAI or Yandex GPT.
- Latency optimization—streaming, caching, granular buffer tuning.
- UI/UX—state visualization (listening, thinking, speaking), sound wave animation, error handling.
- Testing—on real devices with different accents, background noises (cafe, street, subway).
- Documentation and training—integration description, API keys, support procedures.
- Post-launch support—monitoring, model fine-tuning for growing phrase base.
Our Experience and Guarantees
We’ve been working with voice interfaces for over 5 years—implementing more than 30 projects for iOS and Android in Russian and international companies. Our engineers are certified Swift and Kotlin developers with experience integrating Yandex SpeechKit, ElevenLabs, and OpenAI. We guarantee stable operation under loads up to 10,000 requests per day and an SLA on response time of no more than 2 seconds.
Timeline Estimates
- Basic version with native STT/TTS—1 week.
- With cloud engines (Yandex SpeechKit / ElevenLabs), streaming, and ~1s latency—3–5 weeks.
- With wake word and hands-free—+1–2 weeks.
Work Stages
- Analysis and prototype—2–4 days.
- Audio pipeline design—2–3 days.
- Implementation—1–3 weeks.
- Testing and debugging—3–5 days.
- Deploy to App Store / Google Play—1–2 days.
Contact us to evaluate your project—we’ll find the optimal configuration for your budget and requirements. Request a demo of a working voice bot on real data.







