How low is the voice latency for Sarvam Indic voice AI callers?

By integrating WebRTC audio streaming, WebSocket speech-to-text (STT), streaming local LLM inference, and instant TTS, we achieve sub-300ms end-to-end turn-taking latency.

Which Indian languages and code-mixed dialects are supported?

We support Hindi, Hinglish, Punjabi, Tamil, Telugu, Kannada, Bengali, Marathi, Gujarati, and Malayalam with full code-switching capability.

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Low-Latency Conversational Voice AI

Sub-300ms Multilingual Voice AI
& Sarvam Indic Telephony Agents

Deploy ultra-fast, natural conversational voice agents for inbound support, outbound lead qualification, and debt recovery. Integrated directly with SIP trunks, PSTN phone networks, and Sarvam Indic models for seamless Hinglish and Punjabi dialogues.

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Sub-300ms Turn-Taking

Streaming WebSockets and VAD (Voice Activity Detection) allow interruption handling and human-like natural conversation loops.

Code-Mixed Indic Dialects

Flawlessly parses Hinglish, Punjabi, and regional dialect code-switching without losing conversational context.

PSTN & SIP Trunking

Connects to Exotel, Twilio, Tata Tele, or custom Asterisk/FreeSWITCH PBX setups for direct mobile phone call routing.

Technical Voice Stack Architecture

Audio Transport: Opus/PCM WebRTC frames piped over bi-directional WebSockets.

Speech-to-Text (STT): Sarvam Saaras / Whisper Live streaming STT.

LLM Reasoning: Fine-tuned Llama 3.3 70B / Qwen 2.5 streaming token responses.

Text-to-Speech (TTS): Sarvam Bulbul neural TTS producing expressive, natural Indian voices.

Frequently Asked Questions

Can the voice AI handle background noise or call interruptions?

Yes. Built-in Silero VAD (Voice Activity Detection) immediately pauses AI speech output the instant the human user speaks, maintaining true human-like conversation flow.

Is voice call data compliant under TRAI DLT guidelines?

Yes. Our SIP telephony integration complies with TRAI regulations, DLT registered sender IDs, and opt-out preferences.