voice-agents
Skill72 starsChanged 46 days agoApache-2.0View on GitHub
Build real-time Indic voice agents with Sarvam STT/TTS/LLM on LiveKit or Pipecat (Python), plus JS/TS custom-pipeline notes. Use this skill when creating phone bots, IVR, or always-on conversational voice apps. For one-shot in-chat voice reply or dubbing via MCP, use sarvam-mcp instead.
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3 other projects use a version of this.
Adopted by
Most recently taken up by
- ahmedibm9-cyber/VibeGuardtheir own version, first seen Oct 20, 2026
- xfstudio/skillstheir own version, first seen Oct 13, 2026
- ai4brands-design/claude-skillstheir own version, first seen Mar 22, 2026
Other versions3 projects use a version of this.
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Wrote their own version · 3
- xfstudio/skills★ 9same name
- ai4brands-design/claude-skills★ 2same name
- ahmedibm9-cyber/VibeGuard★ 0same name
How matches are found. Word for word: the exact same text. Edited: shares passages with this file (“99% same” means 99% of its five-word phrases also appear here; 10% or more counts). Their own version: similar meaning, or, weakest, the same name. We can’t tell which project had it first. Projects are counted among the 200 most-starred candidates, and mentions are links to the repository we found, up to 50 per site, so both counts are a floor.
In this repository7 other files it gives its agents.
- chatSkill
- doc-aiSkill
- dubbingSkill
- sarvam-mcpSkill
- speech-to-textSkill
- text-to-speechSkill
- vibe-codingSkill
The file
We split this skill into the rules it gives an agent
1 rule. About 0 are common, 0 are distinctive, and 1 is very specific to this project.
≈0
Common · estimate
Many files like this have a rule of this kind.
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Distinctive · estimate
This author's own choices. Worth a look.
1
Very specific
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- Common
- Distinctive
- Very specific
- Very specific: line 65
Run with python agent.py dev, test with python agent.py console.
≈ “Common” is an estimate: at least 1 in 5 of the 350,495 skill files we read have a rule of the same kind, not the same words. “Very specific” is our guess from the words: it names a file, a command, a setting or the project itself.
Files
Voice Agents — Sarvam AI
One-shot in-chat voice/dub → sarvam-mcp (
sarvam_tools_voice/_dub). This skill = LiveKit/Pipecat agent code.
[!IMPORTANT] Auth:
api-subscription-keyheader — NOTAuthorization: Bearer. Env var:SARVAM_API_KEY
LiveKit Quick Start
pip install "livekit-agents[sarvam,silero]" python-dotenvfrom livekit.agents import JobContext, WorkerOptions, cli
from livekit.agents.voice import Agent, AgentSession
from livekit.plugins import sarvam
class VoiceAgent(Agent):
def __init__(self) -> None:
super().__init__(
instructions="You are a helpful voice assistant. Be friendly, concise, and conversational.",
stt=sarvam.STT(
language="unknown", # auto-detect; or "hi-IN", "ta-IN", etc.
model="saaras:v3",
mode="transcribe",
flush_signal=True # emits speech start/end events for turn-taking
),
llm=sarvam.LLM(model="sarvam-105b"),
tts=sarvam.TTS(
target_language_code="en-IN",
model="bulbul:v4-flash",
speaker="simran_en_customer",
),
)
async def on_enter(self):
self.session.generate_reply()
async def entrypoint(ctx: JobContext):
# Do NOT pass vad= — VAD is handled internally by the Sarvam plugin
session = AgentSession(
turn_detection="stt", # let Sarvam STT handle turn detection
min_endpointing_delay=0.07 # ~70ms matches Sarvam STT processing latency
)
await session.start(agent=VoiceAgent(), room=ctx.room)
if __name__ == "__main__":
cli.run_app(WorkerOptions(entrypoint_fnc=entrypoint))Run with python agent.py dev, test with python agent.py console.
Pipecat Quick Start
pip install "pipecat-ai[daily,sarvam]" python-dotenv loguruimport os
from pipecat.pipeline.pipeline import Pipeline
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
from pipecat.services.sarvam.stt import SarvamSTTService
from pipecat.services.sarvam.tts import SarvamTTSService
from pipecat.services.sarvam.llm import SarvamLLMService
stt = SarvamSTTService(
api_key=os.getenv("SARVAM_API_KEY"),
language="unknown", # auto-detect; or "hi-IN", "ta-IN", etc.
model="saaras:v3",
mode="transcribe" # or "translate" for speech-to-English
)
tts = SarvamTTSService(
api_key=os.getenv("SARVAM_API_KEY"),
settings=SarvamTTSService.Settings(
model="bulbul:v4-flash",
voice="simran_en_customer", # persona ID — not a v3 short name
language="en-IN",
pace=1.0,
),
)
llm = SarvamLLMService(
api_key=os.getenv("SARVAM_API_KEY"),
settings=SarvamLLMService.Settings(model="sarvam-105b"),
)
messages = [{"role": "system", "content": "You are a friendly AI assistant. Keep responses brief."}]
context_aggregator = LLMContextAggregatorPair(LLMContext(messages))
pipeline = Pipeline([
transport.input(), # Daily or WebRTC transport
stt,
context_aggregator.user(),
llm,
tts,
transport.output(),
context_aggregator.assistant(),
])JavaScript/TypeScript Note
LiveKit and Pipecat agents are Python-only. For JS/TS voice pipelines, use the individual SDK methods directly:
import { SarvamAIClient } from "sarvamai";
const client = new SarvamAIClient({ apiSubscriptionKey: "YOUR_SARVAM_API_KEY" });
// STT: client.speechToText.transcribe({...})
// TTS: client.textToSpeech.convertStream({...}) // returns BinaryResponse
// LLM: client.chat.completions({...})Gotchas
| Gotcha | Detail |
LiveKit: no vad= | Do NOT pass vad= to AgentSession — VAD is handled internally by the Sarvam plugin. Set turn_detection="stt" and min_endpointing_delay=0.07 instead. |
LiveKit: flush_signal=True | Required on sarvam.STT for speech start/end events and proper turn-taking. |
| TTS voice params differ by plugin | LiveKit (livekit-plugins-sarvam through 1.5.2): target_language_code + speaker. language_code= on sarvam.TTS() is rejected. Pipecat (1.0): settings=SarvamTTSService.Settings(voice=..., language=..., model=...) — voice, not speaker, and not bare constructor kwargs. |
| v4 speakers are persona IDs | model="bulbul:v4-flash" needs voice_language_style (simran_en_customer, aparna_hi_customer). Short names (shubh, anand, priya) 400. LiveKit defaults a missing speaker to anushka when the model is not v3 — always pass speaker. Hindi: aparna_hi_customer. Tamil: gokul_ta_narration. No v4 personas for ml-IN / od-IN — use bulbul:v3 there. |
| Pipecat class names | SarvamSTTService/SarvamTTSService/SarvamLLMService from pipecat.services.sarvam.stt/.tts/.llm — NOT SarvamSTT. LLM model goes in SarvamLLMService.Settings(model=...), system prompt in LLMContext messages. |
sarvam-30b is deprecated — and now hard-errors at the API | Verified live: calling sarvam-30b via the chat API now returns 400: "Model 'sarvam-30b' has been deprecated. Please use one of the available models instead: sarvam-105b, sarvam-105b-conversations." A Pipecat pipeline built on sarvam-30b will still construct fine (see below) but every LLM turn will start failing at runtime — this isn't a future risk, it's already broken if you're on it. |
sarvam-105b-conversations may not be in your installed Pipecat's allowlist yet | Sarvam ships sarvam-105b-conversations — a 32K-context variant post-trained specifically for real-time dialogue/voice agents, i.e. the slot sarvam-30b used to fill. Verified live against pipecat-ai==0.0.108: SarvamLLMService._SUPPORTED_MODELS is frozenset({"sarvam-105b", "sarvam-105b-32k", "sarvam-30b", "sarvam-30b-16k"}) — it does not include sarvam-105b-conversations yet, and constructing SarvamLLMService(settings=SarvamLLMService.Settings(model="sarvam-105b-conversations")) raises ValueError: Unsupported Sarvam LLM model 'sarvam-105b-conversations'. Allowed values: sarvam-105b, sarvam-105b-32k, sarvam-30b, sarvam-30b-16k. immediately. Also note the same allowlist still happily accepts sarvam-30b/sarvam-30b-16k at construction time even though those are now dead at the API level (see row above) — construction succeeding is not a signal the model works. Check your installed pipecat-ai version's allowlist before switching to sarvam-105b-conversations; use plain sarvam-105b if it's not there yet. |
max_tokens budget | Sarvam models reason internally. Don't set low max_tokens or content will be None. Omit, set 500+, or disable with reasoning_effort=None. |
| TTS pitch/loudness | On bulbul:v4-flash, pitch is −0.5–0.5 and loudness is 0.1–2.5. Bulbul v3 now accepts them too (live-verified 2026-10-06); pace is 0.5–2.0. Out-of-range values return 400. |
| STT WebSocket codecs | Only wav/pcm — no MP3/AAC/OGG for streaming. |
| HTTP Stream for TTS | convert_stream returns binary audio directly (no base64), better for pipelines. |
| Telephony | For phone agents (e.g. Exotel), set audio_in_sample_rate=8000 and audio_out_sample_rate=8000 to match telephony audio. |
Full Docs
Fetch framework integration guides, environment setup, and advanced patterns from:
- https://docs.sarvam.ai/llms.txt — comprehensive docs index
- LiveKit Guide
- Pipecat Guide
- Bulbul v4 Flash voices
- Rate Limits
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