tuner-livekit-sdk
Automatically ingest LiveKit Agents session data into the Tuner observability API.
Drop it into your entrypoint and every session — transcript, tool calls, timing metrics, usage, cost — is submitted to Tuner when the call ends. No manual event wiring required.
Features
- Zero-friction wiring — one line after
AgentSessioncreation; the plugin hooks itself into session lifecycle events. - Rich per-turn timing — STT/LLM/TTS latency, end-of-turn delay, and end-to-end latency captured per transcript segment.
- Tool call tracking — function calls and their results are merged into the timeline with real timing data.
- LangGraph / LangChain support — via the bundled
tuner-langchainpackage, capture node transitions and tool calls straight from your graph. - SIP simulation correlation — match Tuner-initiated simulation calls to your production SIP trunk.
- Cost calculation — plug in your own pricing function; Tuner reports cost per call.
- Resilient delivery — configurable timeouts and retries on transient failures.
Table of Contents
- Installation
- Quickstart
- Configuration
- Options
- Data captured
- LangGraph / LangChain observability
- Simulation correlation (SIP)
- Privacy & data handling
- Requirements
- Support
- Development
- License
Installation of the Library into your Livekit project
pip install tuner-livekit-sdk
Quickstart
Set credentials via environment variables:
export TUNER_API_KEY="tr_api_..."
export TUNER_WORKSPACE_ID="123"
export TUNER_AGENT_ID="my-agent"
Then drop the plugin in right after creating your AgentSession:
from tuner import TunerPlugin
async def entrypoint(ctx: JobContext):
session = AgentSession(...)
TunerPlugin(session, ctx) # wires itself automatically
await session.start(...)
That's it. The plugin listens to session events and submits call data to Tuner when the session ends.
Configuration
Environment variables
| Variable | Required | Description |
|---|---|---|
TUNER_API_KEY |
✅ | Bearer token (starts with tr_api_) |
TUNER_WORKSPACE_ID |
✅ | Integer workspace ID |
TUNER_AGENT_ID |
✅ | Agent identifier from Tuner Agent Settings |
TUNER_BASE_URL |
— | API base URL (default: https://api.usetuner.ai) |
Credentials from code
Pass credentials directly instead of (or to override) environment variables:
TunerPlugin(
session, ctx,
api_key="tr_api_...",
workspace_id=123,
agent_id="my-agent",
)
Options
Call type
By default the plugin auto-detects the call type (phone_call for SIP participants, web_call otherwise). Override it explicitly:
TunerPlugin(session, ctx, call_type="phone_call")
TunerPlugin(session, ctx, call_type="web_call")
Recipient (callee)
Pass the phone number or SIP URL of the called party when your agent initiates or routes outbound calls. This field is not auto-collected — supply it explicitly when known:
# E.164 phone number
TunerPlugin(session, ctx, recipient="+15551234567")
# SIP URI
TunerPlugin(session, ctx, recipient="sip:alice@example.com")
recipient is optional. If omitted it is simply not included in the call record.
Recording URL
Tuner requires a recording_url for every call. If you don't provide a resolver the plugin logs a warning and submits "pending" as a placeholder:
# Static URL
async def my_resolver(room_name: str, job_id: str) -> str:
return f"https://cdn.example.com/recordings/{job_id}.ogg"
TunerPlugin(session, ctx, recording_url_resolver=my_resolver)
# LiveKit Egress → S3
async def egress_resolver(room_name: str, job_id: str) -> str:
url = await my_egress_db.get_recording_url(room_name)
return url or "pending"
TunerPlugin(session, ctx, recording_url_resolver=egress_resolver)
Cost calculation
Provide a callable that receives a UsageSummary and returns the call cost in cents:
def calculate_cost(usage) -> float:
llm_cost = usage.llm_prompt_tokens * 0.000_003
llm_cost += usage.llm_completion_tokens * 0.000_015
tts_cost = usage.tts_characters_count * 0.000_030
stt_cost = usage.stt_audio_duration * 0.000_006
total_dollars = llm_cost + tts_cost + stt_cost
return round(total_dollars * 100, 2) # dollars -> cents
TunerPlugin(session, ctx, cost_calculator=calculate_cost)
Extra metadata
Attach arbitrary key-value data to every call record:
TunerPlugin(
session, ctx,
extra_metadata={
"env": "production",
"region": "us-east-1",
"deployment": "v2.3.1",
},
)
Retry and timeout
TunerPlugin(
session, ctx,
timeout_seconds=15.0, # per-request timeout (default: 30.0)
max_retries=5, # retries on 5xx / 429 / network errors (default: 3)
)
Agent version tracking
Track which version of your agent handled each call — useful when you update a prompt, swap a model, or change your pipeline:
AGENT_VERSION=42 python agent.py start
Tuner reads it automatically. Bump the number on every deployment.
Override in code (takes priority over the env var):
TunerPlugin(session, ctx, agent_version=42, ...)
Disable the plugin
Useful for local development or test environments:
import os
TunerPlugin(
session, ctx,
enabled=os.getenv("ENV") == "production",
)
Data captured
Each item in session.history is mapped to a transcript segment (role, text, start_ms, end_ms, metadata). For user / agent segments, metadata includes the following per-turn timing and quality fields:
| Field | Type | Description |
|---|---|---|
id |
string |
LiveKit ChatContext item ID |
interrupted |
bool |
Whether this turn was interrupted before completing |
transcript_confidence |
float | null |
STT confidence score for user turns |
stt_node_ttfb |
int (ms) | null |
Time to obtain the transcript after the user stopped speaking |
eou_delay |
int (ms) | null |
Time between end of user speech and the decision to end their turn |
llm_node_ttft |
int (ms) | null |
Time for the LLM node to return its first token |
tts_node_ttfb |
int (ms) | null |
Time for the TTS node to return its first audio chunk |
e2e_latency |
int (ms) | null |
Time from end of user speech to the agent beginning its response |
All timing fields above are taken directly from LiveKit's own per-turn metrics (EOUMetrics, STT/LLM/TTS metrics) and passed through unchanged — the SDK does not recompute them.
Tool calls (agent_function / agent_result roles) and, when LangGraph/LangChain instrumentation is enabled, node_transition segments are interleaved into the same timeline, sorted by start_ms.
LangGraph / LangChain observability
tuner-langchain ships as a dependency of tuner-livekit-sdk, so no separate
install step is needed. If your agent uses LangGraph or LangChain as the
orchestration layer, wire it in with wrap_graph() or wrap_chain(). Each
returns a drop-in replacement for the graph/chain you pass in — hand it
straight to LLMAdapter, no callbacks to wire up:
from tuner import TunerPlugin
from livekit.plugins import langchain
plugin = TunerPlugin(session, ctx)
llm = langchain.LLMAdapter(
plugin.wrap_graph(my_graph),
stream_mode="messages",
)
Use wrap_chain() instead of wrap_graph() for a plain (non-graph) LangChain
runnable.
To limit what data is forwarded to Tuner, pass a CaptureConfig:
from tuner import TunerPlugin
from tuner_langchain import CaptureConfig
plugin = TunerPlugin(session, ctx)
wrapped_graph = plugin.wrap_graph(
my_graph,
capture=CaptureConfig(
tool_inputs=False,
node_instructions=False,
),
)
Simulation correlation (SIP)
Tuner simulations dial into your agent through the same SIP trunk that handles production phone calls. To match a simulation run with the session your agent submits, the SDK forwards LiveKit's sip.callIDFull attribute as a sip_call_id.
This section covers the SDK wiring only. For LiveKit platform setup (SIP URI, inbound trunk, dispatch rule, Tuner SIP settings), see:
→ docs.usetuner.ai/docs/api-and-integrations/connecting-to-livekit/simulation-setup
Requirements
tuner-livekit-sdk >= 0.1.5(thesip_call_idargument was added in 0.1.5)
Step 1 — The _extract_sip_call_id helper
This helper scans the LiveKit room for the SIP caller and returns their sip.callIDFull — the value Tuner uses to match a simulation run to the session your agent submits.
from livekit import rtc
def _extract_sip_call_id(ctx: JobContext) -> str | None:
for participant in ctx.room.remote_participants.values():
if participant.kind != rtc.ParticipantKind.PARTICIPANT_KIND_SIP:
continue
attributes = dict(getattr(participant, "attributes", {}) or {})
sip_call_id_full = attributes.get("sip.callIDFull")
if isinstance(sip_call_id_full, str) and sip_call_id_full:
return sip_call_id_full
return None
How it works:
- Loops through remote participants and keeps only the SIP one (rooms can hold web clients, observers, etc.).
- Reads
sip.callIDFullfrom that participant's attributes — this is the full SIPCall-IDTuner stamps on its outbound leg (not the shortersip.callID). - Returns
Nonefor web calls or non-simulation SIP calls;TunerPluginacceptsNoneand simply skips correlation.
Step 2 — Pass it to TunerPlugin
Once you have the helper, the wiring in entrypoint is three lines: connect, extract, attach.
async def entrypoint(ctx: JobContext):
session = AgentSession(...)
await ctx.connect()
sip_call_id = _extract_sip_call_id(ctx)
TunerPlugin(
session,
ctx,
sip_call_id=sip_call_id,
# ...other options
)
await session.start(...)
⚠️ Order matters:
ctx.room.remote_participantsis empty untilawait ctx.connect()completes. If you call the helper too early it will always returnNoneand you'll silently lose correlation for every simulation — no error, just missing data in Tuner. Always: buildAgentSession→await ctx.connect()→ extract ID → attach plugin →await session.start(...).
Step 3 — Full example
Putting the helper, the plugin wiring, and the usual options (cost, recording URL, metadata) together:
import os
from livekit import rtc
from livekit.agents import JobContext, AgentSession
from tuner import TunerPlugin
def _extract_sip_call_id(ctx: JobContext) -> str | None:
for participant in ctx.room.remote_participants.values():
if participant.kind != rtc.ParticipantKind.PARTICIPANT_KIND_SIP:
continue
attributes = dict(getattr(participant, "attributes", {}) or {})
sip_call_id_full = attributes.get("sip.callIDFull")
if isinstance(sip_call_id_full, str) and sip_call_id_full:
return sip_call_id_full
return None
def calculate_cost(usage) -> float:
total_dollars = (
usage.llm_prompt_tokens * 0.000_003
+ usage.llm_completion_tokens * 0.000_015
+ usage.tts_characters_count * 0.000_030
)
return round(total_dollars * 100, 2) # cents — the Tuner API expects cents
async def get_recording_url(room_name: str, job_id: str) -> str:
return await my_storage.get_url(job_id) or "pending"
async def entrypoint(ctx: JobContext):
session = AgentSession(...)
await ctx.connect()
sip_call_id = _extract_sip_call_id(ctx)
TunerPlugin(
session,
ctx,
api_key=os.environ["TUNER_API_KEY"],
workspace_id=int(os.environ["TUNER_WORKSPACE_ID"]),
agent_id="customer-support-v3",
call_type="phone_call",
recording_url_resolver=get_recording_url,
cost_calculator=calculate_cost,
sip_call_id=sip_call_id,
extra_metadata={"env": "prod", "region": "us-east-1"},
timeout_seconds=20.0,
max_retries=3,
enabled=True,
)
await session.start(...)
Privacy & data handling
The plugin submits the following to the Tuner API for each call: the plain-text transcript, per-turn timing metadata (see Data captured), tool call names/arguments/results, token/character/audio usage counts, and any extra_metadata you configure. It does not access raw audio, video, or room recordings directly — recordings are only referenced by the URL your recording_url_resolver returns.
Use CaptureConfig (see LangGraph / LangChain observability) to exclude tool inputs or node instructions from LangGraph/LangChain traces, and enabled=False to disable submission entirely for local development or test environments.
Requirements
- Python ≥ 3.10
livekit-agents >= 1.4tuner-livekit-sdk >= 0.1.5(needed forsip_call_id/ SIP correlation)aiohttp >= 3.9tuner-langchain >= 0.1.1(installed automatically as a dependency; used bywrap_graph()/wrap_chain())
Support
- Docs: docs.usetuner.ai
- Issues / feature requests: GitHub Issues
- Email: support@usetuner.ai
- Release notes: CHANGELOG.md
Development
Set up a local environment:
uv sync --dev
source .venv/bin/activate
Run the test suite:
uv run pytest -v
Publishing to PyPI
pip install build twine
python -m build
twine upload dist/*
License
MIT
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