Agent Transport

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Transport library (SIP/RTP & Audio Streaming) for voice AI agents to be used with frameworks like LiveKit Agents and Pipecat.

Agent Transport provides signaling and media primitives that AI agent frameworks need to make and receive voice calls. The core is written in Rust for efficient, low-jitter packet processing — audio pacing, RTP handling, and jitter buffering. Framework adapters for LiveKit Agents and Pipecat are provided as drop-in plugins. Bindings in Python and TypeScript/Node.js are also available for other use cases.

Transports

SIP/RTP — Register with any SIP provider, make and receive calls over RTP. G.711 codecs (PCMU/PCMA), DTMF (RFC 2833), NAT traversal (TCP signaling with Via alias, STUN for RTP), hold/unhold, call transfer. No server required, directly connect with telephony providers over SIP like Plivo.

Audio Streaming — Websocket based audio streaming that works with cloud telephony providers like Plivo that support bidirectional audio streaming.

Both transports produce and consume the same AudioFrame format (int16 PCM, 16kHz mono), so agent code works identically regardless of transport.

Framework Adapters

LiveKit Agents

Same AgentSession pipeline -- add ctx.session = session to wire SIP/audio stream transport:

SIP/RTP:

# LiveKit WebRTC                                # Agent Transport SIP/RTP
from livekit.agents import AgentServer,         from agent_transport.sip.livekit import
    JobProcess                                      AgentServer, JobProcess
server = AgentServer()                          server = AgentServer(sip_username=..., sip_password=...)

def prewarm(proc: JobProcess):                  def prewarm(proc: JobProcess):
    proc.userdata["vad"] = silero.VAD.load()        proc.userdata["vad"] = silero.VAD.load()
server.setup_fnc = prewarm                      server.setup_fnc = prewarm

@server.rtc_session()                           @server.sip_session()
async def entrypoint(ctx):                      async def entrypoint(ctx):
    session = AgentSession(                         session = AgentSession(
        vad=ctx.proc.userdata["vad"], ...)              vad=ctx.proc.userdata["vad"], ...)
    await session.start(                            ctx.session = session
        agent=Assistant(),                          await session.start(
        room=ctx.room)                                  agent=Assistant(), room=ctx.room)
cli.run_app(server)                             server.run()

Audio Streaming:

# LiveKit WebRTC                                # Agent Transport AudioStream
from livekit.agents import AgentServer,         from agent_transport.audio_stream.livekit import
    JobProcess                                      AudioStreamServer, JobProcess
server = AgentServer()                          server = AudioStreamServer(listen_addr="0.0.0.0:8765")

def prewarm(proc: JobProcess):                  def prewarm(proc: JobProcess):
    proc.userdata["vad"] = silero.VAD.load()        proc.userdata["vad"] = silero.VAD.load()
server.setup_fnc = prewarm                      server.setup_fnc = prewarm

@server.rtc_session()                           @server.audio_stream_session()
async def entrypoint(ctx):                      async def entrypoint(ctx):
    session = AgentSession(                         session = AgentSession(
        vad=ctx.proc.userdata["vad"], ...)              vad=ctx.proc.userdata["vad"], ...)
    await session.start(                            ctx.session = session
        agent=Assistant(),                          await session.start(
        room=ctx.room)                                  agent=Assistant(), room=ctx.room)
cli.run_app(server)                             server.run()

Full examples: sip_agent.py · sip_multi_agent.py · audio_stream_agent.py · audio_stream_multi_agent.py

See LiveKit SIP Transport docs for recording, Prometheus metrics, outbound API, and full reference.

Pipecat

Same Pipeline — swap transport, everything else stays identical. Audio pacing moves from Python to Rust:

# Pipecat + Plivo (Python audio pacing)          # Agent Transport (Rust audio pacing)
from pipecat.serializers.plivo import             from agent_transport.audio_stream.pipecat \
    PlivoFrameSerializer                              .serializers.plivo import PlivoFrameSerializer
from pipecat.transports.websocket.fastapi import  from agent_transport.audio_stream.pipecat \
    FastAPIWebsocketTransport                         .transports.websocket import WebsocketServerTransport

serializer = PlivoFrameSerializer(                serializer = PlivoFrameSerializer(
    stream_id=..., call_id=...,                       auth_id=..., auth_token=...)
    auth_id=..., auth_token=...)                  server = WebsocketServerTransport(
transport = FastAPIWebsocketTransport(                serializer=serializer)
    websocket=ws, params=Params(
        serializer=serializer))                   @server.handler()
                                                  async def run_bot(transport):
pipeline = Pipeline([                                 pipeline = Pipeline([
    transport.input(), stt, llm, tts,                     transport.input(), stt, llm, tts,
    transport.output()])                                   transport.output()])
task = PipelineTask(pipeline)                         task = PipelineTask(pipeline)

@transport.event_handler("on_client_connected")       @transport.event_handler("on_client_connected")
async def on_connected(transport, client):            async def on_connected(transport):
    await task.queue_frames([LLMRunFrame()])               await task.queue_frames([LLMRunFrame()])

await PipelineRunner().run(task)                      await PipelineRunner().run(task)

                                                  server.run()

Also available for SIP/RTP: from agent_transport.sip.pipecat import SipTransport

Full examples: audio_stream_agent.py · sip_agent.py

Installation

Python

For LiveKit Agents

pip install "agent-transport[livekit]"

For Pipecat

pip install "agent-transport[pipecat]"

Minimum versions: livekit-agents>=1.5, pipecat-ai>=0.0.108

Node.js

For LiveKit Agents

npm install agent-transport @livekit/agents @livekit/rtc-node

macOS note: The ONNX runtime bundled with @livekit/agents-plugin-livekit (v1.21+) has a known mutex crash on macOS that kills the turn detection inference process. Add "overrides": { "onnxruntime-node": "1.20.1" } to your package.json to pin a working version.

Building from source | Local development

Examples

Example Description
livekit/sip_agent.py SIP voice agent with tool calling, turn detection, preemptive generation
livekit/sip_agent.ts TypeScript SIP agent with tool calling, turn detection, metrics
livekit/sip_multi_agent.py Multi-agent with greeter -> sales/support handoff and tool calling
livekit/sip_multi_agent.ts TypeScript multi-agent with class inheritance and llm.handoff()
livekit/audio_stream_agent.py LiveKit agent over Plivo audio streaming
livekit/audio_stream_agent.ts TypeScript agent over Plivo audio streaming
livekit/audio_stream_multi_agent.py Audio streaming multi-agent with handoff and tool calling
livekit/audio_stream_multi_agent.ts TypeScript audio streaming multi-agent
pipecat/sip_agent.py Pipecat pipeline over SIP/RTP with VAD
pipecat/sip_multi_agent.py Pipecat multi-agent with greeter → sales/support handoff
pipecat/audio_stream_agent.py Pipecat over Plivo audio streaming with Rust recorder + mixer
pipecat/audio_stream_multi_agent.py Pipecat audio streaming multi-agent with handoff
cli/phone.py Interactive CLI softphone with mic/speaker, DTMF, mute, hold/unhold

See also: Feature Flags & CLI Phone docs

Releasing

Publishing is label-driven. Bump the version, add release-python-sdk or release-node-sdk label to your PR, and merge — CI handles the rest. Python and Node releases are independent.

License

MIT

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License expression

MIT
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Requires

Python >=3.10

Provides Extra

all livekit pipecat test

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