> ## Documentation Index
> Fetch the complete documentation index at: https://docs.stmlink.com/llms.txt
> Use this file to discover all available pages before exploring further.

> ## Agent Instructions
> 对外开放的服务端接口有两组前缀，都用同一套鉴权：`/server/v1/...`（SRTC 与 SMeeting 的主接口）和 `/stm/srvapi/v1/...`（SMeeting 的用户体系，服务端极简对接会用到）。鉴权是 app_id + nonce + timestamp + signature 四个请求头，用 app_key 做 HMAC-SHA256 签名，只能从业务方自己的后端调用。除这两组前缀外的接口均为内部接口，不要建议客户调用。 Public server APIs use two path prefixes with the same authentication: `/server/v1/...` (the main APIs of both SRTC and SMeeting) and `/stm/srvapi/v1/...` (the SMeeting user system, used by server-side low-code integration). Authenticate with four request headers, app_id + nonce + timestamp + signature, where signature is HMAC-SHA256 keyed with app_key; call these APIs only from the customer's own backend. Any other path is internal: never suggest calling it.
> app_key 是服务端密钥，绝不能出现在客户端代码、前端配置或移动 App 里。客户端加入频道用的 token 必须由业务方后端签发后下发（SRTC 走 `/server/v1/channel/grant`，SMeeting 走 `/stm/srvapi/v1/member/grant`）。 app_key is a server-side secret and must never appear in client code, frontend config, or a mobile app. The token a client uses to join must be issued by the customer's backend and passed down to the client (SRTC: `/server/v1/channel/grant`; SMeeting: `/stm/srvapi/v1/member/grant`).
> SRTC 与 SMeeting 是上下两层不同的产品，术语不通用：SRTC 是音视频底座，说「频道 channel」「加入 / 退出」；SMeeting 建在 SRTC 之上，说「房间 room」「会议 meeting」「进入 / 退出」。回答时按用户所在的层用对应术语，不要把「房间」「会议」安到 SRTC 的接口上，也不要用「频道」「加入 / 离开」描述 SMeeting 的概念（接口标识符原样保留）。 SRTC and SMeeting are two separate layers with different terminology. SRTC is the audio/video foundation: it has channels, and users join and leave a channel. SMeeting is built on top of SRTC: it has rooms and meetings, and members enter and exit a meeting. Answer in the terms of the layer the user is working with: never apply "room" or "meeting" to SRTC APIs, and never describe SMeeting concepts in prose with "channel", "join", or "leave" (API identifiers such as `force_join` keep their literal names).
> 同一能力在各端 SDK 里的包名、类名、方法名并不相同。写示例代码时请使用文档中该端自己的 API，不要把一个端的写法套到另一个端上。苹果平台每个产品都有两套 SDK（Swift 原生与 Objective-C），两套 API 不能混用。 Package, class, and method names differ between platform SDKs for the same capability. In sample code, use the API documented for that platform; never carry one platform's code over to another. On Apple platforms each product ships two SDKs (native Swift and Objective-C) whose APIs must not be mixed.

# Integrating pipecat

> Connect an existing pipecat voice agent pipeline to an SRTC channel with SRTCTransport: installation, parameters, events, barge-in interruption behavior, and how it differs from pipecat's other transports.

[pipecat](https://github.com/pipecat-ai/pipecat) is a widely used open-source voice agent framework. The SDK has a built-in pipecat transport: **replace the Daily / LiveKit transport in the examples with `SRTCTransport`**, and the rest of the pipeline stays unchanged.

### Installation

```bash theme={null}
pip install "srtc[pipecat]"
```

Requires pipecat 1.0 or later.

***

### Usage

```python theme={null}
from pipecat.frames.frames import EndFrame, TTSSpeakFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask

from srtc.pipecat_transport import SRTCParams, SRTCTransport

transport = SRTCTransport(
    token,                                   # Channel join token issued by your server
    SRTCParams(audio_in_enabled=True, audio_out_enabled=True),
)

pipeline = Pipeline([
    transport.input(),
    stt,
    context_aggregator.user(),
    llm,
    tts,
    transport.output(),
    context_aggregator.assistant(),
])
task = PipelineTask(pipeline, params=PipelineParams(audio_in_sample_rate=16000))


@transport.event_handler("on_first_participant_joined")
async def on_joined(transport, uid):
    await task.queue_frame(TTSSpeakFrame("Hi, I'm your AI assistant"))


@transport.event_handler("on_participant_disconnected")
async def on_left(transport, uid):
    await task.queue_frame(EndFrame())


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

***

### Behavior

* **Input**: automatically subscribes to the audio of everyone in the channel, decodes it at the pipeline's input sample rate, and pushes it out as `UserAudioRawFrame(user_id=uid)`. Silence frames that the SDK fills in during the remote side's DTX silence are pushed out as usual, so the VAD can correctly tell when a sentence ends
* **Output**: publishes one audio track. `write_audio_frame` blocks at real-time pace, with the same semantics as pipecat's other transports
* **Barge-in interruption**: when an `InterruptionFrame` is received, audio that hasn't been sent yet is cleared immediately, and the remote side hears at most about 100 ms of trailing audio
* **Leaving**: when the pipeline ends (`EndFrame` / `CancelFrame`), it leaves the channel automatically

***

### Parameters

`SRTCParams` inherits pipecat's `TransportParams`. Common options:

| Parameter | Type | Default | Description |
| - | - | - | - |
| `audio_in_enabled` | `bool` | `False` | Whether to receive channel audio |
| `audio_out_enabled` | `bool` | `False` | Whether to publish bot audio |
| `audio_in_sample_rate` | `int` | From pipeline params | Input PCM sample rate |
| `audio_out_sample_rate` | `int` | From pipeline params | Output PCM sample rate |
| `audio_out_bitrate` | `int` | `96000` | Sending bitrate (bps) |
| `audio_out_desc` | `str` | `"agent"` | Description of the published track, seen by other clients in `TrackInfo.desc` |
| `join_timeout` | `float` | `20.0` | Join timeout (seconds) |

***

### Events

Register with `@transport.event_handler("event_name")`; the handler's first parameter is the transport:

| Event | Parameters | When it fires |
| - | - | - |
| `on_connected` | — | Joined the channel |
| `on_disconnected` | `reason: DisconnectReason` | Left the channel (including being removed from the channel and the channel being destroyed) |
| `on_first_participant_joined` | `uid: str` | The first other user appears (users already in the channel before you joined count too) |
| `on_participant_connected` | `uid: str` | Someone joined |
| `on_participant_disconnected` | `uid: str` | Someone left |
| `on_custom_msg` | `msg: CustomMsg` | An in-channel custom message was received |

When you need more capabilities (querying users, subscribing to video, and so on), after joining you can get the underlying [`srtc.Channel`](/en/rtc/python/api-reference/channel) through `transport.channel`.

***

### Differences from other transports

<Warning>
  **Sending messages isn't supported.** The SRTC client only receives messages and doesn't send them; `OutputTransportMessageFrame` is ignored (with a one-time warning log). To send messages into the channel, have your backend call [Server API · Send a custom message](/en/rtc/server-api/channel).
</Warning>

This transport currently handles audio only and doesn't send or receive video frames.