> ## 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.

# FAQ

> Common Web SDK problems and fixes: camera/microphone permission denied, can receive but not publish in local development, autoplay blocked by the browser, screen sharing limits on Safari, TokenExpired when joining, and joining multiple channels at once.

### What if the browser says "access to the camera/microphone is not allowed"?

**Cause:** the browser permission was denied, or the page isn't in a secure context (HTTPS / localhost).

**Steps to fix:**

1. Make sure the page is accessed over HTTPS or localhost (plain HTTP domains can't access media devices)
2. Click the camera/microphone icon on the right side of the browser address bar and change the permission to "Allow"
3. If you previously chose "Always block", reset the permission manually in the browser's site settings:
   * Chrome: "🔒" on the left of the address bar → "Site settings" → "Camera/Microphone" → "Allow"
4. Refresh the page and try again

You can check in advance whether the environment can capture media through the `secure` and `mediaDevices` fields of `srtc.getEnvInfo()`:

```typescript theme={null}
const env = srtc.getEnvInfo();
if (!env.secure) {
  alert('Please access this page over HTTPS or localhost');
} else if (!env.mediaDevices) {
  alert('This browser does not support media device access. Please check your permission settings');
}
```

***

### What if I can receive but not publish during local development?

**Cause:** most likely the page is running over plain HTTP, and the browser restricts WebRTC publishing.

**Solution:** access your local development environment via `http://localhost` or `http://127.0.0.1`, not `http://[local LAN IP]` (publishing isn't supported over that protocol).

For protocol and publishing support, see [Integration - URL protocol restrictions](/en/rtc/web/integration#url-protocol-restrictions).

***

### How do I handle video/audio autoplay being blocked by the browser?

The browser's autoplay policy forbids playing media without a user gesture; the symptom is no audio or no video after calling `startPlay()`.

The SDK notifies you through the `TRACK_AUTOPLAY_FAIL` event:

```typescript theme={null}
srtc.onNotifyChannelEvent = (evt) => {
  if (evt.type === ChannelEventType.TRACK_AUTOPLAY_FAIL) {
    // Option 1: the SDK's built-in guide dialog (on by default)
    // Playback starts automatically after the user clicks the dialog

    // Option 2: handle it yourself and play again when the user clicks a button
    const failedTrack = evt.data; // BaseTrack
    document.querySelector('#play-btn')!.addEventListener('click', async () => {
      if (failedTrack instanceof RemoteAudioMixTrack) {
        await failedTrack.startPlay({ disableAutoPlayDialog: true });
      }
    }, { once: true });
  }
};
```

> **Best practice:** call `join` and `startPlay` after the user actively clicks a "Join" or "Start call" button; this user gesture context usually gets around the autoplay restriction.

***

### Is screen sharing unavailable on Safari?

Safari 15.4 and above support `getDisplayMedia` (screen sharing), with these limitations:

* Capturing **system audio** isn't supported (`getAudioTrack()` returns `undefined`)
* **Screen sharing isn't supported** on iOS Safari (a system limitation)
* Some older Safari versions (\< 15.4) don't support screen sharing at all

You can check in advance whether the current environment supports it with `srtc.getEnvInfo().screenshare`:

```typescript theme={null}
const env = srtc.getEnvInfo();
if (!env.screenshare) {
  alert('This browser does not support screen sharing. Please use Chrome or desktop Safari 15.4+');
}
```

***

### What if joining the channel fails with TokenExpired?

The token is issued by the server and has a short validity period (usually from a few minutes to a few hours, decided by the issuer).

**Solution:**

* Get a fresh token from the server right before calling `srtc.join()`; don't cache old tokens
* Check that the client and server system clocks are in sync (clock skew can make the token expire early)

***

### How do I join multiple channels at once?

Call `join` multiple times on one `SRTC` instance. Each `join` returns that channel's Channel object, and each channel is operated independently:

```typescript theme={null}
const channel1 = await srtc.join(token1);
const channel2 = await srtc.join(token2);

await channel1.publishLocalTrack(track);
await channel2.leave(); // Leave channel 2 without affecting channel 1
```

For event isolation, publishing one captured track to multiple channels, and more, see [Multi-channel](/en/rtc/web/advanced/multi-channel).

Note: running multiple channels in parallel uses more bandwidth and system resources, so use it only when needed.