> ## 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.

# RTSP streams

> Bring any IP camera, encoder, NVR, or media server that outputs an RTSP URL into meetings: register the stream URL in the admin console, choose UDP or TCP, look up URL formats for common brands, and invite the stream into the meeting.

As long as a device can provide an RTSP URL (`rtsp://...`), you can pull its video into a meeting. The device doesn't need to support GB28181
or register—the device gateway pulls the stream itself.

***

## 1. Prepare the stream URL

First open the stream URL in a player (such as VLC). **Continue only once it plays.** Common formats:

| Brand | Main stream URL format |
| - | - |
| Hikvision | `rtsp://username:password@IP:554/Streaming/Channels/101` |
| Dahua | `rtsp://username:password@IP:554/cam/realmonitor?channel=1&subtype=0` |
| Uniview | `rtsp://username:password@IP:554/media/video1` |

Check the device manual for specifics. If the device requires login, put the username and password in the URL.

***

## 2. Register it in the console

Admin console **Device management** → **RTSP** → **Add account**:

![Add an RTSP stream](https://mintcdn.com/freewind/EF59QLb6N-9opFfC/zh/product/device-access/images/admin-rtsp-add.png?fit=max\&auto=format\&n=EF59QLb6N-9opFfC\&q=85\&s=a6b4f344c66c7483b1acd014addcb122)

| Field | What to enter |
| - | - |
| Transport | **UDP** (default) or **TCP**. TCP is more stable when the network has packet loss, crosses subnets, or passes through firewalls |
| Stream URL | The RTSP URL verified in step 1; must start with `rtsp` |
| Display name | The name shown in the meeting |
| Device gateway | Which gateway pulls the stream; this gateway must be able to reach the device's IP |

The list after registering:

![RTSP stream list](https://mintcdn.com/freewind/EF59QLb6N-9opFfC/zh/product/device-access/images/admin-rtsp-list.png?fit=max\&auto=format\&n=EF59QLb6N-9opFfC\&q=85\&s=5df109e546a012bfdd43b6da166085b1)

RTSP streams have no online status, so `--` in the list is normal—the gateway pulls the stream only when it's invited into a meeting.

***

## 3. Invite into the meeting

* **Admin console**: meeting details → **Invite** → **Invite device** → **Surveillance cameras**
* **Meeting client**: the host clicks **Invite** → **Invite device** → **Surveillance cameras**
* **Business system**: call [Invite devices to the channel](https://docs.stmlink.com/en/rtc/server-api/agent.md#invite-devices-to-the-channel)

***

## Codecs

| | Support |
| - | - |
| H.264 video | Forwarded directly, with no quality loss and no transcoding resources used |
| H.265 video | The gateway converts it to H.264 automatically, which uses gateway CPU; evaluate server capacity for many streams |
| Audio | If the stream has audio, it enters the meeting too; the gateway transcodes automatically |

> [!TIP]
>
> Set the camera to **H.264 main stream** at **1080P or lower** to use the fewest resources. H.265 works, but every stream needs real-time transcoding.


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