### WebRTC Stream Proxy API Requests

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

Example cURL commands to create WebRTC stream proxies using SFU or P2P signaling.

```bash
# WHIP/WHEP 模式拉流
curl -X POST "http://127.0.0.1/index/api/addStreamProxy" \
  -d "secret=your_secret" \
  -d "vhost=__defaultVhost__" \
  -d "app=live" \
  -d "stream=test" \
  -d "url=webrtc://source.server.com:80/live/source_stream?signaling_protocols=0"

# P2P 模式拉流
curl -X POST "http://127.0.0.1/index/api/addStreamProxy" \
  -d "secret=your_secret" \
  -d "vhost=__defaultVhost__" \
  -d "app=live" \
  -d "stream=test" \
  -d "url=webrtc://signaling.server.com:3000/live/source_stream??signaling_protocols=1%26peer_room_id=target_room_id"
```

--------------------------------

### Get WebRTC Proxy Player Info API

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

Retrieves connection information and status for WebRTC proxy players.

```APIDOC
## GET /index/api/getWebrtcProxyPlayerInfo

### Description
Gets the connection information and status of WebRTC proxy players.

### Method
GET

### Endpoint
/index/api/getWebrtrcProxyPlayerInfo

### Parameters
#### Query Parameters
- **secret** (string) - Required - API access secret.
- **key** (string) - Required - Identifier for the proxy player.

### Response
(Success/Error response details not provided in the source text)
```

--------------------------------

### Build Docker Image from Source

Source: https://zlmediakit.github.io/ZLMediaKit

Compile a custom Docker image using the provided build script.

```bash
bash build_docker_images.sh
```

--------------------------------

### Configure STUN/TURN Service Credentials

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

Set the username and password for the STUN/TURN service in the `[rtc]` section of `config.ini`. These are used for ICE authentication.

```ini
#STUN/TURN 服务Ice密码
iceUfrag=ZLMediaKit
icePwd=ZLMediaKit
```

--------------------------------

### Configure WebRTC Signaling and ICE Ports

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

Configure the signaling server port, STUN/TURN server port, and whether to enable TURN services in the `[rtc]` section of `config.ini`.

```ini
[rtc]
#webrtc 信令服务器端口
signalingPort=3000
#STUN/TURN服务器端口
icePort=3478
#STUN/TURN端口是否使能TURN服务
enableTurn=1
```

--------------------------------

### Deploy ZLMediaKit via Docker

Source: https://zlmediakit.github.io/ZLMediaKit

Run the latest ZLMediaKit container using the master branch image.

```bash
docker run -id -p 1935:1935 -p 8080:80 -p 8443:443 -p 8554:554 -p 10000:10000 -p 10000:10000/udp -p 8000:8000/udp -p 9000:9000/udp zlmediakit/zlmediakit:master
```

--------------------------------

### Push RTSP Stream to WHIP Server

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

Use this command to push an RTSP stream to a WHIP-compatible server. Ensure the secret, vhost, app, and stream parameters are correctly set.

```bash
curl -X POST "http://127.0.0.1/index/api/addStreamPusherProxy" \
  -d "secret=your_secret" \
  -d "schema=rtsp" \
  -d "vhost=__defaultVhost__" \
  -d "app=live" \
  -d "stream=test" \
  -d "dst_url=webrtc://target.server.com:80/live/target_stream?signaling_protocols=0"
```

--------------------------------

### Configure OBS for SRT Publishing

Source: https://zlmediakit.github.io/ZLMediaKit/srt/srt.html

Use this URL format in OBS to publish a stream to ZLMediaKit.

```text
srt://192.168.1.105:9000?streamid=#!::r=live/test,m=publish
```

--------------------------------

### Configure VLC for SRT Pulling

Source: https://zlmediakit.github.io/ZLMediaKit/srt/srt.html

Settings for VLC media player to pull an SRT stream.

```text
#!::r=live/test
```

```text
srt://192.168.1.105:9000
```

--------------------------------

### WebRTC SFU Architecture Diagram

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

Visual representation of the SFU mode using WHIP for ingestion and WHEP for playback.

```text
                    WebRTC SFU 模式 (WHIP/WHEP)
                         
     推流端 (WHIP)                                          拉流端 (WHEP)
   +----------------+                                 +-----------------+
   |   Encoder      |                                 |   Player        |
   |  (Browser/ZLM) |                                 |  (Browser/ZLM)  |
   +----------------+                                 +-----------------+
          |                                               |
          | WHIP Protocol                                 | WHEP Protocol
          | (WebRTC ingest)                               | (WebRTC playback)
          |                                               |
          v                                               v
    +-------------------------------------------------------------------+
    |                          ZLMediaKit Server                        |
    +-------------------------------------------------------------------+
    - WHIP: WebRTC-HTTP Ingestion Protocol (推流)
    - WHEP: WebRTC-HTTP Egress Protocol (拉流)
```

--------------------------------

### Publish SRT Stream with FFmpeg

Source: https://zlmediakit.github.io/ZLMediaKit/srt/srt.html

Command to push a TS file to the ZLMediaKit server via SRT.

```bash
ffmpeg -re -stream_loop -1 -i test.ts -c:v copy -c:a copy -f mpegts srt://192.168.1.105:9000?streamid=#!::r=live/test,m=publish
```

--------------------------------

### WebRTC P2P Architecture Diagram

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

Visual representation of the P2P mode using WebSocket-based signaling.

```text
                    WebRTt WC P2P 模式
                         
    客户端 A                                      客户端 B
   +------------+                                +-------------+
   | Browser/ZLM|                                | Browser/ZLM |
   +------------+                                +-------------+
        |                                               |
        | 1. 信令交换 (SDP Offer/Answer)                 |
        | 2. ICE Candidate 交换                         |
        +----------------  -----+-----------------------+
        |                       |                       |
        |           +-----------------------+           |
        |           | ZLMediaKit Server     |           |
        |           | 信令服务器 (WebSocket) |           |
        |           | STUN 服务器            |           |
        |           | TURN 服务器            |           |
        |           +-----------------------+           |
        |                                               |
        +-----------------------------------------------+
                            直接P2P连接
```

--------------------------------

### Pull SRT Stream with FFplay

Source: https://zlmediakit.github.io/ZLMediaKit/srt/srt.html

Command to play an SRT stream from the ZLMediaKit server.

```bash
ffplay -i srt://192.168.1.105:9000?streamid=#!::r=live/test
```

--------------------------------

### WebRTC Room Management API

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

APIs for managing WebRTC rooms and their associated signaling servers.

```APIDOC
## POST /index/api/addWebrtcRoomKeeper

### Description
Adds a WebRTC signaling server to maintain room connections.

### Method
POST

### Endpoint
/index/api/addWebrtcRoomKeeper

### Parameters
#### Request Body
- **secret** (string) - Required - API access key
- **server_host** (string) - Required - Signaling server host address
- **server_port** (integer) - Required - Signaling server port
- **room_id** (string) - Required - Room ID, checked for uniqueness by the signaling server

## POST /index/api/delWebrtcRoomKeeper

### Description
Deletes a specified signaling server.

### Method
POST

### Endpoint
/index/api/delWebrtcRoomKeeper

### Parameters
#### Request Body
- **secret** (string) - Required - API access key
- **room_key** (string) - Required - Unique identifier of the room keeper

## POST /index/api/listWebrtcRoomKeepers

### Description
Lists all configured signaling servers.

### Method
POST

### Endpoint
/index/api/listWebrtcRoomKeepers

### Parameters
#### Request Body
- **secret** (string) - Required - API access key
```

--------------------------------

### Configure TURN Port Range and ICE Transport Policy

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

Define the port range for TURN service allocation and set the ICE transport policy (e.g., Relay only, P2P only) in the `[rtc]` section of `config.ini`.

```ini
#TURN服务分配端口池
portRange=50000-65000

#ICE传输策略：0=不限制(默认)，1=仅支持Relay转发，2=仅支持P2P直连
iceTransportPolicy=0
```

--------------------------------

### WebRTC Pusher Proxy URL Formats

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

URL schemes for configuring WebRTC push proxies in SFU and P2P modes.

```text
# HTTP
webrtc://target_server:port/app/stream_id?signaling_protocols=0
   
# HTTPS (暂未实现)
webrtcs://target_server:port/app/stream_id?signaling_protocols=0
```

```text
# WebSocket 
webrtc://signaling_server:port/app/stream_id?signaling_protocols=1&peer_room_id=target_room
```

--------------------------------

### Push RTSP Stream to P2P Room

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

This command pushes an RTSP stream to a P2P room using WebSocket signaling. The `peer_room_id` parameter is crucial for specifying the target room.

```bash
curl -X POST "http://127.0.0.1/index/api/addStreamPusherProxy" \
  -d "secret=your_secret" \
  -d "schema=rtsp" \
  -d "vhost=__defaultVhost__" \
  -d "app=live" \
  -d "stream=test" \
  -d "dst_url=webrtc://signaling.server.com:3000/live/room_stream?signaling_protocols=1%26peer_room_id=target_room_id"
```

--------------------------------

### WebRTC Stream Proxy API

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

APIs for creating WebRTC stream proxies for pulling and pushing streams.

```APIDOC
## POST /index/api/addStreamProxy

### Description
Creates a WebRTC pull stream proxy. Supports two signaling protocol modes: WHIP/WHEP (SFU) and WebSocket P2P.

### Method
POST

### Endpoint
/index/api/addStreamProxy

### Parameters
#### Request Body
- **secret** (string) - Required - API access key
- **vhost** (string) - Optional - Virtual host name, defaults to `__defaultVhost__`
- **app** (string) - Required - Application name
- **stream** (string) - Required - Stream ID
- **url** (string) - Required - WebRTC source URL. Supported formats:
  1. WHIP/WHEP (SFU): `webrtc://server_host:server_port/app/stream_id?signaling_protocols=0`
  2. WebSocket P2P: `webrtc://signaling_server_host:signaling_server_port/app/stream_id?signaling_protocols=1&peer_room_id=target_room_id`

### Request Example
```
# WHIP/WHEP Mode Pull
curl -X POST "http://127.0.0.1/index/api/addStreamProxy" \
  -d "secret=your_secret" \
  -d "vhost=__defaultVhost__" \
  -d "app=live" \
  -d "stream=test" \
  -d "url=webrtc://source.server.com:80/live/source_stream?signaling_protocols=0"

# P2P Mode Pull
curl -X POST "http://127.0.0.1/index/api/addStreamProxy" \
  -d "secret=your_secret" \
  -d "vhost=__defaultVhost__" \
  -d "app=live" \
  -d "stream=test" \
  -d "url=webrtc://signaling.server.com:3000/live/source_stream??signaling_protocols=1%26peer_room_id=target_room_id"
```

## POST /index/api/addStreamPusherProxy (Not Implemented)

### Description
Creates a WebRTC push stream proxy. This endpoint is currently not implemented.

### Method
POST

### Endpoint
/index/api/addStreamPusherProxy

### Parameters
#### Request Body
- **secret** (string) - Required - API access key
- **schema** (string) - Required - Source stream protocol (e.g., rtmp, rtsp, hls)
- **vhost** (string) - Required - Virtual host name
- **app** (string) - Required - Application name
- **stream** (string) - Required - Source stream ID
- **dst_url** (string) - Required - Destination WebRTC push URL. Supported formats:
  1. WHIP Mode (SFU): `webrtc://target_server:port/app/stream_id?signaling_protocols=0`
  2. WebSocket P2P Mode: `webrtc://signaling_server:port/app/stream_id?signaling_protocols=1&peer_room_id=target_room`
```

--------------------------------

### WebRTC URL Formats for Stream Proxy

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

URL schemes for configuring WebRTC stream proxies in SFU and P2P modes.

```text
# HTTP
webrtc://server_host:server_port/app/stream_id?signaling_protocols=0
   
# HTTPS (暂未实现）
webrtcs://server_host:server_port/app/stream_id?signaling_protocols=0
```

```text
# WebSocket
webrtc://signaling_server_host:signaling_server_port/app/stream_id?signaling_protocols=1&peer_room_id=target_room_id
   
# WebSocket Secure (暂未实现)
webrtcs://signaling_server_host:signaling_server_port/app/stream_id?signaling_protocols=1&peer_room_id=target_room_id
```

--------------------------------

### Add Stream Pusher Proxy API

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

This API allows you to push RTSP streams to a WHIP server or a P2P room using WebRTC.

```APIDOC
## POST /index/api/addStreamPusherProxy

### Description
Adds a stream pusher proxy to send RTSP streams to a specified destination URL, supporting both WHIP/WHEP and WebSocket P2P modes.

### Method
POST

### Endpoint
/index/api/addStreamPusherProxy

### Parameters
#### Request Body
- **secret** (string) - Required - API access secret.
- **schema** (string) - Required - Schema of the source stream (e.g., "rtsp").
- **vhost** (string) - Required - Virtual host of the source stream.
- **app** (string) - Required - Application name of the source stream.
- **stream** (string) - Required - Stream name of the source stream.
- **dst_url** (string) - Required - Destination URL for the WebRTC stream. This can be a WHIP/WHEP endpoint or a P2P room URL.

### Request Example
```json
{
  "secret": "your_secret",
  "schema": "rtsp",
  "vhost": "__defaultVhost__",
  "app": "live",
  "stream": "test",
  "dst_url": "webrtc://target.server.com:80/live/target_stream?signaling_protocols=0"
}
```

### Request Example (P2P)
```json
{
  "secret": "your_secret",
  "schema": "rtsp",
  "vhost": "__defaultVhost__",
  "app": "live",
  "stream": "test",
  "dst_url": "webrtc://signaling.server.com:3000/live/room_stream?signaling_protocols=1&peer_room_id=target_room_id"
}
```

### URL Parameters Description for dst_url
- **signaling_protocols** (integer) - Signaling protocol type.
  - `0`: WHIP/WHEP mode (default). Protocol: Standard WebRTC signaling protocol based on HTTP. Use case: SFU mode, suitable for broadcasting and multi-party conferences.
  - `1`: WebSocket P2P mode. Protocol: Custom signaling protocol based on WebSocket. Use case: Peer-to-peer direct connection, suitable for low-latency calls and private communication.
- **peer_room_id** (string) - Target room ID in P2P mode (required only for P2P mode).

### Response
(Success/Error response details not provided in the source text)
```

--------------------------------

### WebRTC Room Session Management API

Source: https://zlmediakit.github.io/ZLMediaKit/webrtc/USAGE.html

API for listing active WebRTC peer sessions.

```APIDOC
## POST /index/api/listWebrtcRooms

### Description
Lists all active WebRTC Peer session information.

### Method
POST

### Endpoint
/index/api/listWebrtcRooms

### Parameters
#### Request Body
- **secret** (string) - Required - API access key
```

=== COMPLETE CONTENT === This response contains all available snippets from this library. No additional content exists. Do not make further requests.