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7e49007d16
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| 7e49007d16 |
2
.gitignore
vendored
2
.gitignore
vendored
@ -23,5 +23,3 @@
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hs_err_pid*
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hs_err_pid*
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objs
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objs
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.idea
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.idea
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run
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@ -575,16 +575,11 @@ SIP消息应支持基于UDP和TCP的传输。
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## 4.3.5媒体回放控制协议
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## 4.3.5媒体回放控制协议
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历史视音频的回放控制命令应采用MANSRTSP,协议描述应符合附录B的规定,实现设备在端到
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历史视音频的回放控制命令应采用MANSRTSP,协议描述应符合附录B的规定,实现设备在端到 端之间对视音频流的正常、快速、暂停、停止、随机拖动播放等远程控制。历史媒体的回放控制命令采用 SIP消息INFO的消息体携带传输。
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端之间对视音频流的正常、快速、暂停、停止、随机拖动播放等远程控制。历史媒体的回放控制命令采用 SIP消息INFO的消息体携带传输。
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## 4.3.6媒体传输和媒体编解码协议
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## 4.3.6媒体传输和媒体编解码协议
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媒体流在联网系统 IP 网络上传输时应支持
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值功能。RTP的负载应采用如下两种格式之一:基于PS封装的视音频数据或视音频基本流数据,应符 间戳信息及各数据流的同步,同一帧视音频数据包封装成的所有RTP数据包的RTP时间戳相同,且与 议,为按序传输数据包提供可靠保证,提供流量控制和拥塞控制。
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RTP传输,媒体流发送源端应支持控制媒体流发送峰值功能。RTP的负载应采用如下两种格式之一:基于PS封装的视音频数据或视音频基本流数据,应符合附录C的规定。媒体流的传输应采用
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IETF RFC 3550规定的RTP 协议,提供实时数据传输中的时间戳信息及各数据流的同步,同一帧视音频数据包封装成的所有 RTP数据包的RTP
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时间戳相同,且与不同帧视音频数据包的RTP数据包的RTP 时间戳不同;宣采用 IETF RFC 3550规定的 RTCP
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协议,为按序传输数据包提供可靠保证,提供流量控制和拥塞控制。
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## 5传输要求
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## 5传输要求
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@ -1,78 +1,4 @@
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# note to me
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```bash
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```bash
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docker compose build --network host
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docker compose build --network host
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```
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```
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```
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docker compose up -d --force-recreate
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```
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# TODO
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- [ ] let user choose whether use mirror (use which mirror) when building Dockerfile
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---
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Based on the logs and the **GB/T 28181-2022** standard you provided, here is the explanation:
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Yes, this is **SIP**, but the content *inside* the SIP message is **SDP (Session Description Protocol)**.
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While XML is used for *control* (like PTZ), **SDP** is used for **Media Negotiation** (setting up the video stream).
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### The Architecture from your logs
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1. **SIP (The Envelope):** Starts the conversation ("I want to watch video").
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2. **SDP (The Letter inside):** Describes technical details ("Send video to IP X, Port Y, using Format Z").
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3. **RTP (The result):** After this SIP/SDP handshake finishes, the actual binary video stream (PS/H.264) starts flowing over a separate TCP/UDP connection.
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### Breakdown of your Log
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This log shows a **Real-time Live View** handshake.
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#### 1. The Request (SRS Server -> Camera)
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The Server asks the Camera to send video.
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```ini
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INVITE sip:34020000001320000001@3402000000 SIP/2.0
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Content-Type: application/sdp
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s=Play # "Play" = Real-time Live View (Standard 9.2.2.1)
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c=IN IP4 192.168.2.184 # The Media Server IP
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m=video 9000 TCP/RTP/AVP 96 # "Send video to my Port 9000 via TCP"
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a=recvonly # "I will only receive, not send"
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y=0911024252 # **GB/T 28181 Special**: The SSRC (Stream ID)
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```
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#### 2. The Response (Camera -> SRS Server)
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The Camera agrees and tells the server its own details.
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```ini
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SIP/2.0 200 OK
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Content-Type: application/sdp
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c=IN IP4 192.168.2.64 # The Camera IP
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m=video 15060 TCP/RTP/AVP 96 # "I am sending from Port 15060"
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a=sendonly # "I will only send"
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a=setup:active # "I will initiate the TCP connection to you"
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y=0911024252 # Matches the SSRC provided
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f=v/2/2560x1440/25/2/8192a/... # **GB/T 28181 Special**: Media Info
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```
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### Key Differences from Standard SDP
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GB/T 28181 modifies standard SDP with two specific fields mandatory for this protocol:
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1. **`y=` (SSRC)**:
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* **Standard SDP:** Does not have a `y` line.
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* **GB/T 28181:** Uses `y` to define the **SSRC** (Synchronization Source). This 10-digit number is crucial because it marks every binary video packet sent later. If the binary stream headers don't match this `y` value, the stream is rejected.
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2. **`f=` (Media Info)**:
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* **Standard SDP:** Does not have an `f` line.
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* **GB/T 28181:** Uses `f` to describe video parameters. In your log: `v/2/2560x1440/25...` means:
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* `v`: Video
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* `2`: Coding format (likely H.264 or H.265 mapped)
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* `2560x1440`: Resolution
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* `25`: Frame rate
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### Summary of Cooperation
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1. **XML (SIP MESSAGE):** Used for "remote control" (PTZ, Query, Keepalive).
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2. **SDP (SIP INVITE):** Used to *negotiate* the pipeline.
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3. **Binary (RTP/PS):** The actual heavy video data that flows through the pipe created by the SDP.
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@ -277,8 +277,8 @@ SRS-SIP 是一个基于 GB28181 协议的视频监控系统,提供设备管理
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- `down`: 向下
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- `down`: 向下
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- `left`: 向左
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- `left`: 向左
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- `right`: 向右
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- `right`: 向右
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- `zoomin`: 放大
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- `zoom_in`: 放大
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- `zoomout`: 缩小
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- `zoom_out`: 缩小
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- `stop`: 停止
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- `stop`: 停止
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- `speed`: 控制速度(1-9)
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- `speed`: 控制速度(1-9)
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@ -14,10 +14,7 @@ services:
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- ./run/conf/config.yaml:/usr/local/srs-sip/config.yaml:ro
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- ./run/conf/config.yaml:/usr/local/srs-sip/config.yaml:ro
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- ./run/logs:/usr/local/srs-sip/logs
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- ./run/logs:/usr/local/srs-sip/logs
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- ./run/srs/conf/srs.conf:/usr/local/srs/conf/srs.conf:ro
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- ./run/srs/conf/srs.conf:/usr/local/srs/conf/srs.conf:ro
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# use docker logs
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- ./run/srs/logs:/var/log/srs/
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- ./run/srs/logs:/var/log/srs/
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# for recording
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- ./run/data:/data
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ports:
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ports:
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# SRS RTMP
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# SRS RTMP
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- "1985:1985"
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- "1985:1985"
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@ -1,6 +1,6 @@
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<script setup lang="ts">
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<script setup lang="ts">
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import { ref, computed } from 'vue'
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import { ref, computed } from 'vue'
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import { ArrowRight, VideoCamera, ZoomIn, ZoomOut } from '@element-plus/icons-vue'
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import { ArrowRight, VideoCamera } from '@element-plus/icons-vue'
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import {
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import {
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ArrowUp,
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ArrowUp,
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ArrowDown,
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ArrowDown,
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@ -158,27 +158,6 @@ const isDisabled = computed(() => !props.activeWindow)
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<div class="direction-center"></div>
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<div class="direction-center"></div>
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</div>
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</div>
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</div>
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</div>
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<div class="zoom-controls">
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<el-button
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class="zoom-btn"
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:disabled="isDisabled"
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@mousedown="handlePtzStart('zoomin')"
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@mouseup="handlePtzStop"
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@mouseleave="handlePtzStop"
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>
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<el-icon><ZoomIn /></el-icon>
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</el-button>
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<div class="zoom-label">变倍</div>
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<el-button
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class="zoom-btn"
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:disabled="isDisabled"
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@mousedown="handlePtzStart('zoomout')"
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@mouseup="handlePtzStop"
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@mouseleave="handlePtzStop"
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>
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<el-icon><ZoomOut /></el-icon>
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</el-button>
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</div>
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<div class="speed-control">
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<div class="speed-control">
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<div class="speed-value">{{ speed }}</div>
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<div class="speed-value">{{ speed }}</div>
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<el-slider
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<el-slider
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@ -367,48 +346,6 @@ const isDisabled = computed(() => !props.activeWindow)
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border-radius: 4px;
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border-radius: 4px;
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}
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}
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.zoom-controls {
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display: flex;
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flex-direction: column;
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align-items: center;
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gap: 6px;
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height: 120px;
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justify-content: center;
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}
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.zoom-btn {
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--el-button-bg-color: var(--el-color-primary-light-8);
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--el-button-border-color: var(--el-color-primary-light-5);
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--el-button-hover-bg-color: var(--el-color-primary-light-7);
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--el-button-hover-border-color: var(--el-color-primary-light-4);
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--el-button-active-bg-color: var(--el-color-primary-light-5);
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--el-button-active-border-color: var(--el-color-primary);
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width: 36px;
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height: 36px;
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padding: 0;
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margin: 0;
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border-radius: 4px;
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.el-icon {
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font-size: 18px;
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}
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&:hover {
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transform: scale(1.05);
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}
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&:active {
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transform: scale(0.95);
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}
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}
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.zoom-label {
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font-size: 12px;
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color: var(--el-text-color-secondary);
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font-weight: 500;
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}
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.control-groups,
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.control-groups,
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.control-group {
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.control-group {
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display: none;
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display: none;
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20
run/conf/config.yaml
Normal file
20
run/conf/config.yaml
Normal file
@ -0,0 +1,20 @@
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# 通用配置
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common:
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# [debug, info, warn, error]
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log-level: "info"
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log-file: "logs/srs-sip.log"
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# GB28181配置
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gb28181:
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serial: "34020000002000000001"
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realm: "3402000000"
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host: "0.0.0.0"
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port: 5060
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auth:
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enable: false
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password: "123456"
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# HTTP服务配置
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http:
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listen: 8025
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dir: ./html
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60
run/srs/conf/srs.conf
Normal file
60
run/srs/conf/srs.conf
Normal file
@ -0,0 +1,60 @@
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listen 1935;
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max_connections 1000;
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# For docker, please use docker logs to manage the logs of SRS.
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# See https://docs.docker.com/config/containers/logging/
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srs_log_tank console;
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daemon off;
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disable_daemon_for_docker off;
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http_api {
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enabled on;
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listen 1985;
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raw_api {
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enabled on;
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allow_reload on;
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}
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}
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http_server {
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enabled on;
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listen 8080;
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dir ./objs/nginx/html;
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}
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stream_caster {
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enabled on;
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caster gb28181;
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output rtmp://127.0.0.1/live/[stream];
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listen 9000;
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sip {
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enabled off;
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}
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}
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rtc_server {
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enabled on;
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listen 8000; # UDP port
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
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candidate $CANDIDATE;
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# Disable for Oryx.
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use_auto_detect_network_ip off;
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api_as_candidates off;
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}
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vhost __defaultVhost__ {
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http_remux {
|
||||||
|
enabled on;
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mount [vhost]/[app]/[stream].flv;
|
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}
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rtc {
|
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|
enabled on;
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|
nack on;
|
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|
twcc on;
|
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|
stun_timeout 30;
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dtls_role passive;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
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rtmp_to_rtc on;
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keep_bframe off;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
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rtc_to_rtmp on;
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pli_for_rtmp 6.0;
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||||||
|
}
|
||||||
|
}
|
||||||
Reference in New Issue
Block a user