# wvp-gb28181

**Repository Path**: njqt/wvp-gb28181-talk

## Basic Information

- **Project Name**: wvp-gb28181
- **Description**: 基于gb28181的摄像头实现方案代码
- **Primary Language**: Unknown
- **License**: MIT
- **Default Branch**: master
- **Homepage**: None
- **GVP Project**: No

## Statistics

- **Stars**: 0
- **Forks**: 18
- **Created**: 2024-08-17
- **Last Updated**: 2024-08-17

## Categories & Tags

**Categories**: Uncategorized

**Tags**: None

## README

# 国标28181对讲实现思路以及注意事项

标准国标对讲信令

- 流程图

![对讲流程](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2Fduijiangliucheng.png)

- 具体信令描述

  ```shell
  #广播通知 BroadCast
  #平台侧------->设备侧
  
  MESSAGE sip:34020000001110000001@3402000000 SIP/2.0
  From: <sip:34020000002000000001@3402000000>;tag=bccedfd000006
  To: <sip:34020000001110000001@3402000000>
  Content-Length: 176
  CSeq: 5 MESSAGE
  Route: <sip:34020000001110000001>
  Call-ID: 12345678900006
  Via: SIP/2.0/UDP 192.168.1.101:5060;wlsscid=377aa9afcf1b36f;branch=123133532300006
  Content-Type: Application/MANSCDP+xml
  Max-Forwards: 70
  
  <?xml version="1.0"?>
  <Notify>
  <CmdType>Broadcast</CmdType>
  <SN>17298</SN>
  <!--语音输入设备的设备编码-->
  <SourceID>34020000002000000001</SourceID>
  <!--语音输出设备的设备编码-->
  <TargetID>34020000001370000001</TargetID>
  </Notify>
  
  ```

  ```shell
  #设备测收到广播通知回复200 ok
  #设备测------>平台侧
  
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.1.101:5060;wlsscid=377aa9afcf1b36f;branch=123133532300006
  From: <sip:34020000002000000001@3402000000>;tag=bccedfd000006
  To: <sip:34020000001110000001@3402000000>;tag=880670885
  Call-ID: 12345678900006
  CSeq: 5 MESSAGE
  User-Agent: IP Camera
  Content-Length: 0
  
  ```

  ```shell
  #设备测主动发起invite
  #设备测------>平台侧
  
  INVITE sip:34020000002000000001@192.168.1.101:5060 SIP/2.0
  Via: SIP/2.0/UDP 192.168.1.64:5060;rport;branch=z9hG4bK1490643408
  From: <sip:34020000001110000001@192.168.1.64:5060>;tag=1499847265		//
  To: <sip:34020000002000000001@192.168.1.101:5060>
  Call-ID: 1420309512
  CSeq: 20 INVITE
  Contact: <sip:34020000001110000001@192.168.1.64:5060>
  Content-Type: application/sdp
  Max-Forwards: 70
  User-Agent: IP Camera
  # 媒体流发送者ID:发送方媒体流序列号,媒体流接收者ID:接收方媒体流序列号 
  Subject: 34020000002000000001:1,34020000001110000001:2		
  Content-Length:   214
  
  v=0
  o=34020000001110000001 2418 2418 IN IP4 192.168.1.64
  s=Play
  c=IN IP4 192.168.1.64
  t=0 0
  # 音频 端口  RTP-over-UDP 负载类型( 8-PCMA, 96-PS) 
  # 注意 这一步就是设备侧开的收语音流数据的端口
  # RTP/AVP传输模式是udp，这种只能在局域网对讲
  # TCP/RTP/AVP 这种才可以在公网对讲
  m=audio 15062 RTP/AVP 8 96		
  a=recvonly
  # RTP + 音频流: 负载类型 
  a=rtpmap:8 PCMA/8000			
  a=rtpmap:96 PS/90000
  # SSRC(同步信源标识符): SSRC值由媒体流发送设备所在的SIP监控域产生,作为媒体流的标识使用 
  y=0200000017	
  # v/编码格式/分辨率/帧率/码率类型/码率大小  a/编码格式/码率大小/采样率   
  #   								  G.711 / 64kbps / 8kHz
  f=v/a/1/8/1					
  ```


```shell
#平台侧回复invite ok
#平台侧------->设备侧
》》》》》》 IPC
SIP/2.0 200 OK
From: <sip:34020000001110000001@192.168.1.64:5060>;tag=1499847265
To: <sip:34020000002000000001@192.168.1.101:5060>;tag=1420309512
Call-ID: 1420309512
CSeq: 20 INVITE
User-Agent: General SIP UAS V1.0
Via: SIP/2.0/UDP 192.168.1.64:5060;rport;branch=z9hG4bK1490643408
Contact: <sip:34020000002000000001@192.168.1.101:5060>
Content-Type: application/SDP
Content-Length: 182

v=0
o=34020000002000000001 0 0 IN IP4 192.168.1.101
s=Play
c=IN IP4 192.168.1.101
t=0 0
# 这是我们平台侧开的端口，使用这个端口发送音频数据到设备
m=audio 8000 RTP/AVP 8		//  端口：8000
a=rtpmap:8 PCMA/8000		//rtpmap:8 PCMA及g711a  采样率：8000HZ
a=sendonly
y=0200000017
```



```shell
#设备测------>平台侧
ACK sip:34020000002000000001@192.168.1.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.64:5060;rport;branch=z9hG4bK106578989
From: <sip:34020000001110000001@192.168.1.64:5060>;tag=1499847265
To: <sip:34020000002000000001@192.168.1.101:5060>;tag=1420309512
Call-ID: 1420309512
CSeq: 20 ACK
Contact: <sip:34020000001110000001@192.168.1.64:5060>
Max-Forwards: 70
User-Agent: IP Camera
Content-Length: 0

#这一步过后我们就可以发送音频数据了
```

- 关闭对讲

```shell
# 发送BYE信令
BYE sip:34020000001370000001@3402000000 SIP/2.0
From: <sip:34020000002000000001@3402000000>;tag=1420309512
To: <sip:34020000001370000001@3402000000>;tag=1499847265
CSeq: 2 BYE
Call-ID: 1420309512
Via: SIP/2.0/UDP 192.168.1.64:5060;branch=z9hG4bKee5c5d98-00007
Max-Forwards: 70
Content-Length: 0
```

```
# BYE ok
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.64:5060;branch=z9hG4bKee5c5d98-00007;received=192.168.1.101
From: <sip:34020000002000000001@3402000000>;tag=1420309512
To: <sip:34020000001370000001@3402000000>;tag=1499847265
Call-ID: 1420309512
CSeq: 2 BYE
User-Agent: IP Camera
Content-Length: 0

```

大华TALK对讲模式

- 流程图

![](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2Fdahua-talk.png)

- 具体信令

  ```shell
  #主动发送invite到摄像机
  #平台侧------->设备侧
  INVITE sip:34020000001370000001@3402000000 SIP/2.0
  Via: SIP/2.0/UDP 192.168.32.33:14000;rport;branch=z9hG4bK7d0821ae
  From: <sip:34020000002000000033@3402000000>;tag=1752762308
  To: <sip:34020000001370000001@3402000000>
  Call-ID: 1949196054
  CSeq: 5 INVITE
  Content-Type: APPLICATION/SDP
  Contact: <sip:34020000002000000033@192.168.32.33:14000>
  Max-forwards: 70
  User-Agent: tiamaes
  Subject: 34020000001370000001:2,34020000002000000033:2
  Content-Length: 183
  
  v=0
  o=34020000002000000033 0 0 IN IP4 192.168.32.33
  s=Talk
  c=IN IP4 192.168.32.33
  t=0 0
  # 此处我们可以控制数据的传输方式  可以使用  TCP/RTP/AVP
  # 25000 是平台侧发送语音数据的端口
  m=audio 25000 TCP/RTP/AVP 8
  a=sendrecv
  a=rtpmap:8 PCMA/8000
  f=v/a/1/8/1
  y=0200000002
  
  ```

  ```shell
  #设备回复invite ok
  #设备测------>平台侧
  SIP/2.0 100 Trying
  Call-ID: 1949196054
  Content-Length: 0
  CSeq: 5 INVITE
  From: <sip:34020000002000000033@3402000000>;tag=1752762308
  To: <sip:34020000001370000001@3402000000>
  User-Agent: SIP UAS V3.0.0.1049939
  Via: SIP/2.0/UDP 192.168.32.33:14000;rport=14000;branch=z9hG4bK7d0821ae
  
  <!--接收到摄像机响应-->
  SIP/2.0 101 Dialog Establishment
  Call-ID: 1949196054
  Contact: <sip:34020000001180005001@192.168.32.13:5060>
  Content-Length: 0
  CSeq: 5 INVITE
  From: <sip:34020000002000000033@3402000000>;tag=1752762308
  To: <sip:34020000001370000001@3402000000>;tag=5aaf5dffc1105842785cdf093029bcf8
  User-Agent: SIP UAS V3.0.0.1049939
  Via: SIP/2.0/UDP 192.168.32.33:14000;rport=14000;branch=z9hG4bK7d0821ae
  
  <!--接收到摄像机响应-->
  SIP/2.0 200 OK
  Call-ID: 1949196054
  Contact: <sip:34020000001180005001@192.168.32.13:5060>
  Content-Length: 270
  Content-Type: application/sdp
  CSeq: 5 INVITE
  From: <sip:34020000002000000033@3402000000>;tag=1752762308
  To: <sip:34020000001370000001@3402000000>;tag=5aaf5dffc1105842785cdf093029bcf8
  User-Agent: SIP UAS V3.0.0.1049939
  Via: SIP/2.0/UDP 192.168.32.33:14000;rport=14000;branch=z9hG4bK7d0821ae
  
  v=0
  o=34020000001310000001 0 0 IN IP4 192.168.32.13
  s=Talk
  i=VCam Talk Session
  c=IN IP4 192.168.32.13
  t=0 0
  # 9712是设备测开的接收语音数据流的端口
  m=audio 9712 TCP/RTP/AVP 8	
  a=recvonly
  a=rtpmap:8 PCMA/8000/1
  m=audio 9712 TCP/RTP/AVP 8	
  a=sendonly
  a=rtpmap:8 PCMA/8000/1
  y=0200000002
  f=v/0/0/0/0/0a/0/0/0
  
  ```

  ```shell
  #发送ack到摄像机
  ACK sip:34020000001370000001@3402000000 SIP/2.0
  Via: SIP/2.0/UDP 192.168.32.33:14000;rport;branch=z9hG4bK6e554aa5
  From: <sip:34020000002000000033@3402000000>;tag=1752762308
  To: <sip:34020000001370000001@3402000000>;tag=5aaf5dffc1105842785cdf093029bcf8
  Call-ID: 1949196054
  CSeq: 5 ACK
  Contact: <sip:34020000002000000033@192.168.32.33:14000>
  Max-forwards: 70
  User-Agent: tiamaes
  Content-Length: 0
  # 这一步进行完就可以对讲了
  ```

  > **通过对比国标标准信令和大华talk信令我们可以看出，标准的国标对讲sip信令，invite是设备测发起，我们没有办法控制数据的传输模式，而大华的talk模式invite是平台侧发起，我们可以控制数据的传输模式使用udp还是tcp（当然也得设备测支持tcp传输模式才可，要不然还是白扯）**

实现思路流程图

- **文字语言描述**

1. 点击对讲按钮，客户端根据zlmediakit的规则生成webrtc推流地址
2. 调用麦克风，通过webrtc推语音流到zlmediakit
3. 如果推流成功，开始调用wvp国标信令协商接口进行对讲信令协商（说通俗点就是告诉设备测我现在要进行语音对讲，你开个端口用于接收语音数据，我开个端口用于发送语音数据），如果推流失败，则在客户端提示用户具体原因
4. 国标对讲信令协商成功，我们就可以开始讲话，设备侧就可以听到，如果失败，具体原因具体分析。
5. 再次点击按钮，停止对讲，停止推流，发送bye信令，断开连接，关闭端口

- **对讲流程图**

![](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2Fduijiangliucheng.jpg)

- **关键节点**

  1. 推流地址是如何确认的（这个是zlmeida提供的一个api接口）参考[zlmedia webrtc推流](https://github.com/ZLMediaKit/ZLMediaKit/wiki/webrtc%E4%BF%A1%E4%BB%A4%E4%BA%A4%E4%BA%92%E6%A0%BC%E5%BC%8F)

  ​     ![](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2Fzlwebrtctl.jpg)

  参数（自己定义）：

  app：audio

  stream：deviceid+channelid

  type:push

  2.信令协商完成后如何把推到zlmedia的流再推给设备，也是调用zl的api主要涉及两个api

  - udp

  ![](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2FsatrtSendRtp.jpg)

  - TCP被动模式调用这个

  ![](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2FstartSendRtpPassive.jpg)



对讲源码分析

  > 说明：涉及的api接口都在TalkbackController这个类中

  1.获取推流地址（如果开启了推流鉴权需要加上参数sign,这一块我是写死的）

  ``` java
  /**
       * 获取webrtc推流地址
       * @param deviceId
       * @param channelId
       * @return
       */
      @GetMapping("/getWebRtcAddr/{deviceId}/{channelId}")
      public WVPResult getWebRtcAddr(@PathVariable("deviceId") String deviceId, @PathVariable("channelId") String channelId) {
  
          //首先判断设备是否正在对讲
          if (redisCatchStorage.isBroadcastItem(deviceId)) {
              return WVPResult.fail(ErrorCode.ERROR603);
          }
  
          Device device = deviceService.getDevice(deviceId);
          MediaServerItem mediaServerItem = playService.getNewMediaServerItem(device);
  
          if (mediaServerItem == null) {
              logger.error("流媒体未找到");
              return WVPResult.fail(ErrorCode.ERROR600);
          }
  
          Map<String, Object> result = new HashMap<>(16);
          String app = "audio";
          String stream = deviceId + "_" + channelId;
          String type = "push";
          LoginUser userInfo = SecurityUtils.getUserInfo();
          String sign = Md5Utils.hash(userService.getUserByUsername(userInfo.getUsername()).getPushKey()); //获取推流鉴权密钥
          //示例 https://192.168.126.111:9443/index/api/webrtc?app=live&stream=test&type=play&sign=...
          String sdpIp;
          if (!ObjectUtils.isEmpty(device.getSdpIp())) {
              sdpIp = device.getSdpIp();
          }else {
              sdpIp = mediaServerItem.getSdpIp();
          }
          String webRtcPushUrl = String.format("https://%s:%s/index/api/webrtc?app=%s&stream=%s&type=%s&sign=%s", sdpIp, mediaServerItem.getHttpSSlPort(), app, stream, type,sign);
          result.put("app",app);
          result.put("stream",stream);
          result.put("type",type);
          result.put("sign",sign);
          result.put("webRtcPushUrl", webRtcPushUrl);
          logger.info("获取webrtc推流地址:{}",webRtcPushUrl);
          return WVPResult.success(result);
      }
  ```

  2.下发国标信令接口，只贴出关键点

  - **broadcast**

  ![](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2Fstart1.jpg)

  ![](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2Fsatrt2.jpg)

  - 设备invite 

  ![1685004717697](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2Finvite_startSendRtp.png)

3. 关于对讲的讨论大家可以参考这个issue，这里边是我和作者的讨论

[对讲参考issue](https://github.com/ZLMediaKit/ZLMediaKit/issues/2217)



对讲的必要条件，重要！！！

1. 网站必须得是https，因为浏览器安全机制，如果网站不是https，调用不起来计算机设备，也就是麦克风，如果开启了https，**因为我们大多数都是自签名证书，需要信任证书才可以进入网站，zlmedia的https站点我们也需要信任一下，**要不然我们没办法往zlmedia推流

   - spring boot配置https

     ![](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2Fspringboothttps.jpg)

2. **通道的音频选项需要打开,现在默认设备上线就是打开的**

![](https://image-1318164152.cos.ap-nanjing.myqcloud.com/wvp-pro-talk-redme%2Fchannel1.jpg)

> **说明：如果没有满足这两个需求，对讲按钮是不能点击的，悬浮会有提示**