如何在Android平台GB28181接入终端实现语音广播和语音对讲

技术背景

在之前的blog，我们以Android平台国标接入终端为例，分别介绍了一些常规的功能，比如REGISTER、CATALOG、INVITE、Keepalive、SUBSCRIBE、NOTIFY等常规操作，今天主要介绍下语音广播和语音对讲这部分。

GB28181平台广播和对讲这块，重要性不言而喻，没有广播的接入终端，数据只是单向流入，加入后，指挥中心和终端之间的联系更紧密，实时双向沟通更方便，适用的行业范围也更广泛。

相关SPEC解读

关于语音广播和对讲，感兴趣的开发者可直接参阅GBT 28181-2016.pdf相关技术规范里面的9.12章节，以下是部分精选介绍：

命令交互流程

命令描述流程

a) 1:SIP服务器向语音流接收者发送语音广播通知消息,消息中通过 To头域标明作为目的地址 的语音流接收者ID,消息采用 Message方法携带。

举例说明：

代码语言：javascript复制

MESSAGE sip:34020000001380000001@192.168.2.212:12070 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.154:15060;rport;branch=z9hG4bK311226558
From: <sip:34020000002000000001@3402000000>;tag=280226558
To: <sip:34020000001380000001@192.168.2.212:12070>
Call-ID: 172226558
CSeq: 207 MESSAGE
Content-Type: Application/MANSCDP xml
Max-Forwards: 70
Content-Length: 206

<?xml version="1.0" encoding="GB2312"?>
<Notify>
  <CmdType>Broadcast</CmdType>
  <SN>461226558</SN>
  <SourceID>34020000002000000001</SourceID>
  <TargetID>34020000001380000001</TargetID>
</Notify>

b) 2:语音流接收者收到语音广播通知消息后,向SIP服务器发送200OK 响应。

代码语言：javascript复制

SIP/2.0 200 OK
CSeq: 207 MESSAGE
Call-ID: 172226558
From: <sip:34020000002000000001@3402000000>;tag=280226558
To: <sip:34020000001380000001@192.168.2.212:12070>
Via: SIP/2.0/UDP 192.168.2.154:15060;rport=15060;branch=z9hG4bK311226558;received=192.168.2.154
Content-Length: 0

c) 3:语音流接收者向SIP服务器发送语音广播应答消息,消息中通过 To头域标明作为目的地 址的SIP服务器ID,消息采用 Message方法携带。

代码语言：javascript复制

MESSAGE sip:34020000002000000001@3402000000 SIP/2.0
Call-ID: 0fc1f2c83c28898a29e146d7ef581908@192.168.2.212
CSeq: 337044229 MESSAGE
From: <sip:34020000001380000001@3402000000>;tag=93882333
To: <sip:34020000002000000001@3402000000>
Via: SIP/2.0/UDP 192.168.2.212:12070;rport;branch=z9hG4bK-363733-79fd88c45667975e5ebaf18f84b91a8e
Max-Forwards: 70
User-Agent: NT GB28181 User Agent V1.2(daniusdk.com)
Content-Type: Application/MANSCDP xml
Content-Length: 180

<?xml version="1.0" encoding="GB2312"?>
<Response>
<CmdType>Broadcast</CmdType>
<SN>461226558</SN>
<DeviceID>34020000001380000001</DeviceID>
<Result>OK</Result>
</Response>

d) 4:SIP服务器收到语音广播应答消息后,向语音流接收者发送200OK 响应。

代码语言：javascript复制

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.212:12070;rport=12070;received=192.168.2.212;branch=z9hG4bK-363733-79fd88c45667975e5ebaf18f84b91a8e
From: <sip:34020000001380000001@3402000000>;tag=93882333
To: <sip:34020000002000000001@3402000000>;tag=355226593
CSeq: 337044229 MESSAGE
Call-ID: 0fc1f2c83c28898a29e146d7ef581908@192.168.2.212
Content-Length: 0

e) 5:语音流接收者向SIP服务器发送Invite消息,消息中通过 To头域标明作为目的地址的语音 流发送者ID,消息头域中携带Subject字段,表明请求的语音流发送者ID、发送方媒体流序列 号、语音流接收者ID、接收方媒体流序列号等参数,SDP消息体中s字段为“Play”代表实时点 播,m 字段中媒体参数标识为“audio”表示请求语音媒体流。

代码语言：javascript复制

INVITE sip:34020000002000000001@3402000000 SIP/2.0
Call-ID: 2b4f0f0512aa1a49ffc645618d0e8bae@192.168.2.212
CSeq: 44264 INVITE
From: <sip:34020000001380000001@3402000000>;tag=32ecf22a
To: <sip:34020000002000000001@3402000000>
Via: SIP/2.0/UDP 192.168.2.212:12070;rport;branch=z9hG4bK-363733-15283c8a0ea0a1e9dbf295ce2359dbe7
Max-Forwards: 70
Contact: <sip:34020000001380000001@192.168.2.212:12070>
Subject: 34020000002000000001:0200006727,34020000001380000001:0
User-Agent: NT GB28181 User Agent V1.2(daniusdk.com)
Content-Type: APPLICATION/SDP
Content-Length: 221

v=0
o=34020000002000000001 0 0 IN IP4 192.168.2.212
s=Play
c=IN IP4 192.168.2.212
t=0 0
m=audio 25002 TCP/RTP/AVP 8
a=setup:active
a=connection:new
a=recvonly
a=rtpmap:8 PCMA/8000
y=0200006727
f=v/////a/1/8/1

f) 6:SIP服务器收到Invite请求后,通过三方呼叫控制建立媒体服务器和语音流发送者之间的媒体连接。向媒体服务器发送Invite消息,此消息不携带SDP消息体。

g) 7:媒体服务器收到SIP服务器的Invite请求后,回复200OK 响应,携带SDP消息体,消息体 中描述了媒体服务器接收媒体流的IP、端口、媒体格式等内容。

h) 8:SIP服务器收到媒体服务器返回的200OK 响应后,向语音流发送者发送Invite请求,消息 中通过 To头域标明作为目的地址的语音流发送者ID,消息头域中携带 Subject字段,表明请 求的语音流发送者ID、发送方媒体流序列号、语音流接收者ID、接收方媒体流序列号等参数, 请求中携带消息7中媒体服务器回复的200OK 响应消息体,s字段为“Play”代表实时点播, m 字段中媒体参数标识为“audio”表示请求语音媒体流,增加y字段描述SSRC值,f字段描述 媒体参数。

代码语言：javascript复制

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.212:12070;rport=12070;received=192.168.2.212;branch=z9hG4bK-363733-15283c8a0ea0a1e9dbf295ce2359dbe7
From: <sip:34020000001380000001@3402000000>;tag=32ecf22a
To: <sip:34020000002000000001@3402000000>;tag=954226632
CSeq: 44264 INVITE
Call-ID: 2b4f0f0512aa1a49ffc645618d0e8bae@192.168.2.212
Contact: <sip:34020000002000000001@192.168.2.154:15060>
Content-Length: 222
Content-Type: APPLICATION/SDP

v=0
o=34020000002000000001 0 0 IN IP4 192.168.2.154
s=Play
c=IN IP4 192.168.2.154
t=0 0
m=audio 30005 TCP/RTP/AVP 8
a=sendonly
a=rtpmap:8 PCMA/8000
a=setup:passive
a=connection:new
y=0200006727
f=v/////a/1/8/1

i) 9:语音流发送者收到SIP服务器的Invite请求后,回复200OK 响应,携带SDP消息体,消息 体中描述了媒体流发送者发送媒体流的IP、端口、媒体格式、SSRC 字段等内容,s字段为 “Play”代表实时点播,m 字段中媒体参数标识为“audio”表示请求语音媒体流。

j) 10:SIP服务器收到语音流发送者返回的200OK 响应后,向媒体服务器发送 ACK 请求,请求 中携带消息9中语音流发送者回复的200OK 响应消息体,完成与媒体服务器的Invite会话 建立过程。

k) 11:SIP服务器收到语音流发送者返回的200OK 响应后,向语音流发送者发送 ACK 请求,请 求中不携带消息体,完成与语音流发送者的Invite会话建立过程。

l) 12:完成三方呼叫控制后,SIP服务器通过 B2BUA 代理方式建立语音流接收者和媒体服务器 之间的媒体连接。在消息5中增加SSRC值,转发给媒体服务器。

m)13:媒体服务器收到Invite请求,回复200OK 响应,携带SDP消息体,消息体中描述了媒体服 务器发送媒体流的IP、端口、媒体格式、SSRC值等内容,s字段为“Play”代表实时点播,m 字段 中媒体参数标识为“audio”表示请求语音媒体流。

n) 14:SIP服务器将消息13转发给语音流接收者。

o) 15:语音流接收者收到200OK 响应后,回复 ACK 消息,完成与SIP服务器的Invite会话建立 过程。

p) 16:SIP服务器将消息15转发给媒体服务器,完成与媒体服务器的Invite会话建立过程。

q) 17:SIP服务器向语音流接收者发送 BYE消息,断开消息5、14、15建立的Invite会话。

r) 18:语音流接收者收到 BYE消息后回复200OK 响应,会话断开。

s) 19:SIP服务器向媒体服务器发送 BYE 消息,断开消息 12、13、16 建立的同媒体服务器的 Invite会话。

t) 20:媒体服务器收到 BYE消息后回复200OK 响应,会话断开。

u) 21:SIP服务器向媒体服务器发送 BYE消息,断开消息6、7、10建立的同媒体服务器的Invite 会话。

v) 22:媒体服务器收到 BYE消息后回复200OK 响应,会话断开。

w)23:SIP服务器向语音流发送者发送 BYE 消息,断开消息8、9、11建立的同语音流发送者的 Invite会话。

x) 24:语音流发送者收到 BYE消息后回复200OK 响应,会话断开。

注：语音广播通知消息除上述流程中通过SIP服务器发出外,也可由语音流发送者发出,消息中通过 To头域标明 作为目的地址的语音流接收者ID,经SIP服务器中转后发往语音流接收者;语音流接收者处理后发送应答消 息,消息中通过 To头域标明作为目的地址的语音流发送者ID,经SIP服务器中转后回复给语音流发送者。后续呼叫流程与上述流程相同。

语音对讲

语音对讲功能实现中心用户与前端用户之间的一对一语音对讲功能。 语音对讲功能由下述两个独立的流程组合实现:

a) 通过9.2的实时视音频点播功能,中心用户获得前端设备的实时视音频媒体流;

b) 通过9.12的语音广播功能,中心用户向前端对讲设备发送实时音频媒体流,语音流的封装格 式见 C.2.4音频流的 RTP封装定义。

C.2.4 音频流的 RTP封装

语音比特流宜采用标准的 RTP协议进行打包，这里只摘录G.711A律的：

在一个 RTP包中,音频载荷数据应为整数个音频编码帧,且时间长度在20ms~180ms之间。

音频载荷数据的 RTP封装参数如下:

a) G.711的主要参数 G.711A律语音编码 RTP包的负载类型(PayloadType)的参数规定如下(见IETFRFC3551— 2003中的表4): 1）负载类型(PT):8; 2) 编码名称(encodingname):PCMA; 3) 时钟频率(clockrate):8kHz; 4) 通道数:1; 5) SDP描述中“m”字段的“media”项:audio。

技术实现

语音广播接收这块，由于有之前的RTMP和RTSP播放器积累，直接在player端做相应扩展即可，当收到广播后，GB28181语音广播按钮使能。

相关接口设计如下：

代码语言：javascript复制

/*                  RTP Receiver                      */
  //GitHub: https://github.com/daniulive/SmarterStreaming
  //WebSite: https://daniusdk.com
  /*
   * 创建RTP Receiver
   *
   * @param reserve：保留参数传0
   *
   * @return RTP Receiver 句柄，0表示失败
   */
  public native long CreateRTPReceiver(int reserve);


  /**
   *设置 RTP Receiver传输协议
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   * @param transport_protocol, 0:UDP, 1:TCP, 默认是UDP
   *
   * @return {0} if successful
   */
  public native int SetRTPReceiverTransportProtocol(long rtp_receiver_handle, int transport_protocol);


  /**
   *设置 RTP Receiver IP地址类型
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   * @param ip_address_type, 0:IPV4, 1:IPV6, 默认是IPV4
   *
   * @return {0} if successful
   */
  public native int SetRTPReceiverIPAddressType(long rtp_receiver_handle, int ip_address_type);


  /**
   *设置 RTP Receiver RTP Socket本地端口
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   * @param port, 必须是偶数，设置0的话SDK会自动分配, 默认值是0
   *
   * @return {0} if successful
   */
  public native int SetRTPReceiverLocalPort(long rtp_receiver_handle, int port);


  /**
   *设置 RTP Receiver SSRC
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   * @param ssrc, 如果设置的话，这个字符串要能转换成uint32类型, 否则设置失败
   *
   * @return {0} if successful
   */
  public native int SetRTPReceiverSSRC(long rtp_receiver_handle, String ssrc);


  /**
   *创建 RTP Receiver 会话
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   * @param reserve, 保留值，目前传0
   *
   * @return {0} if successful
   */
  public native int CreateRTPReceiverSession(long rtp_receiver_handle, int reserve);


  /**
   *获取 RTP Receiver RTP Socket本地端口
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   *
   * @return 失败返回0, 成功的话返回响应的端口, 请在CreateRTPReceiverSession返回成功之后调用
   */
  public native int GetRTPReceiverLocalPort(long rtp_receiver_handle);


  /**
   *设置 RTP Receiver Payload 相关信息
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   *
   * @param payload_type, 请参考 RFC 3551
   *
   * @param encoding_name, 编码名, 请参考 RFC 3551, 如果payload_type不是动态的, 可能传null就好
   *
   * @param media_type, 媒体类型, 请参考 RFC 3551, 1 是视频, 2是音频
   *
   * @param clock_rate, 请参考 RFC 3551
   *
   * @return {0} if successful
   */
  public native int SetRTPReceiverPayloadType(long rtp_receiver_handle, int payload_type, String encoding_name, int media_type, int clock_rate);


  /**
   *设置 RTP Receiver 音频采样率
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   * @param sampling_rate, 音频采样率
   *
   * @return {0} if successful
   */
  public native int SetRTPReceiverAudioSamplingRate(long rtp_receiver_handle, int sampling_rate);

  /**
   *设置 RTP Receiver 音频通道数
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   * @param channels, 音频通道数
   *
   * @return {0} if successful
   */
  public native int SetRTPReceiverAudioChannels(long rtp_receiver_handle, int channels);


  /**
   *设置 RTP Receiver 远端地址
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   * @param address, IP地址
   * @param port, 端口
   *
   * @return {0} if successful
   */
  public native int SetRTPReceiverRemoteAddress(long rtp_receiver_handle, String address, int port);

  /**
   *初始化 RTP Receiver
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   *
   * @return {0} if successful
   */
  public native int InitRTPReceiver(long rtp_receiver_handle);

  /**
   *UnInit RTP Receiver
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   *
   * @return {0} if successful
   */
  public native int UnInitRTPReceiver(long rtp_receiver_handle);


  /**
   *Destory RTP Receiver Session
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   *
   * @return {0} if successful
   */
  public native int DestoryRTPReceiverSession(long rtp_receiver_handle);


  /**
   *Destory RTP Receiver
   *
   * @param rtp_receiver_handle, CreateRTPReceiver
   *
   * @return {0} if successful
   */
  public native int DestoryRTPReceiver(long rtp_receiver_handle);


  /*                  RTP Receiver                      */

相关调用代码：

代码语言：javascript复制

class ButtonGB28181AudioBroadcastListener implements OnClickListener {
        public void onClick(View v) {
            if (gb_broadcast_source_id_ != null && gb_broadcast_target_id_ != null) {
                if (gb28181_agent_ != null ) {
                    if (gb28181_agent_.byeAudioBroadcast(gb_broadcast_source_id_, gb_broadcast_target_id_) ) {
                        gb_broadcast_source_id_ = null;
                        gb_broadcast_target_id_ = null;
                        btnGB28181AudioBroadcast.setText("GB28181语音广播");
                        btnGB28181AudioBroadcast.setEnabled(false);
                    }
                }
            }

            stopAudioPlayer();
            destoryRTPReceiver();
        }
    }

代码语言：javascript复制

@Override
public void ntsOnNotifyBroadcastCommand(String fromUserName, String fromUserNameAtDomain, String sn, String sourceID, String targetID) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnNotifyBroadcastCommand, fromUserName:"  from_user_name_   ", fromUserNameAtDomain:"  from_user_name_at_domain_
              ", SN:"   sn_   ", sourceID:"   source_id_   ", targetID:"   target_id_);

      if (gb28181_agent_ != null ) {
        gb28181_agent_.respondBroadcastCommand(from_user_name_, from_user_name_at_domain_,sn_,source_id_, target_id_, true);
        btnGB28181AudioBroadcast.setText("收到GB28181语音广播通知");
      }
    }

    private String from_user_name_;
    private String from_user_name_at_domain_;
    private String sn_;
    private String source_id_;
    private String target_id_;

    public Runnable set(String from_user_name, String from_user_name_at_domain, String sn, String source_id, String target_id) {
      this.from_user_name_ = from_user_name;
      this.from_user_name_at_domain_ = from_user_name_at_domain;
      this.sn_ = sn;
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(fromUserName, fromUserNameAtDomain, sn, sourceID, targetID),0);
}

@Override
public void ntsOnAudioBroadcast(String commandFromUserName, String commandFromUserNameAtDomain, String sourceID, String targetID) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnAudioBroadcastPlay, fromFromUserName:"   command_from_user_name_
              " FromUserNameAtDomain:"   command_from_user_name_at_domain_
              " sourceID:"   source_id_   ", targetID:"   target_id_);

      stopAudioPlayer();
      destoryRTPReceiver();

      if (gb28181_agent_ != null ) {
        String local_ip_addr = IPAddrUtils.getIpAddress(context_);

        boolean is_tcp = true; // 考虑到跨网段, 默认用TCP传输rtp包
        rtp_receiver_handle_ = lib_player_.CreateRTPReceiver(0);
        if (rtp_receiver_handle_ != 0 ) {
          lib_player_.SetRTPReceiverTransportProtocol(rtp_receiver_handle_, is_tcp?1:0);
          lib_player_.SetRTPReceiverIPAddressType(rtp_receiver_handle_, 0);

          if (0 == lib_player_.CreateRTPReceiverSession(rtp_receiver_handle_, 0) ) {
            int local_port = lib_player_.GetRTPReceiverLocalPort(rtp_receiver_handle_);
            boolean ret = gb28181_agent_.inviteAudioBroadcast(command_from_user_name_,command_from_user_name_at_domain_,
                                                              source_id_, target_id_, "IP4", local_ip_addr, local_port, is_tcp?"TCP/RTP/AVP":"RTP/AVP");

            if (!ret ) {
              destoryRTPReceiver();
              btnGB28181AudioBroadcast.setText("GB28181语音广播");
            }
            else {
              btnGB28181AudioBroadcast.setText("GB28181语音广播呼叫中");
            }
          } else {
            destoryRTPReceiver();
            btnGB28181AudioBroadcast.setText("GB28181语音广播");
          }
        }
      }
    }

    private String command_from_user_name_;
    private String command_from_user_name_at_domain_;
    private String source_id_;
    private String target_id_;

    public Runnable set(String command_from_user_name, String command_from_user_name_at_domain, String source_id, String target_id) {
      this.command_from_user_name_ = command_from_user_name;
      this.command_from_user_name_at_domain_ = command_from_user_name_at_domain;
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(commandFromUserName, commandFromUserNameAtDomain, sourceID, targetID),0);
}

@Override
public void ntsOnInviteAudioBroadcastException(String sourceID, String targetID, String errorInfo) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnInviteAudioBroadcastException, sourceID:"   source_id_   ", targetID:"   target_id_);

      destoryRTPReceiver();
      btnGB28181AudioBroadcast.setText("GB28181语音广播");
    }

    private String source_id_;
    private String target_id_;

    public Runnable set(String source_id, String target_id) {
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(sourceID, targetID),0);
}

@Override
public void ntsOnInviteAudioBroadcastTimeout(String sourceID, String targetID) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnInviteAudioBroadcastTimeout, sourceID:"   source_id_   ", targetID:"   target_id_);

      destoryRTPReceiver();
      btnGB28181AudioBroadcast.setText("GB28181语音广播");
    }

    private String source_id_;
    private String target_id_;

    public Runnable set(String source_id, String target_id) {
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(sourceID, targetID),0);
}

代码语言：javascript复制

@Override
public void ntsOnInviteAudioBroadcastResponse(String sourceID, String targetID, int statusCode, PlaySessionDescription sessionDescription) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnInviteAudioBroadcastResponse, statusCode:"   status_code_  " sourceID:"   source_id_   ", targetID:"   target_id_);

      boolean is_need_destory_rtp = true;

      if (gb28181_agent_ != null ) {
        boolean is_need_bye = 200==status_code_;

        if (200 == status_code_ && session_description_ != null && rtp_receiver_handle_ != 0 ) {
          MediaSessionDescription audio_des = session_description_.getAudioDescription();

          SDPRtpMapAttribute audio_attr = null;
          if (audio_des != null && audio_des.getRtpMapAttributes() != null && !audio_des.getRtpMapAttributes().isEmpty() )
            audio_attr = audio_des.getRtpMapAttributes().get(0);

          if ( audio_des != null && audio_attr != null ) {
            lib_player_.SetRTPReceiverSSRC(rtp_receiver_handle_, audio_des.getSSRC());

            lib_player_.SetRTPReceiverPayloadType(rtp_receiver_handle_, audio_attr.getPayloadType(),
                                                  audio_attr.getEncodingName(), 2, audio_attr.getClockRate());

            lib_player_.SetRTPReceiverRemoteAddress(rtp_receiver_handle_, audio_des.getAddress(), audio_des.getPort());
            lib_player_.InitRTPReceiver(rtp_receiver_handle_);

            if (startAudioPlay()) {
              is_need_bye = false;
              is_need_destory_rtp = false;

              gb_broadcast_source_id_ = source_id_;
              gb_broadcast_target_id_ = target_id_;
              btnGB28181AudioBroadcast.setText("终止GB28181语音广播");
              btnGB28181AudioBroadcast.setEnabled(true);
            }
          }

        } else {
          btnGB28181AudioBroadcast.setText("GB28181语音广播");
        }

        if (is_need_bye)
          gb28181_agent_.byeAudioBroadcast(source_id_, target_id_);
      }

      if (is_need_destory_rtp)
        destoryRTPReceiver();
    }

    private String source_id_;
    private String target_id_;
    private int status_code_;
    private PlaySessionDescription session_description_;

    public Runnable set(String source_id, String target_id, int status_code, PlaySessionDescription session_description) {
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      this.status_code_ = status_code;
      this.session_description_ = session_description;
      return this;
    }

  }.set(sourceID, targetID, statusCode, sessionDescription),0);
}

@Override
public void ntsOnByeAudioBroadcast(String sourceID, String targetID) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnByeAudioBroadcast sourceID:"   source_id_   " targetID:"   target_id_);

      gb_broadcast_source_id_ = null;
      gb_broadcast_target_id_ = null;
      btnGB28181AudioBroadcast.setText("GB28181语音广播");
      btnGB28181AudioBroadcast.setEnabled(false);

      stopAudioPlayer();
      destoryRTPReceiver();
    }

    private String source_id_;
    private String target_id_;

    public Runnable set(String source_id, String target_id) {
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(sourceID, targetID),0);
}

@Override
public void ntsOnTerminateAudioBroadcast(String sourceID, String targetID) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnTerminateAudioBroadcast sourceID:"   source_id_   " targetID:"   target_id_);

      gb_broadcast_source_id_ = null;
      gb_broadcast_target_id_ = null;
      btnGB28181AudioBroadcast.setText("GB28181语音广播");
      btnGB28181AudioBroadcast.setEnabled(false);

      stopAudioPlayer();
      destoryRTPReceiver();
    }

    private String source_id_;
    private String target_id_;

    public Runnable set(String source_id, String target_id) {
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(sourceID, targetID),0);
}

总结

至此、Android平台GB28181接入终端，如位置订阅、语音广播和语音对讲这块已经全面覆盖，加上之前的技术积累，看了下，已覆盖了以下部分：

• ​[视频格式]H.264/H.265(Android H.265硬编码)；
• [音频格式]G.711 A律、AAC；
• [音量调节]Android平台采集端支持实时音量调节；
• [H.264硬编码]支持H.264特定机型硬编码；
• [H.265硬编码]支持H.265特定机型硬编码；
• [软硬编码参数配置]支持gop间隔、帧率、bit-rate设置；
• [软编码参数配置]支持软编码profile、软编码速度、可变码率设置；
• 支持纯视频、音视频PS打包传输；
• 支持RTP OVER UDP和RTP OVER TCP被动模式；
• 支持信令通道网络传输协议TCP/UDP设置；
• 支持注册、注销，支持注册刷新及注册有效期设置；
• 支持设备目录查询应答；
• 支持心跳机制，支持心跳间隔、心跳检测次数设置；
• 支持移动设备位置(MobilePosition)订阅和通知；
• 支持国标GB/T28181—2016平台接入；
• 支持语音广播及语音对讲；
• [实时水印]支持动态文字水印、png水印；
• [实时静音]支持实时静音/取消静音；
• [实时快照]支持实时快照；
• [降噪]支持环境音、手机干扰等引起的噪音降噪处理、自动增益、VAD检测。​

特别是语音广播和语音对讲这块，是GB28181终端接入模块的一个核心扩展功能，在智能门禁、工业与物联网、监控等行业，用途非常广泛，技术实现这块，不要忽略的技术点还有降噪和回音消除这块，由于之前我们有技术积累，可以直接复用，对新入手的开发者来说，也提出了新的挑战，感兴趣的开发者，可以酌情参考。

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