GB28181基于TCP协议的视音频媒体传输探究及实现

我们先看看官方规范针对TCP协议的视音频传输描述：

实时视频点播、历史视频回放与下载的 TCP媒体传输应支持基于RTP封装的视音频PS流，封装格式参照IETFRFC4571。

流媒体服务器宜同时支持作为TCP媒体流传输服务端和客户端。默认情况下，前端设备向流媒体服务器发送媒体流时前端设备应作为TCP媒体流传输客户端,流媒体服务器作为 TCP媒体流传输服务端;同级或跨级流媒体服务器间基于 TCP协议传输视频流时,媒体流的接收方宜作为TCP媒体流传输服务端。

媒体流的发送方和接收方可扩展SDP参数进行TCP媒体流传输服务端和客户端的协商,协商机制参考附录 F及IETFRFC4571的定义。

这里我们看个INVITE信令交互示例：

INVITE sip:34020000001320000001@3402000000 SIP/2.0
Via: SIP/2.0/TCP 192.168.0.105:15060;rport;branch=z9hG4bK630055772
From: <sip:34020000002000000001@3402000000>;tag=562055772
To: <sip:34020000001320000001@3402000000>
Call-ID: 589055668
CSeq: 183 INVITE
Content-Type: APPLICATION/SDP
Contact: <sip:34020000002000000001@192.168.0.105:15060>
Max-Forwards: 70
User-Agent: LiveGB28181
Subject: 34020000001320000001:0200000001,34020000002000000001:0
Content-Length: 222

v=0
o=34020000001320000001 0 0 IN IP4 192.168.0.105
s=Play
c=IN IP4 192.168.0.105
t=0 0
m=video 30076 RTP/AVP 96 97 98
a=recvonly
a=rtpmap:96 PS/90000
a=rtpmap:97 MPEG4/90000
a=rtpmap:98 H264/90000
y=0200000001

判断媒体流走TCP还是UDP，主要看这里：

m=video 30076 RTP/AVP 96 97 98

传输方式采用“RTP/AVP”标识传输层协议为 RTP over UDP，采用“TCP/RTP/AVP”标识传输层协议为 RTP over TCP，需要注意的是，我们实际对接的时候，部分厂商SDP非常随意，有的甚至直接标记个tcp，这让我们对接的时候，很困惑。

技术实现：

 

本文以大牛直播SDK的Android平台GB28181设备接入端为例，启动GB28181，完成注册、catalog等交互后，Invite上来后，设置媒体流通过TCP还是UDP发送出去：

@Override
public void ntsOnInvitePlay(String deviceId, SessionDescription session_des) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      // 先振铃响应下
      gb28181_agent_.respondPlayInvite(180, device_id_);

      MediaSessionDescription video_des = null;
      SDPRtpMapAttribute ps_rtpmap_attr = null;

      // 28181 视频使用PS打包
      Vector<MediaSessionDescription> video_des_list = session_des_.getVideoPSDescriptions();
      if (video_des_list != null && !video_des_list.isEmpty()) {
        for(MediaSessionDescription m : video_des_list) {
          if (m != null && m.isValidAddressType() && m.isHasAddress() ) {
            video_des = m;
            ps_rtpmap_attr = video_des.getPSRtpMapAttribute();
            break;
          }
        }
      }

      if (null == video_des) {
        gb28181_agent_.respondPlayInvite(488, device_id_);
        Log.i(TAG, "ntsOnInvitePlay get video description is null, response 488, device_id:" + device_id_);
        return;
      }

      if (null == ps_rtpmap_attr) {
        gb28181_agent_.respondPlayInvite(488, device_id_);
        Log.i(TAG, "ntsOnInvitePlay get ps rtp map attribute is null, response 488, device_id:" + device_id_);
        return;
      }

      Log.i(TAG,"ntsOnInvitePlay, device_id:" +device_id_+", is_tcp:" + video_des.isRTPOverTCP()
            + " rtp_port:" + video_des.getPort() + " ssrc:" + video_des.getSSRC()
            + " address_type:" + video_des.getAddressType() + " address:" + video_des.getAddress());

      long rtp_sender_handle = libPublisher.CreateRTPSender(0);
      if ( rtp_sender_handle == 0 ) {
        gb28181_agent_.respondPlayInvite(488, device_id_);
        Log.i(TAG, "ntsOnInvitePlay CreateRTPSender failed, response 488, device_id:" + device_id_);
        return;
      }

      gb28181_rtp_payload_type_  = ps_rtpmap_attr.getPayloadType();
      gb28181_rtp_encoding_name_ =  ps_rtpmap_attr.getEncodingName();

      libPublisher.SetRTPSenderTransportProtocol(rtp_sender_handle, video_des.isRTPOverUDP()?0:1);
      libPublisher.SetRTPSenderIPAddressType(rtp_sender_handle, video_des.isIPv4()?0:1);
      libPublisher.SetRTPSenderLocalPort(rtp_sender_handle, 0);
      libPublisher.SetRTPSenderSSRC(rtp_sender_handle, video_des.getSSRC());
      libPublisher.SetRTPSenderSocketSendBuffer(rtp_sender_handle, 2*1024*1024); // 设置到2M
      libPublisher.SetRTPSenderClockRate(rtp_sender_handle, ps_rtpmap_attr.getClockRate());
      libPublisher.SetRTPSenderDestination(rtp_sender_handle, video_des.getAddress(), video_des.getPort());

      if ( libPublisher.InitRTPSender(rtp_sender_handle) != 0 ) {
        gb28181_agent_.respondPlayInvite(488, device_id_);
        libPublisher.DestoryRTPSender(rtp_sender_handle);
        return;
      }

      int local_port = libPublisher.GetRTPSenderLocalPort(rtp_sender_handle);
      if (local_port == 0) {
        gb28181_agent_.respondPlayInvite(488, device_id_);
        libPublisher.DestoryRTPSender(rtp_sender_handle);
        return;
      }

      Log.i(TAG,"get local_port:" + local_port);

      String local_ip_addr = IPAddrUtils.getIpAddress(context_);

      MediaSessionDescription local_video_des = new MediaSessionDescription(video_des.getType());

      local_video_des.addFormat(String.valueOf(ps_rtpmap_attr.getPayloadType()));
      local_video_des.addRtpMapAttribute(ps_rtpmap_attr);

      local_video_des.setAddressType(video_des.getAddressType());
      local_video_des.setAddress(local_ip_addr);
      local_video_des.setPort(local_port);

      local_video_des.setTransportProtocol(video_des.getTransportProtocol());
      local_video_des.setSSRC(video_des.getSSRC());

      if (!gb28181_agent_.respondPlayInviteOK(device_id_,local_video_des) ) {
        libPublisher.DestoryRTPSender(rtp_sender_handle);
        Log.e(TAG, "ntsOnInvitePlay call respondPlayInviteOK failed.");
        return;
      }

      gb28181_rtp_sender_handle_ = rtp_sender_handle;
    }

    private String device_id_;
    private SessionDescription session_des_;

    public Runnable set(String device_id, SessionDescription session_des) {
      this.device_id_ = device_id;
      this.session_des_ = session_des;
      return this;
    }
  }.set(deviceId, session_des),0);
}

接口设计如下：

/**
   *设置 RTP Sender传输协议
   *
   * @param rtp_sender_handle, CreateRTPSender返回值
   * @param transport_protocol, 0:UDP, 1:TCP, 默认是UDP
   *
   * @return {0} if successful
   */
  public native int SetRTPSenderTransportProtocol(long rtp_sender_handle, int transport_protocol);

以上是GB28181基于TCP协议的视音频媒体传输探究及实现，感兴趣的开发者，可以查看相关协议规范，根据需求实现自己的业务逻辑即可。