规范解读
GB28181-2022针对“基于TCP协议的视音频媒体传输”实时点播、历史视频回放与下载中，TCP媒体传输重连机制，做了说明。
修改后的“基于TCP协议的视音频媒体传输要求”如下：
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IETF RFC 4571。
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。在默认情况下,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,协商机制应符合附录G及IETF RFC 4571的定义。
实时视频点播、历史视频回放与下载的TCP媒体传输在建立TCP连接时应支持重连机制。首次TCP连接失败，TCP媒体流传输客户端应间隔一段时间进行重连,重连间隔应不小于l s,重连次数应不小于3次。
代码实现
本文以大牛直播SDK实现的Andorid平台GB28181设备接入模块为例，收到Invite处理如下，其中SetRTPSenderTransportProtocol()设置TCP/UDP传输模式：
ntsOnInvitePlay()处理代码如下：
// Author: daniusdk.com @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); }
收到Ack后：
// Author: daniusdk.com @Override public void ntsOnAckPlay(String deviceId) { handler_.postDelayed(new Runnable() { @Override public void run() { Log.i(TAG,"ntsOnACKPlay, device_id:" +device_id_); if (!isRTSPPublisherRunning && !isPushingRtmp && !isRecording) { InitAndSetConfig(); } libPublisher.SetGB28181RTPSender(publisherHandle, gb28181_rtp_sender_handle_, gb28181_rtp_payload_type_, gb28181_rtp_encoding_name_); int startRet = libPublisher.StartGB28181MediaStream(publisherHandle); if (startRet != 0) { if (!isRTSPPublisherRunning && !isPushingRtmp && !isRecording) { if (publisherHandle != 0) { libPublisher.SmartPublisherClose(publisherHandle); publisherHandle = 0; } } destoryRTPSender(); Log.e(TAG, "Failed to start GB28181 service.."); return; } if (!isRTSPPublisherRunning && !isPushingRtmp && !isRecording) { CheckInitAudioRecorder(); } startLayerPostThread(); isGB28181StreamRunning = true; } private String device_id_; public Runnable set(String device_id) { this.device_id_ = device_id; return this; } }.set(deviceId),0); }
总结
TCP媒体传输重连机制，非常必要，实际上在2022出来之前，我们也已经做了很好的重连处理，GB28181-2022对此专门做了详细的解释说明，具体实现难度不大，感兴趣的开发者可以酌情参考。
