GB28181-2022相对2016版"基于TCP协议的视音频媒体传输要求"调整

时间：2023-02-16 15:02:27

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：

GB28181-2022相对2016版"基于TCP协议的视音频媒体传输要求"调整

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。

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