/*完成ffmpeg推流到zlmedia流媒体服务器功能。*/

src/main/java/com/inspect/tcpserver/sip/media/Gb28181RtpStreamer.java

@@ -0,0 +1,83 @@
+package com.inspect.tcpserver.sip.media;
+
+import lombok.extern.slf4j.Slf4j;
+import org.bytedeco.javacv.*;
+import org.springframework.stereotype.Service;
+
+@Slf4j
+@Service
+public class Gb28181RtpStreamer {
+
+    private volatile boolean running = false;
+    private Thread worker;
+
+    public void start(String rtspUrl, String zlmIp, int zlmPort) {
+
+        if (running) return;
+        running = true;
+
+        worker = new Thread(() -> pushLoop(rtspUrl, zlmIp, zlmPort));
+        worker.start();
+    }
+
+    public void stop() {
+        running = false;
+    }
+
+    private void pushLoop(String rtspUrl, String ip, int port) {
+
+        while (running) {
+            try {
+                push(rtspUrl, ip, port);
+            } catch (Exception e) {
+                log.error("推流断开，5秒重连", e);
+                sleep(5000);
+            }
+        }
+    }
+
+    private void push(String rtspUrl, String ip, int port) throws Exception {
+
+        String outputUrl = "rtp://" + ip + ":" + port +
+                "?pkt_size=1400&localrtpport=50000";
+
+        log.info("开始推流 -> {}", outputUrl);
+
+        // RTSP
+        FFmpegFrameGrabber grabber = new FFmpegFrameGrabber(rtspUrl);
+        grabber.setOption("rtsp_transport", "tcp");
+        grabber.setOption("stimeout", "5000000");
+        grabber.start();
+
+        // 推RTP(MPEGTS)
+        FFmpegFrameRecorder recorder =
+                new FFmpegFrameRecorder(outputUrl,
+                        grabber.getImageWidth(),
+                        grabber.getImageHeight());
+
+        // 关键配置（等价ffmpeg命令）
+        recorder.setFormat("rtp_mpegts");     // ⭐核心
+        recorder.setVideoCodec(grabber.getVideoCodec());
+        recorder.setFrameRate(grabber.getFrameRate());
+        recorder.setVideoBitrate(grabber.getVideoBitrate());
+
+        // AnnexB + SPS PPS
+        recorder.setVideoOption("bsf:v", "h264_mp4toannexb");
+        recorder.setOption("mpegts_flags", "resend_headers");
+
+        recorder.start();
+
+        Frame frame;
+        while (running && (frame = grabber.grab()) != null) {
+            recorder.record(frame);
+        }
+
+        recorder.stop();
+        grabber.stop();
+    }
+
+    private void sleep(long ms) {
+        try { Thread.sleep(ms); } catch (Exception ignored) {}
+    }
+}
+

src/main/java/com/inspect/tcpserver/sip/media/Gb28181StreamService.java

@@ -2,9 +2,7 @@ package com.inspect.tcpserver.sip.media;
 
 import lombok.extern.slf4j.Slf4j;
 import org.bytedeco.ffmpeg.global.avcodec;
-import org.bytedeco.javacv.FFmpegFrameGrabber;
-import org.bytedeco.javacv.FFmpegFrameRecorder;
-import org.bytedeco.javacv.Frame;
+import org.bytedeco.javacv.*;
 import org.springframework.stereotype.Service;
 
 @Slf4j
@@ -14,9 +12,79 @@ public class Gb28181StreamService {
     private Thread pushThread;
     private volatile boolean isPushing = false;
 
+    private volatile boolean running = false;
+
     private final String rtspInputUrl = "rtsp://admin:wd19216811@192.168.1.244:554/h264/ch1/sub/av_stream";
     private final String rtpOutputUrl= "rtp://192.168.1.116:30000";
 
+    public void startPush() {
+        if (running) {
+            log.warn("推流已在运行");
+            return;
+        }
+
+        running = true;
+        pushThread = new Thread(() -> {
+            FFmpegFrameGrabber grabber = null;
+            FFmpegFrameRecorder recorder = null;
+
+            try {
+                // ================= 第一步：拉取 RTSP 源 =================
+                grabber = new FFmpegFrameGrabber("rtsp://admin:wd19216811@192.168.1.244:554/h264/ch1/sub/av_stream");
+                grabber.setOption("rtsp_transport", "tcp");           // 强制 TCP 拉流
+                grabber.setOption("fflags", "nobuffer");              // 降低延迟
+                grabber.setOption("flags", "low_delay");
+                grabber.setOption("analyzeduration", "2000000");      // 分析时长
+                grabber.setOption("probesize", "2000000");
+                grabber.start();
+
+                // ================= 第二步：配置 RTP_MPEGTS 输出 =================
+                String outputUrl = "rtp://192.168.1.116:30000?pkt_size=1400&localrtpport=50000";
+
+                recorder = new FFmpegFrameRecorder(outputUrl, grabber.getImageWidth(), grabber.getImageHeight(), 0); // 0 = 无音频通道
+                recorder.setFormat("rtp_mpegts");                     // 核心：rtp_mpegts 封装
+                recorder.setVideoCodec(avcodec.AV_CODEC_ID_H264);     // H264
+                recorder.setVideoCodecName("copy");                   // -c:v copy（尽量不重新编码）
+                recorder.setVideoBitrate(grabber.getVideoBitrate());  // 继承源码率
+                recorder.setFrameRate(grabber.getFrameRate());        // 继承帧率
+
+                // 模拟 -bsf:v h264_mp4toannexb（JavaCV 会自动处理 Annex-B 输出）
+                recorder.setVideoOption("bsf:v", "h264_mp4toannexb");
+
+                // 模拟 -mpegts_flags resend_headers（重复发送 TS 头）
+                //recorder.setFormatOption("mpegts_flags", "resend_headers");
+                recorder.setFormat("rtp_mpegts");
+                recorder.setVideoCodecName("copy"); // -c:v copy
+
+
+                // pkt_size 和 localrtpport 通过输出 URL 参数已设置
+
+                recorder.start();
+
+                log.info("GB28181 PS over RTP 推流启动：{} -> {}", grabber.getFormat(), outputUrl);
+
+                Frame frame;
+                while (running && (frame = grabber.grabFrame()) != null) {
+                    recorder.record(frame);
+                }
+
+            } catch (FrameGrabber.Exception | FrameRecorder.Exception e) {
+                log.error("推流异常", e);
+            } finally {
+                try {
+                    if (recorder != null) recorder.stop();
+                    if (grabber != null) grabber.stop();
+                } catch (Exception e) {
+                    log.error("关闭资源异常", e);
+                }
+                running = false;
+                log.info("推流线程结束");
+            }
+        }, "rtp-push-thread");
+
+        pushThread.start();
+    }
+
     public void startPushStream() {
         if (isPushing) {
             log.warn("Push is already running.");

src/main/java/com/inspect/tcpserver/sip/service/SipClientService.java

@@ -1,5 +1,6 @@
 package com.inspect.tcpserver.sip.service;
 
+import com.inspect.tcpserver.sip.media.Gb28181RtpStreamer;
 import com.inspect.tcpserver.sip.media.Gb28181StreamService;
 import com.inspect.tcpserver.sip.media.InviteSdpInfo;
 import com.inspect.tcpserver.sip.registry.SipEventRegistry;
@@ -98,6 +99,9 @@ public class SipClientService implements SipListener {
     @Autowired
     Gb28181StreamService streamService;
 
+    @Autowired
+    Gb28181RtpStreamer gb28181RtpStreamer;
+
     private static final String[] IP_SERVICES = {
             "https://api.ipify.org",
             "https://checkip.amazonaws.com",
@@ -855,6 +859,8 @@ public class SipClientService implements SipListener {
                 }
 
                 streamService.startPushStream(rtspInputUrl, rtpStreamHost, rtpStreamPort);
+                //streamService.startPush();//PS OVER RTP
+                //gb28181RtpStreamer.start(rtspInputUrl, rtpStreamHost, rtpStreamPort);
             }
             else {
                 Response notImpl = messageFactory.createResponse(Response.NOT_IMPLEMENTED, request);
@@ -902,6 +908,7 @@ public class SipClientService implements SipListener {
         Response ok = messageFactory.createResponse(Response.OK, request);
 
         String localSipId = "29001002120107000001";
+        // 这个端口几乎没有任何作用，可以改成任何值，只是为了协议的完整性，因为这里的TCP是设备主动连接平台的，不需要平台通过这个端口反过来连接设备
         int localRtpPort = 554;
         // Contact 必须
         Address contactAddress = addressFactory.createAddress(