优化rtcp判断

648540858

2022-11-21 6b03568c5dd128a3d71c02fb1a3a76a4344a4920

 src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/InviteRequestProcessor.java

@@ -4,19 +4,14 @@
import com.genersoft.iot.vmp.conf.DynamicTask;
import com.genersoft.iot.vmp.conf.SipConfig;
import com.genersoft.iot.vmp.conf.UserSetting;
import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException;
import com.genersoft.iot.vmp.gb28181.bean.*;
import com.genersoft.iot.vmp.gb28181.event.SipSubscribe;
import com.genersoft.iot.vmp.gb28181.session.AudioBroadcastManager;
import com.genersoft.iot.vmp.gb28181.session.VideoStreamSessionManager;
import com.genersoft.iot.vmp.gb28181.transmit.SIPProcessorObserver;
import com.genersoft.iot.vmp.gb28181.transmit.callback.DeferredResultHolder;
import com.genersoft.iot.vmp.gb28181.transmit.callback.RequestMessage;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommander;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommanderForPlatform;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.impl.SIPCommander;
import com.genersoft.iot.vmp.gb28181.transmit.SIPSender;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.impl.SIPCommanderFroPlatform;
import com.genersoft.iot.vmp.gb28181.transmit.callback.DeferredResultHolder;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommanderForPlatform;
import com.genersoft.iot.vmp.gb28181.transmit.event.request.ISIPRequestProcessor;
import com.genersoft.iot.vmp.gb28181.transmit.event.request.SIPRequestProcessorParent;
import com.genersoft.iot.vmp.gb28181.utils.SipUtils;
@@ -25,11 +20,7 @@
import com.genersoft.iot.vmp.media.zlm.ZLMRTPServerFactory;
import com.genersoft.iot.vmp.media.zlm.ZlmHttpHookSubscribe;
import com.genersoft.iot.vmp.media.zlm.dto.*;
import com.genersoft.iot.vmp.service.IMediaServerService;
import com.genersoft.iot.vmp.service.IMediaService;
import com.genersoft.iot.vmp.service.IPlayService;
import com.genersoft.iot.vmp.service.IStreamProxyService;
import com.genersoft.iot.vmp.service.IStreamPushService;
import com.genersoft.iot.vmp.service.*;
import com.genersoft.iot.vmp.service.bean.MessageForPushChannel;
import com.genersoft.iot.vmp.service.bean.SSRCInfo;
import com.genersoft.iot.vmp.service.redisMsg.RedisGbPlayMsgListener;
@@ -37,8 +28,6 @@
import com.genersoft.iot.vmp.storager.IRedisCatchStorage;
import com.genersoft.iot.vmp.storager.IVideoManagerStorage;
import com.genersoft.iot.vmp.utils.DateUtil;
import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
import com.genersoft.iot.vmp.vmanager.bean.WVPResult;
import gov.nist.javax.sdp.TimeDescriptionImpl;
import gov.nist.javax.sdp.fields.TimeField;
import gov.nist.javax.sip.message.SIPRequest;
@@ -136,6 +125,9 @@

   @Autowired
   private SipConfig config;

    @Autowired
    private VideoStreamSessionManager streamSession;



@@ -394,7 +386,7 @@
                    }
                    SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
                            device.getDeviceId(), channelId,
                            mediaTransmissionTCP);
                            mediaTransmissionTCP, platform.isRtcp());

                    if (tcpActive != null) {
                        sendRtpItem.setTcpActive(tcpActive);
@@ -530,10 +522,9 @@

                            // 写入redis， 超时时回复
                            redisCatchStorage.updateSendRTPSever(sendRtpItem);
                            MediaServerItem finalMediaServerItem = mediaServerItem;
                            playService.play(mediaServerItem, ssrcInfo, device, channelId, hookEvent, errorEvent, (code, msg) -> {
                                logger.info("[上级点播]超时, 用户：{}， 通道：{}", username, channelId);
                                redisCatchStorage.deleteSendRTPServer(platform.getServerGBId(), channelId, callIdHeader.getCallId(), null);
                                logger.info("[上级点播]超时, 用户：{}， 通道：{}", username, finalChannelId);
                                redisCatchStorage.deleteSendRTPServer(platform.getServerGBId(), finalChannelId, callIdHeader.getCallId(), null);
                            });
                        } else {
                            sendRtpItem.setStreamId(playTransaction.getStream());
@@ -591,7 +582,7 @@
                // 自平台内容
                SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
                        gbStream.getApp(), gbStream.getStream(), channelId,
                        mediaTransmissionTCP);
                        mediaTransmissionTCP, platform.isRtcp());

                if (sendRtpItem == null) {
                    logger.warn("服务器端口资源不足");
@@ -631,7 +622,7 @@
                // 自平台内容
                SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
                        gbStream.getApp(), gbStream.getStream(), channelId,
                        mediaTransmissionTCP);
                        mediaTransmissionTCP, platform.isRtcp());

                if (sendRtpItem == null) {
                    logger.warn("服务器端口资源不足");
@@ -748,7 +739,7 @@
                dynamicTask.stop(callIdHeader.getCallId());
                if (serverId.equals(userSetting.getServerId())) {
                    SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, finalPort, ssrc, requesterId,
                            app, stream, channelId, mediaTransmissionTCP);
                            app, stream, channelId, mediaTransmissionTCP, platform.isRtcp());

                    if (sendRtpItem == null) {
                        logger.warn("上级点时创建sendRTPItem失败，可能是服务器端口资源不足");
@@ -810,7 +801,7 @@
        // 发送redis消息
        redisGbPlayMsgListener.sendMsg(streamPushItem.getServerId(), streamPushItem.getMediaServerId(),
                streamPushItem.getApp(), streamPushItem.getStream(), addressStr, port, ssrc, requesterId,
                channelId, mediaTransmissionTCP, null, responseSendItemMsg -> {
                channelId, mediaTransmissionTCP, platform.isRtcp(), null, responseSendItemMsg -> {
                    SendRtpItem sendRtpItem = responseSendItemMsg.getSendRtpItem();
                    if (sendRtpItem == null || responseSendItemMsg.getMediaServerItem() == null) {
                        logger.warn("服务器端口资源不足");
@@ -908,15 +899,15 @@
        if (audioBroadcastCatch == null) {
            logger.warn("来自设备的Invite请求非语音广播，已忽略，requesterId： {}/{}", requesterId, channelId);
            try {
                responseAck(serverTransaction, Response.FORBIDDEN);
                responseAck(request, Response.FORBIDDEN);
            } catch (SipException | InvalidArgumentException | ParseException e) {
                logger.error("[命令发送失败] 来自设备的Invite请求非语音广播 FORBIDDEN: {}", e.getMessage());
            }
            return;
        }
        Request request = serverTransaction.getRequest();
        if (device != null) {
            logger.info("收到设备" + requesterId + "的语音广播Invite请求");

            try {
                responseAck(request, Response.TRYING);
            } catch (SipException | InvalidArgumentException | ParseException e) {
@@ -985,7 +976,7 @@
                if (mediaServerItem == null) {
                    logger.warn("未找到可用的zlm");
                    try {
                        responseAck(serverTransaction, Response.BUSY_HERE);
                        responseAck(request, Response.BUSY_HERE);
                    } catch (SipException | InvalidArgumentException | ParseException e) {
                        logger.error("[命令发送失败] invite 未找到可用的zlm: {}", e.getMessage());
                    }
@@ -993,11 +984,12 @@
                }
                SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
                        device.getDeviceId(), audioBroadcastCatch.getChannelId(),
                        mediaTransmissionTCP);
                        mediaTransmissionTCP, false);

                if (sendRtpItem == null) {
                    logger.warn("服务器端口资源不足");
                    try {
                        responseAck(serverTransaction, Response.BUSY_HERE);
                        responseAck(request, Response.BUSY_HERE);
                    } catch (SipException | InvalidArgumentException | ParseException e) {
                        logger.error("[命令发送失败] invite 服务器端口资源不足: {}", e.getMessage());
                    }
@@ -1019,12 +1011,16 @@
                sendRtpItem.setStreamId(stream);
                sendRtpItem.setPt(8);
                sendRtpItem.setUsePs(false);
                sendRtpItem.setRtcp(false);
                sendRtpItem.setOnlyAudio(true);
                redisCatchStorage.updateSendRTPSever(sendRtpItem);

                Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItem, app, stream);
                if (streamReady) {
                    sendOk(device, sendRtpItem, sdp, serverTransaction, mediaServerItem, mediaTransmissionTCP, ssrc);
                    SIPResponse sipResponse = sendOk(device, sendRtpItem, sdp, request, mediaServerItem, mediaTransmissionTCP, ssrc);
                    // 添加事务信息
                    streamSession.put(device.getDeviceId(), audioBroadcastCatch.getChannelId(), request.getCallIdHeader().getCallId()
                            , stream,  sendRtpItem.getSsrc(), mediaServerItem.getId(), sipResponse, VideoStreamSessionManager.SessionType.broadcast );
                }else {
                    logger.warn("[语音通话]， 未发现待推送的流,app={},stream={}", app, stream);
                    playService.stopAudioBroadcast(device.getDeviceId(), audioBroadcastCatch.getChannelId());
@@ -1042,7 +1038,8 @@
        }
    }

    void sendOk(Device device, SendRtpItem sendRtpItem, SessionDescription sdp, ServerTransaction serverTransaction,  MediaServerItem mediaServerItem, boolean mediaTransmissionTCP, String ssrc){
    SIPResponse sendOk(Device device, SendRtpItem sendRtpItem, SessionDescription sdp, SIPRequest request,  MediaServerItem mediaServerItem, boolean mediaTransmissionTCP, String ssrc){
        SIPResponse sipResponse = null;
        try {
            sendRtpItem.setStatus(2);
            redisCatchStorage.updateSendRTPSever(sendRtpItem);
@@ -1078,15 +1075,17 @@
            parentPlatform.setServerPort(device.getPort());
            parentPlatform.setServerGBId(device.getDeviceId());

            SIPResponse sipResponse = responseSdpAck(serverTransaction, content.toString(), parentPlatform);
            sipResponse = responseSdpAck(request, content.toString(), parentPlatform);

            AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(device.getDeviceId(), sendRtpItem.getChannelId());

            audioBroadcastCatch.setStatus(AudioBroadcastCatchStatus.Ok);
            audioBroadcastCatch.setSipTransactionInfoByRequset(sipResponse);
            audioBroadcastManager.update(audioBroadcastCatch);

        } catch (SipException | InvalidArgumentException | ParseException | SdpParseException e) {
            logger.error("[命令发送失败] 语音对讲 回复200OK（SDP）: {}", e.getMessage());
        }
        return sipResponse;
    }
}