修复合并的bug

648540858

2022-11-29 38a85d432ae9bb861dbcbf090d68fb3dca0d85f6

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

@@ -11,6 +11,7 @@
import com.genersoft.iot.vmp.gb28181.transmit.SIPProcessorObserver;
import com.genersoft.iot.vmp.gb28181.transmit.SIPSender;
import com.genersoft.iot.vmp.gb28181.transmit.callback.DeferredResultHolder;
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.event.request.ISIPRequestProcessor;
import com.genersoft.iot.vmp.gb28181.transmit.event.request.SIPRequestProcessorParent;
@@ -97,7 +98,7 @@
    private IMediaServerService mediaServerService;

    @Autowired
    private IMediaService mediaService;
    private ISIPCommander commander;

   @Autowired
   private ZLMRESTfulUtils zlmresTfulUtils;
@@ -125,6 +126,9 @@

   @Autowired
   private SipConfig config;

    @Autowired
    private VideoStreamSessionManager streamSession;



@@ -382,8 +386,7 @@
                        return;
                    }
                    SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
                            device.getDeviceId(), channelId,
                            mediaTransmissionTCP);
                            device.getDeviceId(), channelId, mediaTransmissionTCP, platform.isRtcp());

                    if (tcpActive != null) {
                        sendRtpItem.setTcpActive(tcpActive);
@@ -578,8 +581,7 @@
            if (streamReady) {
                // 自平台内容
                SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
                        gbStream.getApp(), gbStream.getStream(), channelId,
                        mediaTransmissionTCP);
                        gbStream.getApp(), gbStream.getStream(), channelId, mediaTransmissionTCP, platform.isRtcp());

                if (sendRtpItem == null) {
                    logger.warn("服务器端口资源不足");
@@ -618,8 +620,7 @@
            if (streamReady) {
                // 自平台内容
                SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
                        gbStream.getApp(), gbStream.getStream(), channelId,
                        mediaTransmissionTCP);
                        gbStream.getApp(), gbStream.getStream(), channelId, mediaTransmissionTCP, platform.isRtcp());

                if (sendRtpItem == null) {
                    logger.warn("服务器端口资源不足");
@@ -736,7 +737,7 @@
                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");
@@ -798,7 +799,7 @@
        // 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");
@@ -904,6 +905,7 @@
        }
        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) {
@@ -980,7 +982,8 @@
                }
                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 {
@@ -998,7 +1001,7 @@
                String stream = device.getDeviceId() + "_" + audioBroadcastCatch.getChannelId();

                CallIdHeader callIdHeader = (CallIdHeader) request.getHeader(CallIdHeader.NAME);
                sendRtpItem.setPlayType(InviteStreamType.PLAY);
                sendRtpItem.setPlayType(InviteStreamType.TALK);
                sendRtpItem.setCallId(callIdHeader.getCallId());
                sendRtpItem.setPlatformId(requesterId);
                sendRtpItem.setStatus(1);
@@ -1006,6 +1009,7 @@
                sendRtpItem.setStreamId(stream);
                sendRtpItem.setPt(8);
                sendRtpItem.setUsePs(false);
                sendRtpItem.setRtcp(false);
                sendRtpItem.setOnlyAudio(true);
                redisCatchStorage.updateSendRTPSever(sendRtpItem);

@@ -1014,6 +1018,11 @@
                    sendOk(device, sendRtpItem, sdp, request, mediaServerItem, mediaTransmissionTCP, ssrc);
                }else {
                    logger.warn("[语音通话]， 未发现待推送的流,app={},stream={}", app, stream);
                    try {
                        responseAck(request, Response.GONE);
                    } catch (SipException | InvalidArgumentException | ParseException e) {
                        logger.error("[命令发送失败] 语音通话 回复410失败， {}", e.getMessage());
                    }
                    playService.stopAudioBroadcast(device.getDeviceId(), audioBroadcastCatch.getChannelId());
                }
            } catch (SdpException e) {
@@ -1029,7 +1038,8 @@
        }
    }

    void sendOk(Device device, SendRtpItem sendRtpItem, SessionDescription sdp, SIPRequest request,  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);
@@ -1065,15 +1075,17 @@
            parentPlatform.setServerPort(device.getPort());
            parentPlatform.setServerGBId(device.getDeviceId());

            SIPResponse sipResponse = responseSdpAck(request, 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;
    }
}