Merge branch 'wvp-28181-2.0'

648540858

2022-11-25 d7623b5ca864d98ab954a458b049641eb82f4f75

 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;
@@ -385,8 +386,7 @@
                        return;
                    }
                    SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
                            device.getDeviceId(), channelId,
                            mediaTransmissionTCP, platform.isRtcp());
                            device.getDeviceId(), channelId, mediaTransmissionTCP, platform.isRtcp());

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

                if (sendRtpItem == null) {
                    logger.warn("服务器端口资源不足");
@@ -621,8 +620,7 @@
            if (streamReady) {
                // 自平台内容
                SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
                        gbStream.getApp(), gbStream.getStream(), channelId,
                        mediaTransmissionTCP, platform.isRtcp());
                        gbStream.getApp(), gbStream.getStream(), channelId, mediaTransmissionTCP, platform.isRtcp());

                if (sendRtpItem == null) {
                    logger.warn("服务器端口资源不足");
@@ -801,7 +799,7 @@
        // 发送redis消息
        redisGbPlayMsgListener.sendMsg(streamPushItem.getServerId(), streamPushItem.getMediaServerId(),
                streamPushItem.getApp(), streamPushItem.getStream(), addressStr, port, ssrc, requesterId,
                channelId, mediaTransmissionTCP, platform.isRtcp(), 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");
@@ -1003,7 +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);
@@ -1017,12 +1015,14 @@

                Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItem, app, stream);
                if (streamReady) {
                    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 );
                    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) {