优化rtcp判断

648540858

2022-11-21 6b03568c5dd128a3d71c02fb1a3a76a4344a4920

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

@@ -126,6 +126,9 @@
   @Autowired
   private SipConfig config;

    @Autowired
    private VideoStreamSessionManager streamSession;



    @Autowired
@@ -383,7 +386,7 @@
                    }
                    SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
                            device.getDeviceId(), channelId,
                            mediaTransmissionTCP);
                            mediaTransmissionTCP, platform.isRtcp());

                    if (tcpActive != null) {
                        sendRtpItem.setTcpActive(tcpActive);
@@ -579,7 +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");
@@ -619,7 +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");
@@ -736,7 +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");
@@ -798,7 +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");
@@ -904,6 +907,7 @@
        }
        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 +984,8 @@
                }
                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 {
@@ -1006,12 +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, request, 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());
@@ -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;
    }
}