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wvp-GB28181-pro
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完善语音对讲

结构优化
648540858 2023-02-22 18:06:52 +08:00
parent a209d17390
commit 100252a253
6 changed files with 74 additions and 42 deletions
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3
src/main/java/com/genersoft/iot/vmp/conf/redis/RedisConfig.java
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@ -3,6 +3,7 @@ package com.genersoft.iot.vmp.conf.redis;
import com.genersoft.iot.vmp.common.VideoManagerConstants;
import com.genersoft.iot.vmp.service.redisMsg.*;
import com.genersoft.iot.vmp.utils.redis.FastJsonRedisSerializer;
import org.springframework.beans.factory.annotation.Autowired;
import org.springframework.cache.annotation.CachingConfigurerSupport;
import org.springframework.context.annotation.Bean;
@ -13,8 +14,6 @@ import org.springframework.data.redis.listener.PatternTopic;
import org.springframework.data.redis.listener.RedisMessageListenerContainer;
import org.springframework.data.redis.serializer.StringRedisSerializer;
import com.genersoft.iot.vmp.utils.redis.FastJsonRedisSerializer;
/**
* @description:Redis中间件配置类，使用spring-data-redis集成，自动从application.yml中加载redis配置

50
src/main/java/com/genersoft/iot/vmp/gb28181/bean/AudioBroadcastCatch.java
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@ -1,6 +1,7 @@
package com.genersoft.iot.vmp.gb28181.bean;
import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem;
import gov.nist.javax.sip.message.SIPResponse;
/**
@ -10,10 +11,18 @@ import gov.nist.javax.sip.message.SIPResponse;
public class AudioBroadcastCatch {
public AudioBroadcastCatch(String deviceId, String channelId, AudioBroadcastCatchStatus status) {
public AudioBroadcastCatch(String deviceId,
String channelId,
AudioBroadcastCatchStatus status,
MediaServerItem mediaServerItem,
String app,
String stream) {
this.deviceId = deviceId;
this.channelId = channelId;
this.status = status;
this.mediaServerItem = mediaServerItem;
this.app = app;
this.stream = stream;
}
public AudioBroadcastCatch() {
@ -29,6 +38,21 @@ public class AudioBroadcastCatch {
*/
private String channelId;
/**
* 使用的流媒体
*/
private MediaServerItem mediaServerItem;
/**
* 待推送给设备的流应用名
*/
private String app;
/**
* 待推送给设备的流ID
*/
private String stream;
/**
* 语音广播状态
*/
@ -68,6 +92,30 @@ public class AudioBroadcastCatch {
return sipTransactionInfo;
}
public MediaServerItem getMediaServerItem() {
return mediaServerItem;
}
public void setMediaServerItem(MediaServerItem mediaServerItem) {
this.mediaServerItem = mediaServerItem;
}
public String getApp() {
return app;
}
public void setApp(String app) {
this.app = app;
}
public String getStream() {
return stream;
}
public void setStream(String stream) {
this.stream = stream;
}
public void setSipTransactionInfo(SipTransactionInfo sipTransactionInfo) {
this.sipTransactionInfo = sipTransactionInfo;
}

40
src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/InviteRequestProcessor.java
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@ -903,8 +903,8 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
// 非上级平台请求，查询是否设备请求（通常为接收语音广播的设备）
Device device = redisCatchStorage.getDevice(requesterId);
AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(requesterId, channelId);
if (audioBroadcastCatch == null) {
AudioBroadcastCatch broadcastCatch = audioBroadcastManager.get(requesterId, channelId);
if (broadcastCatch == null) {
logger.warn("来自设备的Invite请求非语音广播，已忽略，requesterId： {}/{}", requesterId, channelId);
try {
responseAck(request, Response.FORBIDDEN);
@ -915,13 +915,13 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
}
if (device != null) {
logger.info("收到设备" + requesterId + "的语音广播Invite请求");
String key = VideoManagerConstants.BROADCAST_WAITE_INVITE + device.getDeviceId() + audioBroadcastCatch.getChannelId();
String key = VideoManagerConstants.BROADCAST_WAITE_INVITE + device.getDeviceId() + broadcastCatch.getChannelId();
dynamicTask.stop(key);
try {
responseAck(request, Response.TRYING);
} catch (SipException | InvalidArgumentException | ParseException e) {
logger.error("[命令发送失败] invite BAD_REQUEST: {}", e.getMessage());
playService.stopAudioBroadcast(device.getDeviceId(), audioBroadcastCatch.getChannelId());
playService.stopAudioBroadcast(device.getDeviceId(), broadcastCatch.getChannelId());
return;
}
String contentString = new String(request.getRawContent());
@ -977,7 +977,7 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
responseAck(request, Response.UNSUPPORTED_MEDIA_TYPE); // 不支持的格式，发415
} catch (SipException | InvalidArgumentException | ParseException e) {
logger.error("[命令发送失败] invite 不支持的媒体格式: {}", e.getMessage());
playService.stopAudioBroadcast(device.getDeviceId(), audioBroadcastCatch.getChannelId());
playService.stopAudioBroadcast(device.getDeviceId(), broadcastCatch.getChannelId());
return;
}
return;
@ -986,19 +986,9 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
logger.info("设备{}请求语音流，地址：{}:{}，ssrc：{}, {}", requesterId, addressStr, port, ssrc,
mediaTransmissionTCP ? (tcpActive? "TCP主动":"TCP被动") : "UDP");
MediaServerItem mediaServerItem = playService.getNewMediaServerItem(device);
if (mediaServerItem == null) {
logger.warn("未找到可用的zlm");
try {
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());
playService.stopAudioBroadcast(device.getDeviceId(), audioBroadcastCatch.getChannelId());
}
return;
}
MediaServerItem mediaServerItem = broadcastCatch.getMediaServerItem();
SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, addressStr, port, ssrc, requesterId,
device.getDeviceId(), audioBroadcastCatch.getChannelId(),
device.getDeviceId(), broadcastCatch.getChannelId(),
mediaTransmissionTCP, false);
if (sendRtpItem == null) {
@ -1007,22 +997,20 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
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());
playService.stopAudioBroadcast(device.getDeviceId(), audioBroadcastCatch.getChannelId());
playService.stopAudioBroadcast(device.getDeviceId(), broadcastCatch.getChannelId());
return;
}
return;
}
String app = "broadcast";
String stream = device.getDeviceId() + "_" + audioBroadcastCatch.getChannelId();
CallIdHeader callIdHeader = (CallIdHeader) request.getHeader(CallIdHeader.NAME);
sendRtpItem.setPlayType(InviteStreamType.TALK);
sendRtpItem.setCallId(callIdHeader.getCallId());
sendRtpItem.setPlatformId(requesterId);
sendRtpItem.setStatus(1);
sendRtpItem.setApp(app);
sendRtpItem.setStreamId(stream);
sendRtpItem.setApp(broadcastCatch.getApp());
sendRtpItem.setStreamId(broadcastCatch.getStream());
sendRtpItem.setPt(8);
sendRtpItem.setUsePs(false);
sendRtpItem.setRtcp(false);
@ -1034,22 +1022,22 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
redisCatchStorage.updateSendRTPSever(sendRtpItem);
Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItem, app, stream);
Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItem, broadcastCatch.getApp(), broadcastCatch.getStream());
if (streamReady) {
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);
logger.warn("[语音通话]， 未发现待推送的流,app={},stream={}", broadcastCatch.getApp(), broadcastCatch.getStream());
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());
return;
}
playService.stopAudioBroadcast(device.getDeviceId(), audioBroadcastCatch.getChannelId());
playService.stopAudioBroadcast(device.getDeviceId(), broadcastCatch.getChannelId());
}
} catch (SdpException e) {
logger.error("[SDP解析异常]", e);
playService.stopAudioBroadcast(device.getDeviceId(), audioBroadcastCatch.getChannelId());
playService.stopAudioBroadcast(device.getDeviceId(), broadcastCatch.getChannelId());
}
} else {
logger.warn("来自无效设备/平台的请求");

13
src/main/java/com/genersoft/iot/vmp/media/zlm/ZLMHttpHookListener.java
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@ -274,18 +274,17 @@ public class ZLMHttpHookListener {
logger.info("[ZLM HOOK] 流注销, {}->{}->{}/{}", param.getMediaServerId(), param.getSchema(), param.getApp(), param.getStream());
}
MediaServerItem mediaInfo = mediaServerService.getOne(param.getMediaServerId());
JSONObject json = (JSONObject) JSON.toJSON(param);
taskExecutor.execute(() -> {
ZlmHttpHookSubscribe.Event subscribe = this.subscribe.sendNotify(HookType.on_stream_changed, json);
if (subscribe != null) {
MediaServerItem mediaInfo = mediaServerService.getOne(param.getMediaServerId());
if (mediaInfo != null) {
subscribe.response(mediaInfo, json);
}
}
// 流消失移除redis play
List<OnStreamChangedHookParam.MediaTrack> tracks = param.getTracks();
if (param.isRegist()) {
if (param.getOriginType() == OriginType.RTMP_PUSH.ordinal()
|| param.getOriginType() == OriginType.RTSP_PUSH.ordinal()
@ -343,7 +342,7 @@ public class ZLMHttpHookListener {
}
// 开启语音对讲通道
try {
playService.audioBroadcastCmd(device, channelId, 60, (msg)->{
playService.audioBroadcastCmd(device, channelId, 60, mediaInfo, param.getApp(), param.getStream(), (msg)->{
logger.info("[语音对讲] 通道建立成功, device: {}, channel: {}", deviceId, channelId);
});
} catch (InvalidArgumentException | ParseException | SipException e) {
@ -375,7 +374,7 @@ public class ZLMHttpHookListener {
if (sendRtpItem == null) {
// TODO 可能数据错误，重新开启语音通道
}else {
MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
MediaServerItem mediaServerItem = mediaServerService.getOne(sendRtpItem.getMediaServerId());
logger.info("rtp/{}开始向上级推流, 目标={}:{}，SSRC={}", sendRtpItem.getStreamId(), sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc());
Map<String, Object> sendParam = new HashMap<>(12);
sendParam.put("vhost","__defaultVhost__");
@ -389,12 +388,12 @@ public class ZLMHttpHookListener {
JSONObject jsonObject;
if (sendRtpItem.isTcpActive()) {
jsonObject = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, sendParam);
jsonObject = zlmrtpServerFactory.startSendRtpPassive(mediaServerItem, sendParam);
} else {
sendParam.put("is_udp", sendRtpItem.isTcp() ? "0" : "1");
sendParam.put("dst_url", sendRtpItem.getIp());
sendParam.put("dst_port", sendRtpItem.getPort());
jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaInfo, sendParam);
jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaServerItem, sendParam);
}
if (jsonObject != null && jsonObject.getInteger("code") == 0) {
logger.info("[语音对讲] 自动推流成功, device: {}, channel: {}", deviceId, channelId);

4
src/main/java/com/genersoft/iot/vmp/service/IPlayService.java
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@ -12,9 +12,7 @@ import com.genersoft.iot.vmp.service.bean.PlayBackCallback;
import com.genersoft.iot.vmp.service.bean.SSRCInfo;
import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
import com.genersoft.iot.vmp.vmanager.gb28181.play.bean.AudioBroadcastEvent;
import com.genersoft.iot.vmp.vmanager.bean.WVPResult;
import gov.nist.javax.sip.message.SIPResponse;
import org.springframework.web.context.request.async.DeferredResult;
import javax.sip.InvalidArgumentException;
import javax.sip.SipException;
@ -62,7 +60,7 @@ public interface IPlayService {
AudioBroadcastResult audioBroadcast(Device device, String channelId);
void stopAudioBroadcast(String deviceId, String channelId);
void audioBroadcastCmd(Device device, String channelId, int timeout, AudioBroadcastEvent event) throws InvalidArgumentException, ParseException, SipException;
void audioBroadcastCmd(Device device, String channelId, int timeout, MediaServerItem mediaServerItem, String sourceApp, String sourceStream, AudioBroadcastEvent event) throws InvalidArgumentException, ParseException, SipException;
void pauseRtp(String streamId) throws ServiceException, InvalidArgumentException, ParseException, SipException;

6
src/main/java/com/genersoft/iot/vmp/service/impl/PlayServiceImpl.java
View File

@ -1011,7 +1011,7 @@ public class PlayServiceImpl implements IPlayService {
}
@Override
public void audioBroadcastCmd(Device device, String channelId, int timeout, AudioBroadcastEvent event) throws InvalidArgumentException, ParseException, SipException {
public void audioBroadcastCmd(Device device, String channelId, int timeout, MediaServerItem mediaServerItem, String sourceApp, String sourceStream, AudioBroadcastEvent event) throws InvalidArgumentException, ParseException, SipException {
if (device == null || channelId == null) {
return;
}
@ -1027,7 +1027,6 @@ public class PlayServiceImpl implements IPlayService {
SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, null, null);
if (sendRtpItem != null && sendRtpItem.isOnlyAudio()) {
// 查询流是否存在，不存在则认为是异常状态
MediaServerItem mediaServerItem = mediaServerService.getOne(sendRtpItem.getMediaServerId());
Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServerItem, sendRtpItem.getApp(), sendRtpItem.getStreamId());
if (streamReady) {
logger.warn("语音广播已经开启： {}", channelId);
@ -1042,7 +1041,8 @@ public class PlayServiceImpl implements IPlayService {
// 发送通知
cmder.audioBroadcastCmd(device, channelId, eventResultForOk -> {
// 发送成功
AudioBroadcastCatch audioBroadcastCatch = new AudioBroadcastCatch(device.getDeviceId(), channelId, AudioBroadcastCatchStatus.Ready);
AudioBroadcastCatch audioBroadcastCatch = new AudioBroadcastCatch(device.getDeviceId(), channelId,
AudioBroadcastCatchStatus.Ready, mediaServerItem, sourceApp, sourceStream);
audioBroadcastManager.update(audioBroadcastCatch);
}, eventResultForError -> {
// 发送失败