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wvp-GB28181-pro
mirror of https://github.com/648540858/wvp-GB28181-pro
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Merge branch 'talk' into main-dev 

# Conflicts:
#	src/main/java/com/genersoft/iot/vmp/conf/redis/RedisConfig.java
#	src/main/java/com/genersoft/iot/vmp/gb28181/bean/AudioBroadcastCatch.java
#	src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/InviteRequestProcessor.java
#	src/main/java/com/genersoft/iot/vmp/media/zlm/ZLMHttpHookListener.java
#	src/main/java/com/genersoft/iot/vmp/service/IPlayService.java
#	src/main/java/com/genersoft/iot/vmp/service/impl/PlayServiceImpl.java
结构优化
648540858 2023-03-20 14:32:53 +08:00
parent ebe24a6c32 a000ed6062
commit 039fbf7e24
27 changed files with 426 additions and 336 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

10
src/main/java/com/genersoft/iot/vmp/gb28181/bean/AudioBroadcastCatch.java
View File

@ -63,6 +63,8 @@ public class AudioBroadcastCatch {
*/
private SipTransactionInfo sipTransactionInfo;
private MediaServerItem mediaServerItem;
public String getDeviceId() {
return deviceId;
@ -123,4 +125,12 @@ public class AudioBroadcastCatch {
public void setSipTransactionInfoByRequset(SIPResponse response) {
this.sipTransactionInfo = new SipTransactionInfo(response, false);
}
public MediaServerItem getMediaServerItem() {
return mediaServerItem;
}
public void setMediaServerItem(MediaServerItem mediaServerItem) {
this.mediaServerItem = mediaServerItem;
}
}

23
src/main/java/com/genersoft/iot/vmp/gb28181/bean/SendRtpItem.java
View File

@ -49,7 +49,7 @@ public class SendRtpItem {
/**
* 设备推流的streamId
*/
private String streamId;
private String stream;
/**
* 是否为tcp
@ -117,6 +117,11 @@ public class SendRtpItem {
*/
private InviteStreamType playType;
/**
* 发流的同时收流
*/
private String receiveStream;
public String getIp() {
return ip;
}
@ -181,12 +186,12 @@ public class SendRtpItem {
this.app = app;
}
public String getStreamId() {
return streamId;
public String getStream() {
return stream;
}
public void setStreamId(String streamId) {
this.streamId = streamId;
public void setStream(String stream) {
this.stream = stream;
}
public boolean isTcp() {
@ -292,4 +297,12 @@ public class SendRtpItem {
public void setRtcp(boolean rtcp) {
this.rtcp = rtcp;
}
public String getReceiveStream() {
return receiveStream;
}
public void setReceiveStream(String receiveStream) {
this.receiveStream = receiveStream;
}
}

1
src/main/java/com/genersoft/iot/vmp/gb28181/event/SipSubscribe.java
View File

@ -1,6 +1,5 @@
package com.genersoft.iot.vmp.gb28181.event;
import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException;
import com.genersoft.iot.vmp.gb28181.bean.DeviceNotFoundEvent;
import gov.nist.javax.sip.message.SIPRequest;
import org.slf4j.Logger;

3
src/main/java/com/genersoft/iot/vmp/gb28181/session/VideoStreamSessionManager.java
View File

@ -29,7 +29,8 @@ public class VideoStreamSessionManager {
play,
playback,
download,
broadcast
broadcast,
talk
}
/**

4
src/main/java/com/genersoft/iot/vmp/gb28181/task/SipRunner.java
View File

@ -94,12 +94,12 @@ public class SipRunner implements CommandLineRunner {
if (sendRtpItems.size() > 0) {
for (SendRtpItem sendRtpItem : sendRtpItems) {
MediaServerItem mediaServerItem = mediaServerService.getOne(sendRtpItem.getMediaServerId());
redisCatchStorage.deleteSendRTPServer(sendRtpItem.getPlatformId(),sendRtpItem.getChannelId(), sendRtpItem.getCallId(),sendRtpItem.getStreamId());
redisCatchStorage.deleteSendRTPServer(sendRtpItem.getPlatformId(),sendRtpItem.getChannelId(), sendRtpItem.getCallId(),sendRtpItem.getStream());
if (mediaServerItem != null) {
Map<String, Object> param = new HashMap<>();
param.put("vhost","__defaultVhost__");
param.put("app",sendRtpItem.getApp());
param.put("stream",sendRtpItem.getStreamId());
param.put("stream",sendRtpItem.getStream());
param.put("ssrc",sendRtpItem.getSsrc());
JSONObject jsonObject = zlmresTfulUtils.stopSendRtp(mediaServerItem, param);
if (jsonObject != null && jsonObject.getInteger("code") == 0) {

7
src/main/java/com/genersoft/iot/vmp/gb28181/transmit/cmd/ISIPCommander.java
View File

@ -2,10 +2,7 @@ package com.genersoft.iot.vmp.gb28181.transmit.cmd;
import com.genersoft.iot.vmp.common.StreamInfo;
import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException;
import com.genersoft.iot.vmp.gb28181.bean.Device;
import com.genersoft.iot.vmp.gb28181.bean.DeviceAlarm;
import com.genersoft.iot.vmp.gb28181.bean.InviteStreamCallback;
import com.genersoft.iot.vmp.gb28181.bean.SipTransactionInfo;
import com.genersoft.iot.vmp.gb28181.bean.*;
import com.genersoft.iot.vmp.gb28181.event.SipSubscribe;
import com.genersoft.iot.vmp.media.zlm.ZlmHttpHookSubscribe;
import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem;
@ -131,7 +128,7 @@ public interface ISIPCommander {
*/
void streamByeCmd(Device device, String channelId, String stream, String callId, SipSubscribe.Event okEvent) throws InvalidArgumentException, SipException, ParseException, SsrcTransactionNotFoundException;
void talkStreamCmd(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo, Device device, String channelId, String callId, ZlmHttpHookSubscribe.Event event, ZlmHttpHookSubscribe.Event eventForPush, SipSubscribe.Event okEvent, SipSubscribe.Event errorEvent) throws InvalidArgumentException, SipException, ParseException;
void talkStreamCmd(MediaServerItem mediaServerItem, SendRtpItem sendRtpItem, Device device, String channelId, String callId, ZlmHttpHookSubscribe.Event event, ZlmHttpHookSubscribe.Event eventForPush, SipSubscribe.Event okEvent, SipSubscribe.Event errorEvent) throws InvalidArgumentException, SipException, ParseException;
void streamByeCmd(Device device, String channelId, String stream, String callId) throws InvalidArgumentException, ParseException, SipException, SsrcTransactionNotFoundException;

22
src/main/java/com/genersoft/iot/vmp/gb28181/transmit/cmd/impl/SIPCommander.java
View File

@ -32,7 +32,6 @@ import org.springframework.beans.factory.annotation.Autowired;
import org.springframework.context.annotation.DependsOn;
import org.springframework.stereotype.Component;
import org.springframework.util.ObjectUtils;
import org.springframework.util.StringUtils;
import javax.sip.InvalidArgumentException;
import javax.sip.ResponseEvent;
@ -584,9 +583,9 @@ public class SIPCommander implements ISIPCommander {
}
@Override
public void talkStreamCmd(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo, Device device, String channelId, String callId, ZlmHttpHookSubscribe.Event event, ZlmHttpHookSubscribe.Event eventForPush, SipSubscribe.Event okEvent, SipSubscribe.Event errorEvent) throws InvalidArgumentException, SipException, ParseException {
public void talkStreamCmd(MediaServerItem mediaServerItem, SendRtpItem sendRtpItem, Device device, String channelId, String callId, ZlmHttpHookSubscribe.Event event, ZlmHttpHookSubscribe.Event eventForPush, SipSubscribe.Event okEvent, SipSubscribe.Event errorEvent) throws InvalidArgumentException, SipException, ParseException {
String stream = ssrcInfo.getStream();
String stream = sendRtpItem.getStream();
if (device == null) {
return;
@ -597,7 +596,7 @@ public class SIPCommander implements ISIPCommander {
return;
}
logger.info("[语音对讲] {} 分配的ZLM为: {} [{}:{}]", stream, mediaServerItem.getId(), mediaServerItem.getIp(), ssrcInfo.getPort());
logger.info("[语音对讲] {} 分配的ZLM为: {} [{}:{}]", stream, mediaServerItem.getId(), mediaServerItem.getIp(), sendRtpItem.getPort());
HookSubscribeForStreamChange hookSubscribeForStreamChange = HookSubscribeFactory.on_stream_changed("rtp", stream, true, "rtsp", mediaServerItem.getId());
subscribe.addSubscribe(hookSubscribeForStreamChange, (MediaServerItem mediaServerItemInUse, JSONObject json) -> {
if (event != null) {
@ -622,24 +621,27 @@ public class SIPCommander implements ISIPCommander {
content.append("c=IN IP4 " + mediaServerItem.getSdpIp() + "\r\n");
content.append("t=0 0\r\n");
content.append("m=audio " + ssrcInfo.getPort() + " RTP/AVP 8\r\n");
content.append("m=audio " + sendRtpItem.getPort() + " TCP/RTP/AVP 8\r\n");
content.append("a=setup:passive\r\n");
content.append("a=connection:new\r\n");
content.append("a=sendrecv\r\n");
content.append("a=rtpmap:8 PCMA/8000\r\n");
content.append("y=" + ssrcInfo.getSsrc() + "\r\n");//ssrc
content.append("y=" + sendRtpItem.getSsrc() + "\r\n");//ssrc
// f字段:f= v/编码格式/分辨率/帧率/码率类型/码率大小a/编码格式/码率大小/采样率
content.append("f=v/////a/1/8/1" + "\r\n");
Request request = headerProvider.createInviteRequest(device, channelId, content.toString(), SipUtils.getNewViaTag(), SipUtils.getNewFromTag(), null, ssrcInfo.getSsrc(), callIdHeader);
Request request = headerProvider.createInviteRequest(device, channelId, content.toString(),
SipUtils.getNewViaTag(), SipUtils.getNewFromTag(), null, sendRtpItem.getSsrc(), callIdHeader);
sipSender.transmitRequest(sipLayer.getLocalIp(device.getLocalIp()), request, (e -> {
streamSession.remove(device.getDeviceId(), channelId, ssrcInfo.getStream());
mediaServerService.releaseSsrc(mediaServerItem.getId(), ssrcInfo.getSsrc());
streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
errorEvent.response(e);
}), e -> {
// 这里为例避免一个通道的点播只有一个callID这个参数使用一个固定值
ResponseEvent responseEvent = (ResponseEvent) e.event;
SIPResponse response = (SIPResponse) responseEvent.getResponse();
streamSession.put(device.getDeviceId(), channelId, "talk", stream, ssrcInfo.getSsrc(), mediaServerItem.getId(), response, VideoStreamSessionManager.SessionType.play);
streamSession.put(device.getDeviceId(), channelId, "talk", stream, sendRtpItem.getSsrc(), mediaServerItem.getId(), response, VideoStreamSessionManager.SessionType.talk);
okEvent.response(e);
});
}

2
src/main/java/com/genersoft/iot/vmp/gb28181/transmit/cmd/impl/SIPCommanderFroPlatform.java
View File

@ -694,7 +694,7 @@ public class SIPCommanderFroPlatform implements ISIPCommanderForPlatform {
MediaServerItem mediaServerItem = mediaServerService.getOne(mediaServerId);
if (mediaServerItem != null) {
mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
zlmrtpServerFactory.closeRtpServer(mediaServerItem, sendRtpItem.getStreamId());
zlmrtpServerFactory.closeRtpServer(mediaServerItem, sendRtpItem.getStream());
}
SIPRequest byeRequest = headerProviderPlatformProvider.createByeRequest(platform, sendRtpItem);
if (byeRequest == null) {

6
src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/AckRequestProcessor.java
View File

@ -102,12 +102,12 @@ public class AckRequestProcessor extends SIPRequestProcessorParent implements In
}
String isUdp = sendRtpItem.isTcp() ? "0" : "1";
MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
logger.info("收到ACK，rtp/{}开始向上级推流, 目标={}:{}，SSRC={}, RTCP={}", sendRtpItem.getStreamId(),
logger.info("收到ACK，rtp/{}开始向上级推流, 目标={}:{}，SSRC={}, RTCP={}", sendRtpItem.getStream(),
sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isRtcp());
Map<String, Object> param = new HashMap<>(12);
param.put("vhost","__defaultVhost__");
param.put("app",sendRtpItem.getApp());
param.put("stream",sendRtpItem.getStreamId());
param.put("stream",sendRtpItem.getStream());
param.put("ssrc", sendRtpItem.getSsrc());
param.put("src_port", sendRtpItem.getLocalPort());
param.put("pt", sendRtpItem.getPt());
@ -121,7 +121,7 @@ public class AckRequestProcessor extends SIPRequestProcessorParent implements In
if (mediaInfo == null) {
RequestPushStreamMsg requestPushStreamMsg = RequestPushStreamMsg.getInstance(
sendRtpItem.getMediaServerId(), sendRtpItem.getApp(), sendRtpItem.getStreamId(),
sendRtpItem.getMediaServerId(), sendRtpItem.getApp(), sendRtpItem.getStream(),
sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isTcp(),
sendRtpItem.getLocalPort(), sendRtpItem.getPt(), sendRtpItem.isUsePs(), sendRtpItem.isOnlyAudio());
redisGbPlayMsgListener.sendMsgForStartSendRtpStream(sendRtpItem.getServerId(), requestPushStreamMsg, json -> {

11
src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/ByeRequestProcessor.java
View File

@ -97,7 +97,7 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
if (sendRtpItem != null){
logger.info("[收到bye] {}/{}", sendRtpItem.getPlatformId(), sendRtpItem.getChannelId());
String streamId = sendRtpItem.getStreamId();
String streamId = sendRtpItem.getStream();
MediaServerItem mediaServerItem = mediaServerService.getOne(sendRtpItem.getMediaServerId());
if (mediaServerItem == null) {
return;
@ -105,7 +105,7 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
Boolean ready = zlmrtpServerFactory.isStreamReady(mediaServerItem, sendRtpItem.getApp(), streamId);
if (!ready) {
logger.info("[收到bye] 发现流{}/{}已经结束，不需处理", sendRtpItem.getApp(), sendRtpItem.getStreamId());
logger.info("[收到bye] 发现流{}/{}已经结束，不需处理", sendRtpItem.getApp(), sendRtpItem.getStream());
return;
}
Map<String, Object> param = new HashMap<>();
@ -113,7 +113,7 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
param.put("app",sendRtpItem.getApp());
param.put("stream",streamId);
param.put("ssrc",sendRtpItem.getSsrc());
logger.info("[收到bye] 停止向上级推流：{}", streamId);
logger.info("[收到bye] 停止推流：{}", streamId);
MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
redisCatchStorage.deleteSendRTPServer(sendRtpItem.getPlatformId(), sendRtpItem.getChannelId(), callIdHeader.getCallId(), null);
zlmrtpServerFactory.stopSendRtpStream(mediaInfo, param);
@ -129,15 +129,14 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
try {
logger.warn("[停止点播] {}/{}", sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
cmder.streamByeCmd(device, sendRtpItem.getChannelId(), streamId, null);
} catch (InvalidArgumentException | ParseException | SipException |
SsrcTransactionNotFoundException e) {
} catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
logger.error("[收到bye] {} 无其它观看者，通知设备停止推流， 发送BYE失败 {}",streamId, e.getMessage());
}
}
if (sendRtpItem.getPlayType().equals(InviteStreamType.PUSH)) {
MessageForPushChannel messageForPushChannel = MessageForPushChannel.getInstance(0,
sendRtpItem.getApp(), sendRtpItem.getStreamId(), sendRtpItem.getChannelId(),
sendRtpItem.getApp(), sendRtpItem.getStream(), sendRtpItem.getChannelId(),
sendRtpItem.getPlatformId(), null, null, sendRtpItem.getMediaServerId());
redisCatchStorage.sendStreamPushRequestedMsg(messageForPushChannel);
}

23
src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/InviteRequestProcessor.java
View File

@ -478,7 +478,7 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
if ("Playback".equalsIgnoreCase(sessionName)) {
sendRtpItem.setPlayType(InviteStreamType.PLAYBACK);
SSRCInfo ssrcInfo = mediaServerService.openRTPServer(mediaServerItem, null, device.isSsrcCheck(), true);
sendRtpItem.setStreamId(ssrcInfo.getStream());
sendRtpItem.setStream(ssrcInfo.getStream());
// 写入redis， 超时时回复
redisCatchStorage.updateSendRTPSever(sendRtpItem);
playService.playBack(mediaServerItem, ssrcInfo, device.getDeviceId(), channelId, DateUtil.formatter.format(start),
@ -523,7 +523,7 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
}
SSRCInfo ssrcInfo = mediaServerService.openRTPServer(mediaServerItem, streamId, null, device.isSsrcCheck(), false);
logger.info(JSONObject.toJSONString(ssrcInfo));
sendRtpItem.setStreamId(ssrcInfo.getStream());
sendRtpItem.setStream(ssrcInfo.getStream());
sendRtpItem.setSsrc(ssrc.equals(ssrcDefault) ? ssrcInfo.getSsrc() : ssrc);
// 写入redis， 超时时回复
@ -533,12 +533,12 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
redisCatchStorage.deleteSendRTPServer(platform.getServerGBId(), finalChannelId, callIdHeader.getCallId(), null);
});
} else {
sendRtpItem.setStreamId(playTransaction.getStream());
sendRtpItem.setStream(playTransaction.getStream());
// 写入redis， 超时时回复
redisCatchStorage.updateSendRTPSever(sendRtpItem);
JSONObject jsonObject = new JSONObject();
jsonObject.put("app", sendRtpItem.getApp());
jsonObject.put("stream", sendRtpItem.getStreamId());
jsonObject.put("stream", sendRtpItem.getStream());
hookEvent.response(mediaServerItem, jsonObject);
}
}
@ -982,6 +982,21 @@ public class InviteRequestProcessor extends SIPRequestProcessorParent implements
}
return;
}
String addressStr = sdp.getOrigin().getAddress();
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 = audioBroadcastCatch.getMediaServerItem();
if (mediaServerItem == null) {
logger.warn("未找到语音喊话使用的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;
}
String addressStr = sdp.getConnection().getAddress();
logger.info("设备{}请求语音流， 收流地址：{}:{}，ssrc：{}, {}, 对讲方式：{}", requesterId, addressStr, port, ssrc,
mediaTransmissionTCP ? (tcpActive? "TCP主动":"TCP被动") : "UDP", sdp.getSessionName().getValue());

2
src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/info/InfoRequestProcessor.java
View File

@ -102,7 +102,7 @@ public class InfoRequestProcessor extends SIPRequestProcessorParent implements I
String contentSubType = header.getContentSubType();
if ("Application".equalsIgnoreCase(contentType) && "MANSRTSP".equalsIgnoreCase(contentSubType)) {
SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(null, null, null, callIdHeader.getCallId());
String streamId = sendRtpItem.getStreamId();
String streamId = sendRtpItem.getStream();
StreamInfo streamInfo = redisCatchStorage.queryPlayback(null, null, streamId, null);
if (null == streamInfo) {
responseAck(request, Response.NOT_FOUND, "stream " + streamId + " not found");

2
src/main/java/com/genersoft/iot/vmp/gb28181/transmit/event/request/impl/message/notify/cmd/MediaStatusNotifyMessageHandler.java
View File

@ -90,7 +90,7 @@ public class MediaStatusNotifyMessageHandler extends SIPRequestProcessorParent i
try {
cmder.streamByeCmd(device, ssrcTransaction.getChannelId(), null, callIdHeader.getCallId());
} catch (InvalidArgumentException | ParseException | SsrcTransactionNotFoundException | SipException e) {
} catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
logger.error("[录像流]推送完毕，收到关流通知， 发送BYE失败 {}", e.getMessage());
}

2
src/main/java/com/genersoft/iot/vmp/gb28181/utils/SipUtils.java
View File

@ -123,7 +123,7 @@ public class SipUtils {
}
public static String getNewCallId() {
return (int) Math.floor(Math.random() * 10000) + "";
return (int) Math.floor(Math.random() * 1000000000) + "";
}
public static int getTypeCodeFromGbCode(String deviceId) {

121
src/main/java/com/genersoft/iot/vmp/media/zlm/ZLMHttpHookListener.java
View File

@ -9,9 +9,9 @@ import com.genersoft.iot.vmp.gb28181.bean.*;
import com.genersoft.iot.vmp.gb28181.event.EventPublisher;
import com.genersoft.iot.vmp.gb28181.session.AudioBroadcastManager;
import com.genersoft.iot.vmp.gb28181.session.VideoStreamSessionManager;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.ISIPCommanderForPlatform;
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.ISIPCommanderForPlatform;
import com.genersoft.iot.vmp.gb28181.transmit.cmd.impl.SIPCommander;
import com.genersoft.iot.vmp.media.zlm.dto.HookType;
import com.genersoft.iot.vmp.media.zlm.dto.MediaServerItem;
@ -249,6 +249,7 @@ public class ZLMHttpHookListener {
String channelId = ssrcTransactionForAll.get(0).getChannelId();
DeviceChannel deviceChannel = storager.queryChannel(deviceId, channelId);
if (deviceChannel != null) {
result.setEnable_audio(deviceChannel.isHasAudio());
}
// 如果是录像下载就设置视频间隔十秒
@ -257,6 +258,11 @@ public class ZLMHttpHookListener {
result.setEnable_audio(true);
result.setEnable_mp4(true);
}
// 如果是talk对讲，则默认获取声音
if (ssrcTransactionForAll.get(0).getType() == VideoStreamSessionManager.SessionType.talk) {
result.setEnable_audio(true);
}
}
return result;
}
@ -359,62 +365,30 @@ public class ZLMHttpHookListener {
}
}else if ("talk".equals(param.getApp())){
// 语音对讲推流 stream需要满足格式deviceId_channelId
if (param.isRegist() && param.getStream().indexOf("_") > 0) {
String[] streamArray = param.getStream().split("_");
if (streamArray.length == 2) {
String deviceId = streamArray[0];
String channelId = streamArray[1];
Device device = deviceService.getDevice(deviceId);
if (device != null) {
DeviceChannel deviceChannel = storager.queryChannel(deviceId, channelId);
if (deviceChannel != null) {
if (audioBroadcastManager.exit(deviceId, channelId)) {
// 直接推流
SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(null, null, param.getStream(), null);
if (sendRtpItem == null) {
// TODO 可能数据错误，重新开启语音通道
}else {
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__");
sendParam.put("app",sendRtpItem.getApp());
sendParam.put("stream",sendRtpItem.getStreamId());
sendParam.put("ssrc", sendRtpItem.getSsrc());
sendParam.put("src_port", sendRtpItem.getLocalPort());
sendParam.put("pt", sendRtpItem.getPt());
sendParam.put("use_ps", sendRtpItem.isUsePs() ? "1" : "0");
sendParam.put("only_audio", sendRtpItem.isOnlyAudio() ? "1" : "0");
JSONObject jsonObject;
if (sendRtpItem.isTcpActive()) {
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(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);
}
}
}else {
// 开启语音对讲通道
MediaServerItem mediaServerItem = mediaServerService.getOne(param.getMediaServerId());
playService.talk(mediaServerItem, device, channelId, (mediaServer, jsonObject)->{
System.out.println("开始推流");
}, eventResult -> {
System.out.println(eventResult.msg);
}, ()->{
System.out.println("超时");
});
}
}
}
}
}
if (param.getStream().indexOf("_") > 0) {
String[] streamArray = param.getStream().split("_");
if (streamArray.length == 2) {
String deviceId = streamArray[0];
String channelId = streamArray[1];
Device device = deviceService.getDevice(deviceId);
if (device != null) {
if (param.isRegist()) {
if (audioBroadcastManager.exit(deviceId, channelId)) {
playService.stopAudioBroadcast(deviceId, channelId);
}
// 开启语音对讲通道
playService.talkCmd(device, channelId, mediaInfo, param.getStream(), (msg)->{
logger.info("[语音对讲] 通道建立成功, device: {}, channel: {}", deviceId, channelId);
});
}else {
// 流注销
playService.stopTalk(device, channelId, param.isRegist());
}
} else{
logger.info("[语音对讲] 未找到设备：{}", deviceId);
}
}
}
}else{
if (!"rtp".equals(param.getApp())){
@ -474,16 +448,21 @@ public class ZLMHttpHookListener {
ParentPlatform platform = storager.queryParentPlatByServerGBId(platformId);
Device device = deviceService.getDevice(platformId);
try {
if (platform != null) {
commanderFroPlatform.streamByeCmd(platform, sendRtpItem);
try {
commanderFroPlatform.streamByeCmd(platform, sendRtpItem);
} catch (SipException | InvalidArgumentException | ParseException e) {
logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage());
}
} else {
cmder.streamByeCmd(device, sendRtpItem.getChannelId(), param.getStream(), sendRtpItem.getCallId());
try {
cmder.streamByeCmd(device, sendRtpItem.getChannelId(), param.getStream(), sendRtpItem.getCallId());
} catch (SipException | InvalidArgumentException | ParseException |
SsrcTransactionNotFoundException e) {
logger.error("[命令发送失败] 发送BYE: {}", e.getMessage());
}
}
} catch (SipException | InvalidArgumentException | ParseException |
SsrcTransactionNotFoundException e) {
logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage());
}
}
}
}
@ -526,7 +505,7 @@ public class ZLMHttpHookListener {
logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage());
}
redisCatchStorage.deleteSendRTPServer(parentPlatform.getServerGBId(), sendRtpItem.getChannelId(),
sendRtpItem.getCallId(), sendRtpItem.getStreamId());
sendRtpItem.getCallId(), sendRtpItem.getStream());
}
}
}
@ -555,8 +534,7 @@ public class ZLMHttpHookListener {
try {
cmder.streamByeCmd(device, streamInfoForPlayBackCatch.getChannelId(),
streamInfoForPlayBackCatch.getStream(), null);
} catch (InvalidArgumentException | ParseException | SipException |
SsrcTransactionNotFoundException e) {
} catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
logger.error("[无人观看]回放， 发送BYE失败 {}", e.getMessage());
}
}
@ -572,6 +550,13 @@ public class ZLMHttpHookListener {
ret.put("close", false);
return ret;
}
SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(null, null, param.getStream(), null);
if ("talk".equals(sendRtpItem.getApp())){
ret.put("close", false);
return ret;
}
}else if ("talk".equals(param.getApp()) || "broadcast".equals(param.getApp())){
ret.put("close", false);
} else {
// 非国标流 推流/拉流代理
// 拉流代理
@ -733,7 +718,7 @@ public class ZLMHttpHookListener {
logger.error("[命令发送失败] 国标级联 发送BYE: {}", e.getMessage());
}
redisCatchStorage.deleteSendRTPServer(parentPlatform.getServerGBId(), sendRtpItem.getChannelId(),
sendRtpItem.getCallId(), sendRtpItem.getStreamId());
sendRtpItem.getCallId(), sendRtpItem.getStream());
}
}
});

4
src/main/java/com/genersoft/iot/vmp/media/zlm/ZLMRESTfulUtils.java
View File

@ -291,6 +291,10 @@ public class ZLMRESTfulUtils {
return sendPost(mediaServerItem, "startSendRtpPassive",param, null);
}
public JSONObject startSendRtpPassive(MediaServerItem mediaServerItem, Map<String, Object> param, RequestCallback callback) {
return sendPost(mediaServerItem, "startSendRtpPassive",param, callback);
}
public JSONObject stopSendRtp(MediaServerItem mediaServerItem, Map<String, Object> param) {
return sendPost(mediaServerItem, "stopSendRtp",param, null);
}

8
src/main/java/com/genersoft/iot/vmp/media/zlm/ZLMRTPServerFactory.java
View File

@ -229,7 +229,7 @@ public class ZLMRTPServerFactory {
sendRtpItem.setPort(port);
sendRtpItem.setSsrc(ssrc);
sendRtpItem.setApp(app);
sendRtpItem.setStreamId(stream);
sendRtpItem.setStream(stream);
sendRtpItem.setPlatformId(platformId);
sendRtpItem.setChannelId(channelId);
sendRtpItem.setTcp(tcp);
@ -290,6 +290,10 @@ public class ZLMRTPServerFactory {
return zlmresTfulUtils.startSendRtpPassive(mediaServerItem, param);
}
public JSONObject startSendRtpPassive(MediaServerItem mediaServerItem, Map<String, Object>param, ZLMRESTfulUtils.RequestCallback callback) {
return zlmresTfulUtils.startSendRtpPassive(mediaServerItem, param, callback);
}
/**
* 查询待转推的流是否就绪
*/
@ -343,7 +347,7 @@ public class ZLMRTPServerFactory {
result= true;
logger.info("[停止RTP推流] 成功");
} else {
logger.error("[停止RTP推流] 失败: {}, 参数：{}->\r\n{}",jsonObject.getString("msg"), JSON.toJSON(param), jsonObject);
logger.warn("[停止RTP推流] 失败: {}, 参数：{}->\r\n{}",jsonObject.getString("msg"), JSON.toJSON(param), jsonObject);
}
return result;
}

11
src/main/java/com/genersoft/iot/vmp/service/IPlayService.java
View File

@ -12,6 +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.gb28181.play.bean.AudioEvent;
import gov.nist.javax.sip.message.SIPResponse;
import javax.sip.InvalidArgumentException;
@ -27,10 +28,6 @@ public interface IPlayService {
void onPublishHandlerForPlay(MediaServerItem mediaServerItem, JSONObject resonse, String deviceId, String channelId);
void talk(MediaServerItem mediaServerItem, Device device, String channelId,
ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
Runnable timeoutCallback);
void play(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo, Device device, String channelId,
ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
InviteTimeOutCallback timeoutCallback);
@ -57,7 +54,7 @@ public interface IPlayService {
void zlmServerOnline(String mediaServerId);
AudioBroadcastResult audioBroadcast(Device device, String channelId);
AudioBroadcastResult audioBroadcast(Device device, String channelId, Boolean broadcastMode);
void stopAudioBroadcast(String deviceId, String channelId);
void audioBroadcastCmd(Device device, String channelId, int timeout, MediaServerItem mediaServerItem, String sourceApp, String sourceStream, AudioBroadcastEvent event) throws InvalidArgumentException, ParseException, SipException;
@ -70,4 +67,8 @@ public interface IPlayService {
void startSendRtpStreamHand(SendRtpItem sendRtpItem, ParentPlatform parentPlatform,
JSONObject jsonObject, Map<String, Object> param, CallIdHeader callIdHeader);
void talkCmd(Device device, String channelId, MediaServerItem mediaServerItem, String stream, AudioEvent event);
void stopTalk(Device device, String channelId, Boolean streamIsReady);
}

2
src/main/java/com/genersoft/iot/vmp/service/impl/DeviceServiceImpl.java
View File

@ -202,7 +202,7 @@ public class DeviceServiceImpl implements IDeviceService {
Map<String, Object> param = new HashMap<>();
param.put("vhost", "__defaultVhost__");
param.put("app", sendRtpItem.getApp());
param.put("stream", sendRtpItem.getStreamId());
param.put("stream", sendRtpItem.getStream());
zlmresTfulUtils.stopSendRtp(mediaInfo, param);
}

2
src/main/java/com/genersoft/iot/vmp/service/impl/PlatformServiceImpl.java
View File

@ -253,7 +253,7 @@ public class PlatformServiceImpl implements IPlatformService {
Map<String, Object> param = new HashMap<>(3);
param.put("vhost", "__defaultVhost__");
param.put("app", sendRtpItem.getApp());
param.put("stream", sendRtpItem.getStreamId());
param.put("stream", sendRtpItem.getStream());
zlmrtpServerFactory.stopSendRtpStream(mediaInfo, param);
}
}

461
src/main/java/com/genersoft/iot/vmp/service/impl/PlayServiceImpl.java
View File

@ -41,7 +41,7 @@ import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
import com.genersoft.iot.vmp.vmanager.bean.ErrorCode;
import com.genersoft.iot.vmp.vmanager.bean.StreamContent;
import com.genersoft.iot.vmp.vmanager.bean.WVPResult;
import com.genersoft.iot.vmp.vmanager.gb28181.play.bean.AudioBroadcastEvent;
import com.genersoft.iot.vmp.vmanager.gb28181.play.bean.AudioEvent;
import gov.nist.javax.sip.message.SIPResponse;
import org.slf4j.Logger;
import org.slf4j.LoggerFactory;
@ -134,8 +134,8 @@ public class PlayServiceImpl implements IPlayService {
@Override
public void play(MediaServerItem mediaServerItem, String deviceId, String channelId,
ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
Runnable timeoutCallback) {
ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
Runnable timeoutCallback) {
if (mediaServerItem == null) {
throw new ControllerException(ErrorCode.ERROR100.getCode(), "未找到可用的zlm");
}
@ -243,194 +243,148 @@ public class PlayServiceImpl implements IPlayService {
}
}
@Override
public void talk(MediaServerItem mediaServerItem, Device device, String channelId,
ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
Runnable timeoutCallback) {
String streamId = null;
if (mediaServerItem.isRtpEnable()) {
streamId = String.format("%s_%s", device.getDeviceId(), channelId);
private void talk(MediaServerItem mediaServerItem, Device device, String channelId, String stream,
ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
Runnable timeoutCallback, AudioEvent audioEvent) {
String playSsrc = mediaServerItem.getSsrcConfig().getPlaySsrc();
if (playSsrc == null) {
audioEvent.call("ssrc已经用尽");
return;
}
SSRCInfo ssrcInfo = mediaServerService.openRTPServer(mediaServerItem, streamId, device.isSsrcCheck(), false);
logger.info("[对讲开始] deviceId: {}, channelId: {},收流端口： {}, 收流模式：{}, SSRC: {}, SSRC校验：{}", device.getDeviceId(), channelId, ssrcInfo.getPort(), device.getStreamMode(), ssrcInfo.getSsrc(), device.isSsrcCheck());
SendRtpItem sendRtpItem = new SendRtpItem();
sendRtpItem.setApp("talk");
sendRtpItem.setStream(stream);
sendRtpItem.setSsrc(playSsrc);
sendRtpItem.setDeviceId(device.getDeviceId());
sendRtpItem.setPlatformId(device.getDeviceId());
sendRtpItem.setChannelId(channelId);
sendRtpItem.setRtcp(false);
sendRtpItem.setMediaServerId(mediaServerItem.getId());
sendRtpItem.setOnlyAudio(true);
sendRtpItem.setPlayType(InviteStreamType.TALK);
sendRtpItem.setPt(8);
sendRtpItem.setStatus(1);
sendRtpItem.setTcpActive(false);
sendRtpItem.setTcp(true);
sendRtpItem.setUsePs(false);
sendRtpItem.setReceiveStream(stream + "_talk");
int port = zlmrtpServerFactory.keepPort(mediaServerItem, playSsrc);
//端口获取失败的ssrcInfo 没有必要发送点播指令
if (port <= 0) {
logger.info("[语音对讲] 端口分配异常，deviceId={},channelId={}", device.getDeviceId(), channelId);
audioEvent.call("端口分配异常");
return;
}
sendRtpItem.setLocalPort(port);
sendRtpItem.setPort(port);
logger.info("[语音对讲]开始 deviceId: {}, channelId: {},收流端口： {}, 收流模式：{}, SSRC: {}, SSRC校验：{}", device.getDeviceId(), channelId, sendRtpItem.getLocalPort(), device.getStreamMode(), sendRtpItem.getSsrc(), false);
// 超时处理
String timeOutTaskKey = UUID.randomUUID().toString();
SSRCInfo finalSsrcInfo = ssrcInfo;
System.out.println("设置超时任务： " + timeOutTaskKey);
dynamicTask.startDelay(timeOutTaskKey, () -> {
logger.info("[对讲超时] 收流超时 deviceId: {}, channelId: {}，端口：{}, SSRC: {}", device.getDeviceId(), channelId, finalSsrcInfo.getPort(), finalSsrcInfo.getSsrc());
logger.info("[语音对讲] 收流超时 deviceId: {}, channelId: {}，端口：{}, SSRC: {}", device.getDeviceId(), channelId, sendRtpItem.getPort(), sendRtpItem.getSsrc());
timeoutCallback.run();
// 点播超时回复BYE 同时释放ssrc以及此次点播的资源
try {
cmder.streamByeCmd(device, channelId, finalSsrcInfo.getStream(), null);
} catch (InvalidArgumentException | ParseException | SipException e) {
logger.error("[对讲超时]， 发送BYE失败 {}", e.getMessage());
} catch (SsrcTransactionNotFoundException e) {
cmder.streamByeCmd(device, channelId, sendRtpItem.getStream(), null);
} catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
logger.error("[语音对讲]超时， 发送BYE失败 {}", e.getMessage());
} finally {
timeoutCallback.run();
mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
}
}, userSetting.getPlayTimeout());
final String ssrc = ssrcInfo.getSsrc();
final String stream = ssrcInfo.getStream();
//端口获取失败的ssrcInfo 没有必要发送点播指令
if (ssrcInfo.getPort() <= 0) {
logger.info("[对讲] 端口分配异常，deviceId={},channelId={},ssrcInfo={}", device.getDeviceId(), channelId, ssrcInfo);
return;
}
String callId = SipUtils.getNewCallId();
boolean pushing = false;
zlmrtpServerFactory.releasePort(mediaServerItem, playSsrc);
Map<String, Object> param = new HashMap<>(12);
param.put("vhost","__defaultVhost__");
param.put("app", sendRtpItem.getApp());
param.put("stream", sendRtpItem.getStream());
param.put("ssrc", sendRtpItem.getSsrc());
param.put("src_port", sendRtpItem.getLocalPort());
param.put("pt", sendRtpItem.getPt());
param.put("use_ps", sendRtpItem.isUsePs() ? "1" : "0");
param.put("only_audio", sendRtpItem.isOnlyAudio() ? "1" : "0");
param.put("is_udp", sendRtpItem.isTcp() ? "0" : "1");
param.put("recv_stream_id", sendRtpItem.getReceiveStream());
param.put("close_delay_ms", userSetting.getPlayTimeout() * 1000);
zlmrtpServerFactory.startSendRtpPassive(mediaServerItem, param, jsonObject -> {
if (jsonObject == null || jsonObject.getInteger("code") != 0 ) {
mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
logger.info("[语音对讲]失败 deviceId: {}, channelId: {}", device.getDeviceId(), channelId);
audioEvent.call("失败, " + jsonObject.getString("msg"));
// 查看是否已经建立了通道，存在则发送bye
stopTalk(device, channelId);
}
});
// 查看设备是否已经在推流
// MediaItem mediaItem = zlmrtpServerFactory.getMediaInfo(mediaServerItem, "rtp",ssrcInfo.getStream());
// if (mediaItem != null) {
// SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem,
// mediaItem.getOriginSock().getPeer_ip(), mediaItem.getOriginSock().getPeer_port(), ssrcInfo.getSsrc(), device.getDeviceId(),
// device.getDeviceId(), channelId,
// false);
//
// sendRtpItem.setTcpActive(false);
// sendRtpItem.setCallId(callId);
// sendRtpItem.setPlayType(InviteStreamType.TALK);
// sendRtpItem.setStatus(1);
// sendRtpItem.setIp(mediaItem.getOriginSock().getPeer_ip());
// sendRtpItem.setPort(mediaItem.getOriginSock().getPeer_port());
// sendRtpItem.setTcpActive(false);
// sendRtpItem.setStreamId(ssrcInfo.getStream());
// sendRtpItem.setApp("1000");
// sendRtpItem.setStreamId("1000");
// sendRtpItem.setSsrc(ssrc);
// sendRtpItem.setOnlyAudio(true);
// redisCatchStorage.updateSendRTPSever(sendRtpItem);
//
// Map<String, Object> param = new HashMap<>(12);
// param.put("vhost","__defaultVhost__");
// param.put("app",sendRtpItem.getApp());
// param.put("stream",sendRtpItem.getStreamId());
// param.put("ssrc", sendRtpItem.getSsrc());
// param.put("dst_url", sendRtpItem.getIp());
// param.put("dst_port", sendRtpItem.getPort());
// param.put("src_port", sendRtpItem.getLocalPort());
// param.put("pt", sendRtpItem.getPt());
// param.put("use_ps", sendRtpItem.isUsePs() ? "1" : "0");
// param.put("is_udp", sendRtpItem.isTcp() ? "0" : "1");
// param.put("only_audio", sendRtpItem.isOnlyAudio() ? "1" : "0");
// JSONObject jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaServerItem, param);
// System.out.println(2222);
// System.out.println(jsonObject);
// }else {
try {
cmder.talkStreamCmd(mediaServerItem, ssrcInfo, device, channelId, callId, (MediaServerItem mediaServerItemInuse, JSONObject response) -> {
logger.info("[对讲] 流已生成， 开始推流： " + response.toJSONString());
dynamicTask.stop(timeOutTaskKey);
// TODO 暂不做处理
}, (MediaServerItem mediaServerItemInuse, JSONObject json) -> {
logger.info("[对讲] 设备开始推流： " + json.toJSONString());
dynamicTask.stop(timeOutTaskKey);
// 获取远程IP端口 作为回复语音流的地址
String ip = json.getString("ip");
Integer port = json.getInteger("port");
logger.info("[设备开始推流]{}/{}, 来自ip：{}, 端口：{}", device.getDeviceId(), channelId, ip, port);
// 查看平台推流是否就绪
// Boolean ready = zlmrtpServerFactory.isStreamReady(mediaServerItemInuse, "talk", stream);
// if (!ready) {
// try {
// cmder.streamByeCmd(device, channelId, finalSsrcInfo.getStream(), null);
// } catch (InvalidArgumentException | ParseException | SipException e) {
// logger.error("[对讲超时]， 发送BYE失败 {}", e.getMessage());
// } catch (SsrcTransactionNotFoundException e) {
// timeoutCallback.run();
// mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
// mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
// streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
// }
// }else {
// try {
// Thread.sleep(1000);
// } catch (InterruptedException e) {
// throw new RuntimeException(e);
// }
SendRtpItem sendRtpItem = zlmrtpServerFactory.createSendRtpItem(mediaServerItem, ip, port, ssrcInfo.getSsrc(), device.getDeviceId(),
device.getDeviceId(), channelId,
false, false);
try {
cmder.talkStreamCmd(mediaServerItem, sendRtpItem, device, channelId, callId, (MediaServerItem mediaServerItemInuse, JSONObject response) -> {
logger.info("[语音对讲] 流已生成， 开始推流： " + response.toJSONString());
dynamicTask.stop(timeOutTaskKey);
// TODO 暂不做处理
}, (MediaServerItem mediaServerItemInuse, JSONObject json) -> {
logger.info("[语音对讲] 设备开始推流： " + json.toJSONString());
dynamicTask.stop(timeOutTaskKey);
}, (event) -> {
dynamicTask.stop(timeOutTaskKey);
// if (sendRtpItem.getLocalPort() == 0) {
// logger.warn("服务器端口资源不足");
// try {
// cmder.streamByeCmd(device, channelId, finalSsrcInfo.getStream(), null);
// } catch (InvalidArgumentException | ParseException | SipException e) {
// logger.error("[对讲超时]， 发送BYE失败 {}", e.getMessage());
// } catch (SsrcTransactionNotFoundException e) {
// timeoutCallback.run();
// mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
// mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
// streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
// }
// return;
// }
sendRtpItem.setTcpActive(false);
sendRtpItem.setCallId(callId);
sendRtpItem.setPlayType(InviteStreamType.TALK);
sendRtpItem.setStatus(1);
sendRtpItem.setIp(ip);
sendRtpItem.setPort(port);
sendRtpItem.setTcpActive(false);
sendRtpItem.setApp("1000");
sendRtpItem.setStreamId("1000");
sendRtpItem.setSsrc(ssrc);
sendRtpItem.setOnlyAudio(true);
sendRtpItem.setRtcp(false);
if (event.event instanceof ResponseEvent) {
ResponseEvent responseEvent = (ResponseEvent) event.event;
if (responseEvent.getResponse() instanceof SIPResponse) {
SIPResponse response = (SIPResponse) responseEvent.getResponse();
sendRtpItem.setFromTag(response.getFromTag());
sendRtpItem.setToTag(response.getToTag());
sendRtpItem.setCallId(response.getCallIdHeader().getCallId());
redisCatchStorage.updateSendRTPSever(sendRtpItem);
Map<String, Object> param = new HashMap<>(12);
param.put("vhost","__defaultVhost__");
param.put("app",sendRtpItem.getApp());
param.put("stream",sendRtpItem.getStreamId());
param.put("ssrc", sendRtpItem.getSsrc());
param.put("dst_url", sendRtpItem.getIp());
param.put("dst_port", sendRtpItem.getPort());
param.put("src_port", sendRtpItem.getLocalPort());
param.put("pt", sendRtpItem.getPt());
param.put("use_ps", sendRtpItem.isUsePs() ? "1" : "0");
param.put("is_udp", sendRtpItem.isTcp() ? "0" : "1");
param.put("only_audio", sendRtpItem.isOnlyAudio() ? "1" : "0");
JSONObject jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaServerItemInuse, param);
System.out.println(11111);
System.out.println(sendRtpItem.getIp() + ":" + sendRtpItem.getPort());
// System.out.println(jsonObject);
// }
streamSession.put(device.getDeviceId(), channelId, "talk",
sendRtpItem.getStream(), sendRtpItem.getSsrc(), sendRtpItem.getMediaServerId(),
response, VideoStreamSessionManager.SessionType.talk);
} else {
logger.error("[语音对讲]收到的消息错误，response不是SIPResponse");
}
} else {
logger.error("[语音对讲]收到的消息错误，event不是ResponseEvent");
}
}, (event) -> {
}, (event) -> {
dynamicTask.stop(timeOutTaskKey);
mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
// 释放ssrc
mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
errorEvent.response(event);
});
} catch (InvalidArgumentException | SipException | ParseException e) {
logger.error("[命令发送失败] 对讲消息: {}", e.getMessage());
}, (event) -> {
dynamicTask.stop(timeOutTaskKey);
mediaServerService.closeRTPServer(mediaServerItem, finalSsrcInfo.getStream());
mediaServerService.closeRTPServer(mediaServerItem, sendRtpItem.getStream());
// 释放ssrc
mediaServerService.releaseSsrc(mediaServerItem.getId(), finalSsrcInfo.getSsrc());
mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
errorEvent.response(event);
});
} catch (InvalidArgumentException | SipException | ParseException e) {
streamSession.remove(device.getDeviceId(), channelId, finalSsrcInfo.getStream());
SipSubscribe.EventResult eventResult = new SipSubscribe.EventResult(new CmdSendFailEvent(null));
eventResult.msg = "命令发送失败";
errorEvent.response(eventResult);
}
logger.error("[命令发送失败] 对讲消息: {}", e.getMessage());
dynamicTask.stop(timeOutTaskKey);
mediaServerService.closeRTPServer(mediaServerItem, sendRtpItem.getStream());
// 释放ssrc
mediaServerService.releaseSsrc(mediaServerItem.getId(), sendRtpItem.getSsrc());
streamSession.remove(device.getDeviceId(), channelId, sendRtpItem.getStream());
SipSubscribe.EventResult eventResult = new SipSubscribe.EventResult(new CmdSendFailEvent(null));
eventResult.msg = "命令发送失败";
errorEvent.response(eventResult);
}
// }
}
@Override
public void play(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo, Device device, String channelId,
ZlmHttpHookSubscribe.Event hookEvent, SipSubscribe.Event errorEvent,
@ -446,7 +400,8 @@ public class PlayServiceImpl implements IPlayService {
// 点播超时回复BYE 同时释放ssrc以及此次点播的资源
try {
cmder.streamByeCmd(device, channelId, ssrcInfo.getStream(), null);
} catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
} catch (InvalidArgumentException | ParseException | SipException |
SsrcTransactionNotFoundException e) {
logger.error("[点播超时]， 发送BYE失败 {}", e.getMessage());
} finally {
timeoutCallback.run(1, "收流超时");
@ -483,7 +438,7 @@ public class PlayServiceImpl implements IPlayService {
onPublishHandlerForPlay(mediaServerItemInuse, response, device.getDeviceId(), channelId);
hookEvent.response(mediaServerItemInuse, response);
logger.info("[点播成功] deviceId: {}, channelId: {}", device.getDeviceId(), channelId);
String streamUrl = String.format("http://127.0.0.1:%s/%s/%s.live.flv", mediaServerItemInuse.getHttpPort(), "rtp", ssrcInfo.getStream());
String streamUrl = String.format("http://127.0.0.1:%s/%s/%s.live.flv", mediaServerItemInuse.getHttpPort(), "rtp", ssrcInfo.getStream());
String path = "snap";
String fileName = device.getDeviceId() + "_" + channelId + ".jpg";
// 请求截图
@ -652,8 +607,8 @@ public class PlayServiceImpl implements IPlayService {
@Override
public void playBack(String deviceId, String channelId, String startTime,
String endTime, InviteStreamCallback inviteStreamCallback,
PlayBackCallback callback) {
String endTime, InviteStreamCallback inviteStreamCallback,
PlayBackCallback callback) {
Device device = storager.queryVideoDevice(deviceId);
if (device == null) {
return;
@ -666,9 +621,9 @@ public class PlayServiceImpl implements IPlayService {
@Override
public void playBack(MediaServerItem mediaServerItem, SSRCInfo ssrcInfo,
String deviceId, String channelId, String startTime,
String endTime, InviteStreamCallback infoCallBack,
PlayBackCallback playBackCallback) {
String deviceId, String channelId, String startTime,
String endTime, InviteStreamCallback infoCallBack,
PlayBackCallback playBackCallback) {
if (mediaServerItem == null || ssrcInfo == null) {
return;
}
@ -792,7 +747,6 @@ public class PlayServiceImpl implements IPlayService {
}
@Override
public void download(String deviceId, String channelId, String startTime, String endTime, int downloadSpeed, InviteStreamCallback infoCallBack, PlayBackCallback playBackCallback) {
Device device = storager.queryVideoDevice(deviceId);
@ -977,7 +931,7 @@ public class PlayServiceImpl implements IPlayService {
cmder.streamByeCmd(device, ssrcTransaction.getChannelId(),
ssrcTransaction.getStream(), null);
} catch (InvalidArgumentException | ParseException | SipException |
SsrcTransactionNotFoundException e) {
SsrcTransactionNotFoundException e) {
logger.error("[zlm离线]为正在使用此zlm的设备， 发送BYE失败 {}", e.getMessage());
}
}
@ -986,7 +940,8 @@ public class PlayServiceImpl implements IPlayService {
}
@Override
public AudioBroadcastResult audioBroadcast(Device device, String channelId) {
public AudioBroadcastResult audioBroadcast(Device device, String channelId, Boolean broadcastMode) {
// TODO 必须多端口模式才支持语音喊话鹤语音对讲
if (device == null || channelId == null) {
return null;
}
@ -997,15 +952,15 @@ public class PlayServiceImpl implements IPlayService {
return null;
}
MediaServerItem mediaServerItem = mediaServerService.getMediaServerForMinimumLoad(null);
String app = "broadcast";
// TODO 从sip user agent中判断是什么品牌设备，大华默认使用talk模式，其他使用broadcast模式
// String app = "talk";
if (broadcastMode == null) {
broadcastMode = true;
}
String app = broadcastMode?"broadcast":"talk";
String stream = device.getDeviceId() + "_" + channelId;
StreamInfo broadcast = mediaService.getStreamInfoByAppAndStream(mediaServerItem, "broadcast", stream, null, null, null, false);
AudioBroadcastResult audioBroadcastResult = new AudioBroadcastResult();
audioBroadcastResult.setApp(app);
audioBroadcastResult.setStream(stream);
audioBroadcastResult.setStreamInfo(new StreamContent(mediaService.getStreamInfoByAppAndStream(mediaServerItem, app, stream, null, null, null,false)));
audioBroadcastResult.setStreamInfo(new StreamContent(mediaService.getStreamInfoByAppAndStream(mediaServerItem, app, stream, null, null, null, false)));
audioBroadcastResult.setCodec("G.711");
return audioBroadcastResult;
}
@ -1037,6 +992,18 @@ public class PlayServiceImpl implements IPlayService {
}
}
}
SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, null, null);
if (sendRtpItem != null) {
MediaServerItem mediaServer = mediaServerService.getOne(sendRtpItem.getMediaServerId());
Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServer, "rtp", sendRtpItem.getReceiveStream());
if (streamReady) {
logger.warn("[语音对讲] 进行中： {}", channelId);
event.call("语音对讲进行中");
return;
} else {
stopTalk(device, channelId);
}
}
// 发送通知
cmder.audioBroadcastCmd(device, channelId, eventResultForOk -> {
@ -1053,19 +1020,18 @@ public class PlayServiceImpl implements IPlayService {
}
@Override
public void stopAudioBroadcast(String deviceId, String channelId) {
List<AudioBroadcastCatch> audioBroadcastCatchList = new ArrayList<>();
if (channelId == null) {
audioBroadcastCatchList.addAll(audioBroadcastManager.get(deviceId));
}else {
} else {
audioBroadcastCatchList.add(audioBroadcastManager.get(deviceId, channelId));
}
if (audioBroadcastCatchList.size() > 0) {
for (AudioBroadcastCatch audioBroadcastCatch : audioBroadcastCatchList) {
Device device = deviceService.getDevice(deviceId);
if (device == null || audioBroadcastCatch == null ) {
if (device == null || audioBroadcastCatch == null) {
return;
}
SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(deviceId, audioBroadcastCatch.getChannelId(), null, null);
@ -1075,7 +1041,7 @@ public class PlayServiceImpl implements IPlayService {
Map<String, Object> param = new HashMap<>();
param.put("vhost", "__defaultVhost__");
param.put("app", sendRtpItem.getApp());
param.put("stream", sendRtpItem.getStreamId());
param.put("stream", sendRtpItem.getStream());
zlmresTfulUtils.stopSendRtp(mediaInfo, param);
try {
cmder.streamByeCmd(device, sendRtpItem.getChannelId(), audioBroadcastCatch.getSipTransactionInfo(), null);
@ -1199,12 +1165,12 @@ public class PlayServiceImpl implements IPlayService {
String is_Udp = sendRtpItem.isTcp() ? "0" : "1";
MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
logger.info("收到ACK，rtp/{}开始向上级推流, 目标={}:{}，SSRC={}, RTCP={}", sendRtpItem.getStreamId(),
logger.info("收到ACK，rtp/{}开始向上级推流, 目标={}:{}，SSRC={}, RTCP={}", sendRtpItem.getStream(),
sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isRtcp());
Map<String, Object> param = new HashMap<>(12);
param.put("vhost","__defaultVhost__");
param.put("app",sendRtpItem.getApp());
param.put("stream",sendRtpItem.getStreamId());
param.put("vhost", "__defaultVhost__");
param.put("app", sendRtpItem.getApp());
param.put("stream", sendRtpItem.getStream());
param.put("ssrc", sendRtpItem.getSsrc());
param.put("src_port", sendRtpItem.getLocalPort());
param.put("pt", sendRtpItem.getPt());
@ -1213,12 +1179,12 @@ public class PlayServiceImpl implements IPlayService {
param.put("is_udp", is_Udp);
if (!sendRtpItem.isTcp()) {
// udp模式下开启rtcp保活
param.put("udp_rtcp_timeout", sendRtpItem.isRtcp()? "1":"0");
param.put("udp_rtcp_timeout", sendRtpItem.isRtcp() ? "1" : "0");
}
if (mediaInfo == null) {
RequestPushStreamMsg requestPushStreamMsg = RequestPushStreamMsg.getInstance(
sendRtpItem.getMediaServerId(), sendRtpItem.getApp(), sendRtpItem.getStreamId(),
sendRtpItem.getMediaServerId(), sendRtpItem.getApp(), sendRtpItem.getStream(),
sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isTcp(),
sendRtpItem.getLocalPort(), sendRtpItem.getPt(), sendRtpItem.isUsePs(), sendRtpItem.isOnlyAudio());
redisGbPlayMsgListener.sendMsgForStartSendRtpStream(sendRtpItem.getServerId(), requestPushStreamMsg, json -> {
@ -1233,16 +1199,16 @@ public class PlayServiceImpl implements IPlayService {
if (zlmrtpServerFactory.releasePort(mediaInfo, sendRtpItem.getSsrc())) {
if (sendRtpItem.isTcpActive()) {
startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);
}else {
} else {
param.put("dst_url", sendRtpItem.getIp());
param.put("dst_port", sendRtpItem.getPort());
startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);
}
}
}else {
} else {
if (sendRtpItem.isTcpActive()) {
startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, param);
}else {
} else {
param.put("dst_url", sendRtpItem.getIp());
param.put("dst_port", sendRtpItem.getPort());
startSendRtpStreamResult = zlmrtpServerFactory.startSendRtpStream(mediaInfo, param);
@ -1260,10 +1226,10 @@ public class PlayServiceImpl implements IPlayService {
if (jsonObject == null) {
logger.error("RTP推流失败: 请检查ZLM服务");
} else if (jsonObject.getInteger("code") == 0) {
logger.info("调用ZLM推流接口, 结果： {}", jsonObject);
logger.info("RTP推流成功[ {}/{} ]，{}->{}:{}, " ,param.get("app"), param.get("stream"), jsonObject.getString("local_port"), param.get("dst_url"), param.get("dst_port"));
logger.info("调用ZLM推流接口, 结果： {}", jsonObject);
logger.info("RTP推流成功[ {}/{} ]，{}->{}:{}, ", param.get("app"), param.get("stream"), jsonObject.getString("local_port"), param.get("dst_url"), param.get("dst_port"));
} else {
logger.error("RTP推流失败: {}, 参数：{}",jsonObject.getString("msg"), JSON.toJSONString(param));
logger.error("RTP推流失败: {}, 参数：{}", jsonObject.getString("msg"), JSON.toJSONString(param));
if (sendRtpItem.isOnlyAudio()) {
Device device = deviceService.getDevice(sendRtpItem.getDeviceId());
AudioBroadcastCatch audioBroadcastCatch = audioBroadcastManager.get(sendRtpItem.getDeviceId(), sendRtpItem.getChannelId());
@ -1275,7 +1241,7 @@ public class PlayServiceImpl implements IPlayService {
logger.error("[命令发送失败] 停止语音对讲: {}", e.getMessage());
}
}
}else {
} else {
// 向上级平台
try {
commanderForPlatform.streamByeCmd(parentPlatform, callIdHeader.getCallId());
@ -1285,4 +1251,105 @@ public class PlayServiceImpl implements IPlayService {
}
}
}
@Override
public void talkCmd(Device device, String channelId, MediaServerItem mediaServerItem, String stream, AudioEvent event) {
if (device == null || channelId == null) {
return;
}
// TODO 必须多端口模式才支持语音喊话鹤语音对讲
logger.info("[语音对讲] device： {}, channel: {}", device.getDeviceId(), channelId);
DeviceChannel deviceChannel = storager.queryChannel(device.getDeviceId(), channelId);
if (deviceChannel == null) {
logger.warn("开启语音对讲的时候未找到通道： {}", channelId);
event.call("开启语音对讲的时候未找到通道");
return;
}
// 查询通道使用状态
if (audioBroadcastManager.exit(device.getDeviceId(), channelId)) {
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 mediaServer = mediaServerService.getOne(sendRtpItem.getMediaServerId());
Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServer, sendRtpItem.getApp(), sendRtpItem.getStream());
if (streamReady) {
logger.warn("[语音对讲] 正在语音广播，无法开启语音通话： {}", channelId);
event.call("正在语音广播");
return;
} else {
stopAudioBroadcast(device.getDeviceId(), channelId);
}
}
}
SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, stream, null);
if (sendRtpItem != null) {
MediaServerItem mediaServer = mediaServerService.getOne(sendRtpItem.getMediaServerId());
Boolean streamReady = zlmrtpServerFactory.isStreamReady(mediaServer, "rtp", sendRtpItem.getReceiveStream());
if (streamReady) {
logger.warn("[语音对讲] 进行中： {}", channelId);
event.call("语音对讲进行中");
return;
} else {
stopTalk(device, channelId);
}
}
talk(mediaServerItem, device, channelId, stream, (MediaServerItem mediaServerItem1, JSONObject response) -> {
logger.info("[语音对讲] 收到设备发来的流");
}, eventResult -> {
logger.warn("[语音对讲] 失败，{}/{}, 错误码 {} {}", device.getDeviceId(), channelId, eventResult.statusCode, eventResult.msg);
event.call("失败，错误码 " + eventResult.statusCode + ", " + eventResult.msg);
}, () -> {
logger.warn("[语音对讲] 失败，{}/{} 超时", device.getDeviceId(), channelId);
event.call("失败，超时 ");
stopTalk(device, channelId);
}, errorMsg -> {
logger.warn("[语音对讲] 失败，{}/{} {}", device.getDeviceId(), channelId, errorMsg);
event.call(errorMsg);
stopTalk(device, channelId);
});
}
private void stopTalk(Device device, String channelId) {
stopTalk(device, channelId, null);
}
@Override
public void stopTalk(Device device, String channelId, Boolean streamIsReady) {
logger.info("[语音对讲] 停止， {}/{}", device.getDeviceId(), channelId);
SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(device.getDeviceId(), channelId, null, null);
if (sendRtpItem == null) {
logger.info("[语音对讲] 停止失败， 未找到发送信息，可能已经停止");
return;
}
// 停止向设备推流
String mediaServerId = sendRtpItem.getMediaServerId();
if (mediaServerId == null) {
return;
}
MediaServerItem mediaServer = mediaServerService.getOne(mediaServerId);
if (streamIsReady == null || streamIsReady) {
Map<String, Object> param = new HashMap<>();
param.put("vhost", "__defaultVhost__");
param.put("app", sendRtpItem.getApp());
param.put("stream", sendRtpItem.getStream());
param.put("ssrc", sendRtpItem.getSsrc());
zlmrtpServerFactory.stopSendRtpStream(mediaServer, param);
}
mediaServer.getSsrcConfig().releaseSsrc(sendRtpItem.getSsrc());
SsrcTransaction ssrcTransaction = streamSession.getSsrcTransaction(device.getDeviceId(), channelId, null, sendRtpItem.getStream());
if (ssrcTransaction != null) {
try {
cmder.streamByeCmd(device, channelId, sendRtpItem.getStream(), null);
} catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
logger.info("[语音对讲] 停止消息发送失败，可能已经停止");
}
}
redisCatchStorage.deleteSendRTPServer(device.getDeviceId(), channelId,null, null);
}
}

2
src/main/java/com/genersoft/iot/vmp/storager/impl/RedisCatchStorageImpl.java
View File

@ -378,7 +378,7 @@ public class RedisCatchStorageImpl implements IRedisCatchStorage {
+ sendRtpItem.getMediaServerId() + "_"
+ sendRtpItem.getPlatformId() + "_"
+ sendRtpItem.getChannelId() + "_"
+ sendRtpItem.getStreamId() + "_"
+ sendRtpItem.getStream() + "_"
+ sendRtpItem.getCallId();
RedisUtil.set(key, sendRtpItem);
}

20
src/main/java/com/genersoft/iot/vmp/vmanager/gb28181/play/PlayController.java
View File

@ -19,11 +19,7 @@ import com.genersoft.iot.vmp.service.IMediaService;
import com.genersoft.iot.vmp.service.IPlayService;
import com.genersoft.iot.vmp.storager.IRedisCatchStorage;
import com.genersoft.iot.vmp.storager.IVideoManagerStorage;
import com.genersoft.iot.vmp.vmanager.bean.DeferredResultEx;
import com.genersoft.iot.vmp.vmanager.bean.AudioBroadcastResult;
import com.genersoft.iot.vmp.vmanager.bean.ErrorCode;
import com.genersoft.iot.vmp.vmanager.bean.StreamContent;
import com.genersoft.iot.vmp.vmanager.bean.WVPResult;
import com.genersoft.iot.vmp.vmanager.bean.*;
import io.swagger.v3.oas.annotations.Operation;
import io.swagger.v3.oas.annotations.Parameter;
import io.swagger.v3.oas.annotations.tags.Tag;
@ -253,7 +249,7 @@ public class PlayController {
@Parameter(name = "timeout", description = "推流超时时间(秒)", required = true)
@GetMapping("/broadcast/{deviceId}/{channelId}")
@PostMapping("/broadcast/{deviceId}/{channelId}")
public AudioBroadcastResult broadcastApi(@PathVariable String deviceId, @PathVariable String channelId, Integer timeout) {
public AudioBroadcastResult broadcastApi(@PathVariable String deviceId, @PathVariable String channelId, Integer timeout, Boolean broadcastMode) {
if (logger.isDebugEnabled()) {
logger.debug("语音广播API调用");
}
@ -265,15 +261,7 @@ public class PlayController {
throw new ControllerException(ErrorCode.ERROR400.getCode(), "未找到通道： " + channelId);
}
return playService.audioBroadcast(device, channelId);
}
@GetMapping("/1111")
public void broadcastApi1() {
MediaServerItem defaultMediaServer = mediaServerService.getMediaServerForMinimumLoad(null);
Device device = storager.queryVideoDevice("34020000001320090001");
playService.talk(defaultMediaServer, device, "34020000001370000001", null, null, null);
return playService.audioBroadcast(device, channelId, broadcastMode);
}
@ -289,7 +277,7 @@ public class PlayController {
}
// try {
// playService.stopAudioBroadcast(deviceId, channelId);
// } catch (InvalidArgumentException | ParseException | SsrcTransactionNotFoundException | SipException e) {
// } catch (InvalidArgumentException | ParseException | SipException e) {
// logger.error("[命令发送失败] 停止语音: {}", e.getMessage());
// throw new ControllerException(ErrorCode.ERROR100.getCode(), "命令发送失败: " + e.getMessage());
// }

2
src/main/java/com/genersoft/iot/vmp/vmanager/gb28181/play/bean/AudioBroadcastEvent.java → src/main/java/com/genersoft/iot/vmp/vmanager/gb28181/play/bean/AudioEvent.java
View File

@ -4,6 +4,6 @@ package com.genersoft.iot.vmp.vmanager.gb28181.play.bean;
/**
* @author lin
*/
public interface AudioBroadcastEvent {
public interface AudioEvent {
void call(String msg);
}

2
src/main/java/com/genersoft/iot/vmp/web/gb28181/ApiStreamController.java
View File

@ -185,7 +185,7 @@ public class ApiStreamController {
}
try {
cmder.streamByeCmd(device, code, streamInfo.getStream(), null);
} catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
} catch (InvalidArgumentException | ParseException | SipException | SsrcTransactionNotFoundException e) {
JSONObject result = new JSONObject();
result.put("error","发送BYE失败：" + e.getMessage());
return result;

2
web_src/config/index.js
View File

@ -12,7 +12,7 @@ module.exports = {
assetsPublicPath: './',
proxyTable: {
'/debug': {
target: 'https://default.wvp-pro.cn:18080',
target: 'https://default.wvp-pro.cn:18082',
changeOrigin: true,
pathRewrite: {
'^/debug': '/'

7
web_src/src/components/dialog/devicePlayer.vue
View File

@ -299,6 +299,10 @@
</el-tab-pane>
<el-tab-pane label="语音对讲" name="broadcast">
<div style="padding: 0 10px">
<el-switch v-model="broadcastMode" :disabled="broadcastStatus !== -1" active-color="#409EFF" active-text="喊话"
inactive-text="对讲"></el-switch>
</div>
<div class="trank" style="text-align: center;">
<el-button @click="broadcastStatusClick()" :type="getBroadcastStatus()" :disabled="broadcastStatus === -2"
circle icon="el-icon-microphone" style="font-size: 32px; padding: 24px;margin-top: 24px;"/>
@ -390,6 +394,7 @@ export default {
recordStartTime: 0,
showTimeText: "00:00:00",
streamInfo: null,
broadcastMode: true,
broadcastRtc: null,
broadcastStatus: -1, // -2 正在释放资源 -1 默认状态 0 等待接通 1 接通成功
};
@ -648,7 +653,7 @@ export default {
// 发起语音对讲
this.$axios({
method: 'get',
url: '/api/play/broadcast/' + this.deviceId + '/' + this.channelId + "?timeout=30"
url: '/api/play/broadcast/' + this.deviceId + '/' + this.channelId + "?timeout=30&broadcastMode=" + this.broadcastMode
}).then( (res)=> {
if (res.data.code == 0) {
let streamInfo = res.data.data.streamInfo;