优化合并对讲broadcast级联模式
parent
4362a5b499
commit
6b1d966255
2
pom.xml
2
pom.xml
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@ -6,7 +6,7 @@
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<parent>
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<groupId>org.springframework.boot</groupId>
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<artifactId>spring-boot-starter-parent</artifactId>
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<version>2.7.2</version>
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<version>2.7.9</version>
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</parent>
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<groupId>com.genersoft</groupId>
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@ -790,11 +790,11 @@ public class SIPCommanderFroPlatform implements ISIPCommanderForPlatform {
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content.append("t=0 0\r\n");
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if ("TCP-PASSIVE".equalsIgnoreCase(userSetting.getBroadcastForPlatform())) {
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content.append("m=video " + ssrcInfo.getPort() + " TCP/RTP/AVP 8 96\r\n");
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content.append("m=audio " + ssrcInfo.getPort() + " TCP/RTP/AVP 8 96\r\n");
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} else if ("TCP-ACTIVE".equalsIgnoreCase(userSetting.getBroadcastForPlatform())) {
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content.append("m=video " + ssrcInfo.getPort() + " TCP/RTP/AVP 8 96\r\n");
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content.append("m=audio " + ssrcInfo.getPort() + " TCP/RTP/AVP 8 96\r\n");
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} else if ("UDP".equalsIgnoreCase(userSetting.getBroadcastForPlatform())) {
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content.append("m=video " + ssrcInfo.getPort() + " RTP/AVP 8 96\r\n");
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content.append("m=audio " + ssrcInfo.getPort() + " RTP/AVP 8 96\r\n");
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}
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content.append("a=recvonly\r\n");
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@ -817,12 +817,12 @@ public class SIPCommanderFroPlatform implements ISIPCommanderForPlatform {
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sipSender.transmitRequest(sipLayer.getLocalIp(platform.getDeviceIp()), request, (e -> {
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streamSession.remove(platform.getServerGBId(), channelId, ssrcInfo.getStream());
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mediaServerService.releaseSsrc(mediaServerItem.getId(), ssrcInfo.getSsrc());
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subscribe.removeSubscribe(hookSubscribe);
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errorEvent.response(e);
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}), e -> {
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// 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
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ResponseEvent responseEvent = (ResponseEvent) e.event;
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SIPResponse response = (SIPResponse) responseEvent.getResponse();
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streamSession.put(platform.getServerGBId(), channelId, callIdHeader.getCallId(), stream, ssrcInfo.getSsrc(), mediaServerItem.getId(), response, VideoStreamSessionManager.SessionType.play);
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streamSession.put(platform.getServerGBId(), channelId, callIdHeader.getCallId(), stream, ssrcInfo.getSsrc(), mediaServerItem.getId(), response, VideoStreamSessionManager.SessionType.broadcast);
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okEvent.response(e);
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});
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}
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@ -102,7 +102,7 @@ public class AckRequestProcessor extends SIPRequestProcessorParent implements In
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}
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String isUdp = sendRtpItem.isTcp() ? "0" : "1";
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MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
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logger.info("收到ACK,rtp/{}开始向上级推流, 目标={}:{},SSRC={}, RTCP={}", sendRtpItem.getStream(),
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logger.info("收到ACK,rtp/{}开始级推流, 目标={}:{},SSRC={}, RTCP={}", sendRtpItem.getStream(),
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sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isRtcp());
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Map<String, Object> param = new HashMap<>(12);
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param.put("vhost","__defaultVhost__");
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@ -2,7 +2,10 @@ package com.genersoft.iot.vmp.gb28181.transmit.event.request.impl;
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import com.genersoft.iot.vmp.common.StreamInfo;
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import com.genersoft.iot.vmp.conf.exception.SsrcTransactionNotFoundException;
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import com.genersoft.iot.vmp.gb28181.bean.*;
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import com.genersoft.iot.vmp.gb28181.bean.Device;
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import com.genersoft.iot.vmp.gb28181.bean.InviteStreamType;
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import com.genersoft.iot.vmp.gb28181.bean.SendRtpItem;
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import com.genersoft.iot.vmp.gb28181.bean.SsrcTransaction;
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import com.genersoft.iot.vmp.gb28181.session.AudioBroadcastManager;
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import com.genersoft.iot.vmp.gb28181.session.VideoStreamSessionManager;
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import com.genersoft.iot.vmp.gb28181.transmit.SIPProcessorObserver;
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@ -24,9 +27,10 @@ import org.springframework.beans.factory.InitializingBean;
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import org.springframework.beans.factory.annotation.Autowired;
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import org.springframework.stereotype.Component;
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import javax.sip.*;
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import javax.sip.InvalidArgumentException;
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import javax.sip.RequestEvent;
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import javax.sip.SipException;
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import javax.sip.address.SipURI;
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import javax.sip.header.CallIdHeader;
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import javax.sip.header.FromHeader;
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import javax.sip.header.HeaderAddress;
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import javax.sip.header.ToHeader;
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@ -87,13 +91,15 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
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@Override
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public void process(RequestEvent evt) {
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// TODO 此处需要重构
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SIPRequest request =(SIPRequest) evt.getRequest();
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try {
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responseAck((SIPRequest) evt.getRequest(), Response.OK);
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responseAck(request, Response.OK);
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} catch (SipException | InvalidArgumentException | ParseException e) {
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logger.error("[回复BYE信息失败],{}", e.getMessage());
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}
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CallIdHeader callIdHeader = (CallIdHeader)evt.getRequest().getHeader(CallIdHeader.NAME);
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SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(null, null, null, callIdHeader.getCallId());
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SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(null, null, null, request.getCallIdHeader().getCallId());
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if (sendRtpItem != null){
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logger.info("[收到bye] {}/{}", sendRtpItem.getPlatformId(), sendRtpItem.getChannelId());
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@ -115,7 +121,7 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
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param.put("ssrc",sendRtpItem.getSsrc());
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logger.info("[收到bye] 停止推流:{}", streamId);
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MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
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redisCatchStorage.deleteSendRTPServer(sendRtpItem.getPlatformId(), sendRtpItem.getChannelId(), callIdHeader.getCallId(), null);
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redisCatchStorage.deleteSendRTPServer(sendRtpItem.getPlatformId(), sendRtpItem.getChannelId(), request.getCallIdHeader().getCallId(), null);
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zlmrtpServerFactory.stopSendRtpStream(mediaInfo, param);
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int totalReaderCount = zlmrtpServerFactory.totalReaderCount(mediaInfo, sendRtpItem.getApp(), streamId);
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@ -159,7 +165,7 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
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}
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SsrcTransaction ssrcTransactionForPlay = streamSession.getSsrcTransaction(device.getDeviceId(), channelId, "play", null);
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if (ssrcTransactionForPlay != null){
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if (ssrcTransactionForPlay.getCallId().equals(callIdHeader.getCallId())){
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if (ssrcTransactionForPlay.getCallId().equals(request.getCallIdHeader().getCallId())){
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// 释放ssrc
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MediaServerItem mediaServerItem = mediaServerService.getOne(ssrcTransactionForPlay.getMediaServerId());
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if (mediaServerItem != null) {
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@ -168,7 +174,7 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
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streamSession.remove(device.getDeviceId(), channelId, ssrcTransactionForPlay.getStream());
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}
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}
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SsrcTransaction ssrcTransactionForPlayBack = streamSession.getSsrcTransaction(device.getDeviceId(), channelId, callIdHeader.getCallId(), null);
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SsrcTransaction ssrcTransactionForPlayBack = streamSession.getSsrcTransaction(device.getDeviceId(), channelId, request.getCallIdHeader().getCallId(), null);
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if (ssrcTransactionForPlayBack != null) {
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// 释放ssrc
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MediaServerItem mediaServerItem = mediaServerService.getOne(ssrcTransactionForPlayBack.getMediaServerId());
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@ -178,5 +184,32 @@ public class ByeRequestProcessor extends SIPRequestProcessorParent implements In
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streamSession.remove(device.getDeviceId(), channelId, ssrcTransactionForPlayBack.getStream());
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}
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}
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SsrcTransaction ssrcTransaction = streamSession.getSsrcTransaction(null, null, request.getCallIdHeader().getCallId(), null);
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if (ssrcTransaction != null) {
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// 释放ssrc
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MediaServerItem mediaServerItem = mediaServerService.getOne(ssrcTransaction.getMediaServerId());
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if (mediaServerItem != null) {
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mediaServerService.releaseSsrc(mediaServerItem.getId(), ssrcTransaction.getSsrc());
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}
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switch (ssrcTransaction.getType()) {
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// case play:
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// break;
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// case talk:
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// break;
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// case playback:
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// break;
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// case download:
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// break;
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case broadcast:
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String deviceId = ssrcTransaction.getDeviceId();
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String channelId1 = ssrcTransaction.getChannelId();
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// 如果是
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break;
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default:
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break;
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}
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streamSession.remove(device.getDeviceId(), channelId, ssrcTransaction.getStream());
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}
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}
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}
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@ -102,6 +102,7 @@ public class BroadcastNotifyMessageHandler extends SIPRequestProcessorParent imp
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DeviceChannel deviceChannel = storage.queryChannelInParentPlatform(platform.getServerGBId(), targetId);
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if (deviceChannel == null) {
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logger.warn("[国标级联 语音喊话] 未找到通道 platform: {}, channel: {}", platform.getServerGBId(), targetId);
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responseAck(request, Response.NOT_FOUND, "TargetID not found");
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return;
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}
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@ -123,6 +124,7 @@ public class BroadcastNotifyMessageHandler extends SIPRequestProcessorParent imp
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commanderForPlatform.broadcastResultCmd(platform, deviceChannel, sn, true, eventResult->{
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logger.info("[国标级联] 语音喊话 回复失败 platform: {}, 错误:{}/{}", platform.getServerGBId(), eventResult.statusCode, eventResult.msg);
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}, eventResult->{
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// 消息发送成功, 向上级发送invite,获取推流
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try {
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platformService.broadcastInvite(platform, deviceChannel.getChannelId(), mediaServerForMinimumLoad, (mediaServerItem, response)->{
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@ -132,7 +134,7 @@ public class BroadcastNotifyMessageHandler extends SIPRequestProcessorParent imp
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AudioBroadcastCatch broadcastCatch = audioBroadcastManager.get(device.getDeviceId(), targetId);
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if (broadcastCatch != null ) {
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if (playService.audioBroadcastInUse(device, targetId)) {
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logger.info("[国标级联] 语音喊话 设备正正在使用中 platform: {}, channel: {}",
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logger.info("[国标级联] 语音喊话 设备正在使用中 platform: {}, channel: {}",
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platform.getServerGBId(), deviceChannel.getChannelId());
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// 查看语音通道已经建立且已经占用 回复BYE
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try {
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@ -259,11 +259,11 @@ public class ZLMRTPServerFactory {
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// 订阅 zlm启动事件, 新的zlm也会从这里进入系统
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hookSubscribe.addSubscribe(hookSubscribeForRtpServerTimeout,
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(MediaServerItem mediaServerItem, JSONObject response)->{
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logger.info("[上级点播] {}->监听端口到期继续保持监听", ssrc);
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logger.info("[保持端口] {}->监听端口到期继续保持监听", ssrc);
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keepPort(serverItem, ssrc);
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});
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}
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logger.info("[上级点播] {}->监听端口: {}", ssrc, localPort);
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logger.info("[保持端口] {}->监听端口: {}", ssrc, localPort);
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return localPort;
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}
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@ -271,7 +271,7 @@ public class ZLMRTPServerFactory {
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* 释放保持的端口
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*/
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public boolean releasePort(MediaServerItem serverItem, String ssrc) {
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logger.info("[上级点播] {}->释放监听端口", ssrc);
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logger.info("[保持端口] {}->释放监听端口", ssrc);
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boolean closeRTPServerResult = closeRtpServer(serverItem, ssrc);
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HookSubscribeForRtpServerTimeout hookSubscribeForRtpServerTimeout = HookSubscribeFactory.on_rtp_server_timeout(ssrc, null, serverItem.getId());
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// 订阅 zlm启动事件, 新的zlm也会从这里进入系统
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@ -357,7 +357,7 @@ public class ZLMRTPServerFactory {
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public JSONObject startSendRtp(MediaServerItem mediaInfo, SendRtpItem sendRtpItem) {
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String is_Udp = sendRtpItem.isTcp() ? "0" : "1";
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logger.info("rtp/{}开始向上级推流, 目标={}:{},SSRC={}", sendRtpItem.getStream(), sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc());
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logger.info("rtp/{}开始推流, 目标={}:{},SSRC={}", sendRtpItem.getStream(), sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc());
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Map<String, Object> param = new HashMap<>(12);
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param.put("vhost","__defaultVhost__");
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param.put("app",sendRtpItem.getApp());
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@ -373,7 +373,7 @@ public class PlatformServiceImpl implements IPlatformService {
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errorEvent.response(new SipSubscribe.EventResult(-1, "端口监听失败"));
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return;
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}
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logger.info("[国标级联] 发起语音喊话 deviceId: {}, channelId: {},收流端口: {}, 收流模式:{}, SSRC: {}, SSRC校验:{}",
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logger.info("[国标级联] 语音喊话,发起Invite消息 deviceId: {}, channelId: {},收流端口: {}, 收流模式:{}, SSRC: {}, SSRC校验:{}",
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platform.getServerGBId(), channelId, ssrcInfo.getPort(), userSetting.getBroadcastForPlatform(), ssrcInfo.getSsrc(), ssrcCheck);
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String timeOutTaskKey = UUID.randomUUID().toString();
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@ -396,6 +396,7 @@ public class PlatformServiceImpl implements IPlatformService {
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}
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}, userSetting.getPlayTimeout());
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commanderForPlatform.broadcastInviteCmd(platform, channelId, mediaServerItem, ssrcInfo, (mediaServerItemForInvite, response)->{
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logger.info("[国标级联] 发起语音喊话 收到上级推流 deviceId: {}, channelId: {}", platform.getServerGBId(), channelId);
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dynamicTask.stop(timeOutTaskKey);
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// hook响应
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playService.onPublishHandlerForPlay(mediaServerItemForInvite, response, platform.getServerGBId(), channelId);
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@ -1183,7 +1183,7 @@ public class PlayServiceImpl implements IPlayService {
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String is_Udp = sendRtpItem.isTcp() ? "0" : "1";
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MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
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logger.info("收到ACK,rtp/{}开始向上级推流, 目标={}:{},SSRC={}, RTCP={}", sendRtpItem.getStream(),
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logger.info("收到ACK,rtp/{}开始推流, 目标={}:{},SSRC={}, RTCP={}", sendRtpItem.getStream(),
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sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc(), sendRtpItem.isRtcp());
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Map<String, Object> param = new HashMap<>(12);
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param.put("vhost", "__defaultVhost__");
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@ -181,7 +181,7 @@
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style="font-size: 1.875rem;"></i></div>
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<div style="position: absolute; left: 7.25rem; top: 3.25rem; font-size: 1.875rem;"
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@mousedown="ptzCamera('zoomout')" @mouseup="ptzCamera('stop')"><i
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class="el-icon-zoom-out control-zoom-btn"></i></div>
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class="el-icon-zoom-out control-zoom-btn"></i></div>
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<div class="contro-speed" style="position: absolute; left: 4px; top: 7rem; width: 9rem;">
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<el-slider v-model="controSpeed" :max="255"></el-slider>
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</div>
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@ -300,7 +300,8 @@
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</el-tab-pane>
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<el-tab-pane label="语音对讲" name="broadcast">
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<div style="padding: 0 10px">
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<el-switch v-model="broadcastMode" :disabled="broadcastStatus !== -1" active-color="#409EFF" active-text="喊话"
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<el-switch v-model="broadcastMode" :disabled="broadcastStatus !== -1" active-color="#409EFF"
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active-text="喊话"
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inactive-text="对讲"></el-switch>
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</div>
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<div class="trank" style="text-align: center;">
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@ -565,20 +566,20 @@ export default {
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this.tracks = []
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let _this = this;
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this.$copyText(data).then(
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function (e) {
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_this.$message({
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showClose: true,
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message: '复制成功',
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type: 'success'
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});
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},
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function (e) {
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_this.$message({
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showClose: true,
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message: '复制失败,请手动复制',
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type: 'error'
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});
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}
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function (e) {
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_this.$message({
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showClose: true,
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message: '复制成功',
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type: 'success'
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});
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},
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function (e) {
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_this.$message({
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showClose: true,
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message: '复制失败,请手动复制',
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type: 'error'
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});
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}
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);
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},
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ptzCamera: function (command) {
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@ -654,55 +655,54 @@ export default {
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this.$axios({
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method: 'get',
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url: '/api/play/broadcast/' + this.deviceId + '/' + this.channelId + "?timeout=30&broadcastMode=" + this.broadcastMode
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}).then( (res)=> {
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}).then((res) => {
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if (res.data.code === 0) {
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let streamInfo = res.data.data.streamInfo;
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if (document.location.protocol.includes("https")) {
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this.startBroadcast(streamInfo.rtcs)
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}else {
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} else {
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this.startBroadcast(streamInfo.rtc)
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}
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}else {
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this.$message({
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showClose: true,
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message: res.data.msg,
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type: "error",
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});
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}
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});
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}else if (this.broadcastStatus === 1) {
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this.broadcastStatus = -1;
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this.broadcastRtc.close()
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}
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},
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startBroadcast(url){
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// 获取推流鉴权Key
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this.$axios({
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method: 'post',
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url: '/api/user/userInfo',
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}).then( (res)=> {
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if (res.data.code !== 0) {
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this.$message({
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showClose: true,
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message: "获取推流鉴权Key失败",
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type: "error",
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});
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this.broadcastStatus = -1;
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}else {
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let pushKey = res.data.data.pushKey;
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// 获取推流鉴权KEY
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url += "&sign=" + crypto.createHash('md5').update(pushKey, "utf8").digest('hex')
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console.log("开始语音喊话: " + url)
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this.broadcastRtc = new ZLMRTCClient.Endpoint({
|
||||
debug: true, // 是否打印日志
|
||||
zlmsdpUrl: url, //流地址
|
||||
simulecast: false,
|
||||
useCamera: false,
|
||||
audioEnable: true,
|
||||
videoEnable: false,
|
||||
recvOnly: false,
|
||||
})
|
||||
} else {
|
||||
this.$message({
|
||||
showClose: true,
|
||||
message: res.data.msg,
|
||||
type: "error",
|
||||
});
|
||||
}
|
||||
});
|
||||
} else if (this.broadcastStatus === 1) {
|
||||
this.broadcastStatus = -1;
|
||||
this.broadcastRtc.close()
|
||||
}
|
||||
},
|
||||
startBroadcast(url) {
|
||||
// 获取推流鉴权Key
|
||||
this.$axios({
|
||||
method: 'post',
|
||||
url: '/api/user/userInfo',
|
||||
}).then((res) => {
|
||||
if (res.data.code !== 0) {
|
||||
this.$message({
|
||||
showClose: true,
|
||||
message: "获取推流鉴权Key失败",
|
||||
type: "error",
|
||||
});
|
||||
this.broadcastStatus = -1;
|
||||
} else {
|
||||
let pushKey = res.data.data.pushKey;
|
||||
// 获取推流鉴权KEY
|
||||
url += "&sign=" + crypto.createHash('md5').update(pushKey, "utf8").digest('hex')
|
||||
console.log("开始语音喊话: " + url)
|
||||
this.broadcastRtc = new ZLMRTCClient.Endpoint({
|
||||
debug: true, // 是否打印日志
|
||||
zlmsdpUrl: url, //流地址
|
||||
simulecast: false,
|
||||
useCamera: false,
|
||||
audioEnable: true,
|
||||
videoEnable: false,
|
||||
recvOnly: false,
|
||||
})
|
||||
|
||||
// webrtcPlayer.on(ZLMRTCClient.Events.WEBRTC_ON_REMOTE_STREAMS,(e)=>{//获取到了远端流,可以播放
|
||||
// console.error('播放成功',e.streams)
|
||||
|
@ -715,15 +715,15 @@ export default {
|
|||
// // this.eventcallbacK("LOCAL STREAM", "获取到了本地流")
|
||||
// });
|
||||
|
||||
this.broadcastRtc.on(ZLMRTCClient.Events.WEBRTC_NOT_SUPPORT,(e)=>{// 获取到了本地流
|
||||
console.error('不支持webrtc',e)
|
||||
this.$message({
|
||||
showClose: true,
|
||||
message: '不支持webrtc, 无法进行语音喊话',
|
||||
type: 'error'
|
||||
});
|
||||
this.broadcastStatus = -1;
|
||||
});
|
||||
this.broadcastRtc.on(ZLMRTCClient.Events.WEBRTC_NOT_SUPPORT, (e) => {// 获取到了本地流
|
||||
console.error('不支持webrtc', e)
|
||||
this.$message({
|
||||
showClose: true,
|
||||
message: '不支持webrtc, 无法进行语音喊话',
|
||||
type: 'error'
|
||||
});
|
||||
this.broadcastStatus = -1;
|
||||
});
|
||||
|
||||
this.broadcastRtc.on(ZLMRTCClient.Events.WEBRTC_ICE_CANDIDATE_ERROR, (e) => {// ICE 协商出错
|
||||
console.error('ICE 协商出错')
|
||||
|
|
Loading…
Reference in New Issue