uniapp开发WebRTC语音直播间支持app（android+IOS）和H5，并记录了所有踩得坑_uniapp webrtc

作者：Cpp五条 | 2024-05-15 09:59:00

踩

uniapp webrtc

一、效果图

二、主要功能

1. 创建自己的语音直播间

2. 查询所有直播间列表

3.加入房间

4.申请上位

5.麦克风控制

6.声音控制

7.赠送礼物（特效 + 批量移动动画）

8.退出房间

三、原理

1.uniapp 实现客户端H5、安卓、苹果

2.webRTC实现语音直播间（具体原理网上有很多文章我就不讲了，贴个图）

3.使用node.js搭建信令服务器（我用的是socket）

4.礼物及特效使用svga

四、踩坑及解决方案

1. 客户端（这里重点在于app端）一定要在视图层创建webRTC！！！不要在逻辑层创建！！！因为会要求使用安全连接，也就是说要用到SSL证书，这个很多人都没有，有的话当我没说。如何在视图层创建RTC呢？在uniapp中使用renderjs！


<script module="webRTC" lang="renderjs">
new RTCPeerConnection(iceServers)
</script>

2. （这里重点也在于app）客户端创建和信令服务器进行通信的socket时app端在页面跳转后socket状态消失无法响应信令服务器消息。解决方案是：一定不要在客户端视图层创建socket！！！也就是说socket不要创建在renderjs里，要在逻辑层用uniapp提供的api进行创建，然后使用uniapp文档中说明的逻辑层和视图层的通信方式进行通信，这样虽然在开发中有些繁琐，但是能解决问题。


onShow(){
// socketTask是使用uniapp提供的uni.connectSocket创建出来的socket实例
// watchSocketMessage代理了socket实例的onMessage方法
socketTask.watchSocketMessage = (data) => {
				this.watchSocketMessage(data)
			}
    
}
 
methed:{
    watchSocketMessage(){
        // 这里是收到信令服务器socket后的逻辑
    }
}


// 这里是逻辑层和renderjs通信的方式，通过监听状态的改变从而触发renderjs的对应的方法
// 注意在页面刚加载完成后这些方法会被默认触发一边，所以要在这些放方法做好判断return出去
 
<view :rid="rid" :change:rid="webRTC.initRid" :userId="userId" :change:userId="webRTC.initUserId"
			:giftnum="giftnum" :change:giftnum="webRTC.initgiftnum" :micPosition="micPosition"
			:change:micPosition="webRTC.initMicPositions" :giftPosition="giftPosition"
			:change:giftPosition="webRTC.initGiftPosition" :RTCJoin="RTCJoin" :change:RTCJoin="webRTC.changeRTCjoin"
			:RTCOffier="RTCOffier" :change:RTCOffier="webRTC.changeRTCoffier" :RTCAnswer="RTCAnswer" :isAudio="isAudio"
			:change:isAudio="webRTC.changeIsAudio" :change:RTCAnswer="webRTC.changeRTCAnswer"
			:RTCCandidate="RTCCandidate" :change:RTCCandidate="webRTC.changeRTCCandidate" :isTrue="isTrue"
			:change:isTrue="webRTC.changeIsTrue" :newMess="newMess" :change:newMess="webRTC.changeNewMessage"
			:isMedia="isMedia" :name="name" :change:name="webRTC.changeName" :change:isMedia="webRTC.changeIsMedia"
			:animos="animos" :change:animos="changeAnimos" class="chat">
</view>

3.连接顺序的问题，一定是：新进入的用户通过信令服务器给房间已有用户发送Offer，用户接收到Offer回应Answer，记住这个逻辑！

4.因为webRTC是运行在视图层的（也就是浏览器），而苹果默认浏览器是Safari，Safari浏览器默认机制是在用户主动和页面进行交互后，自动播放声音才会生效（也就是才有声音），所以在IOS端所有用户进入直播房间后默认都是静音的，用户主动开启音频才会受到直播间的声音（这是目前我发现的最好的解决办法）

五、核心代码（只有关键步骤）

1. 客户端socket


const socketTask = {
	socket: null,
	connect: () => {
 
		getApp().globalData.socket = uni.connectSocket({
			url:'ws://180.76.158.110:9000/socket/websocketv',
			// url: 'ws://192.168.3.254:9000/socket/websocketv',
			complete: (e) => {
				console.log(e);
			},
		});
 
		getApp().globalData.socket.onOpen((data) => {
			console.log("111111111");
			getApp().globalData.socket.send({
				data: JSON.stringify({
					type: "newConnect",
					userId: uni.getStorageSync('user').id,
				})
			})
		})
 
		getApp().globalData.socket.onClose((res) => {
			console.log("连接关闭", res);
			getApp().globalData.socket = null;
			setTimeout(() => {
				socketTask.connect()
			}, 3000)
		})
 
		getApp().globalData.socket.onError((err) => {
			console.log("连接异常", err);
			getApp().globalData.socket = null;
			setTimeout(() => {
				socketTask.connect()
			}, 1)
		})
 
		getApp().globalData.socket.onMessage((data) => {
			socketTask.watchSocketMessage(data)
		})
 
	},
	start: function() {
		this.connect()
	},
	watchSocketMessage: function() {
		// 这里实现自己的业务逻辑
	}
}
 
export default socketTask

2.客户端房间列表页


async onShow() {
			if (!getApp().globalData.socket) {
				await socketTask.start();
			}
			socketTask.watchSocketMessage = (data) => {
				console.log("===========收到新消息==========",data);
				this.watchSocketMessages(data)
			}
		},
methed:{
// 监听socket消息
			watchSocketMessages(res) {
				try {
					const socket_msg = JSON.parse(res.data);
					console.log("收到新消息", socket_msg);
					switch (socket_msg.type) {
						case "homeList":
							if (socket_msg.data.length == 0) {
								this.homeList = [];
								uni.showToast({
									title: "暂无房间，快去创建一个吧",
									icon: "none"
								})
							} else {
								this.homeList = socket_msg.data;
							}
							break
						case "leave":
							getApp().globalData.socket.send({
								data: JSON.stringify({
									type: "homeList",
									userId: this.userInfo.userId,
								})
							})
							break
						case "createSuccess":
							uni.redirectTo({
								url: `broadRoom?rid=${socket_msg.data.groupId}&&userId=${this.userInfo.id}&&groupInfo=${JSON.stringify(socket_msg.data)}`
							})
							break
					}
				} catch (e) {
 
				}
			},
}

3.客户端直播间

逻辑层：


async onShow() {
			const that = this;
			if (!getApp().globalData.socket) {
				console.log("socket不存在，重新连接");
				await socketTask.start();
			}
			socketTask.watchSocketMessage = (data) => {
				this.watchSocketMessage(data)
			}
			// 编译平台信息
			uni.getSystemInfo({
				success(res) {
					console.log("当前平台是", res);
					if (res.osName == 'ios') {
						console.log("我是ios", res)
						that.isMedia = 'ios';
					} else {
						console.log("我是安卓", res)
						that.isMedia = 'android';
					}
 
				}
			})
 
		}
 
methed：{
async watchSocketMessage(date) {
				const data = JSON.parse(date.data);
				switch (data.type) {
					case "join":
						console.log("join成功", data);
						this.newMessaGes(data);
						this.setUserList(data.admin);
						this.updataNewMic(data)
						// 找出自己以外的其他用户
						const arr = this.userList.filter((item, index) => {
							return item.userId !== this.userId
						})
						console.log("找出自己以外的其他用户", arr)
						// 通知renderjs层创建RTC
						this.RTCJoin = arr;
						this.updataIsShow()
						break
 
					case "newjoin":
						this.newMessaGes(data);
						this.setUserList(data.admin);
						break
 
					case "offer":
						//通知renderjs层有新人进入创建answer
						console.log("收到offer", data)
						this.RTCOffier = data;
						break
					case "answer":
						// 找到对应peer,设置answer
						console.log("收到offer", data)
						this.RTCAnswer = data;
						break
					case "candidate":
						// 找到对应的peer,将candidate添加进去
						this.RTCCandidate = data;
						break
					case "leave":
						if (data.data == "房主已解散房间") {
							this.closesAdmin()
						} else {
							const datas = {
								data,
							}
							this.newMessaGes(datas)
							this.setUserList(data.admin);
							this.updataNewMic(data);
						}
						break
					case "apply-admin":
						this.updataIsApply(data.data)
						break
					case "newMic":
						this.updataNewMic(data)
						break
					case "uplMicro":
						this.updataNewMic(data)
						break
					case "newMessage":
						this.newMess = data;
						break
				}
			},
}

视图层：


<script module="webRTC" lang="renderjs">
 
// 以下方法都在methed：{}中
 
 
 
// 监听changeRTCCandidate
			async changeRTCCandidate(data) {
				if (!data) {
					return
				}
				console.log("this.otherPeerConnections", this.otherPeerConnections);
				let arrs = this.otherPeerConnections.concat(this.myPeerConnections);
 
 
				if (arrs.length == 0) {
					return
				}
 
				let peerr = arrs.filter(item => {
					return item.otherId == data.userId
				})
				
				if (peerr[0].peer == {}) {
					return
				} else {
					console.log("candidatecandidate", data.candidate)
					await peerr[0].peer.addIceCandidate(new RTCIceCandidate(data.candidate))
				}
			},
			// 监听answer,找到对应peer设置answer
			async changeRTCAnswer(data) {
				if (!data) {
					return
				}
				let peers = this.myPeerConnections.filter(item => {
					return item.otherId == data.userId
				})
				console.log("peers[0]", peers[0])
				await peers[0].peer.setRemoteDescription(new RTCSessionDescription(data.answer))
			},
			// 监听offier,RTCAnswer的创建
			async changeRTCoffier(data) {
				if (!data) {
					return
				}
				let pear = null;
				try {
					pear = new RTCPeerConnection(iceServers);
				} catch (e) {
					console.log("实例化RTC-pear失败", e);
				}
 
				// 将音频流加入到Peer中
				this.localStream.getAudioTracks()[0].enabled = this.isTrue;
				this.localStream.getTracks().forEach(
					(track) => pear.addTrack(track, this.localStream)
				);
				this.otherPeerConnections.push({
					peer: pear,
					otherId: data.userId
				})
				//当远程用户向对等连接添加流时，我们将显示它
				pear.ontrack = (event) => {
					// 为该用户创建audio
					const track = event.track || event.streams[0]?.getTracks()[0];
					if (track && track.kind === 'audio') {
						console.log("存在音轨", event.streams[0]);
						this.renderAudio(data.userId, event.streams[0]);
					} else {
						console.warn("No audio track found in the received stream.");
					}
				};
 
				// 通过监听onicecandidate事件获取candidate信息
				pear.onicecandidate = async (event) => {
					if (event.candidate) {
						// 通过信令服务器发送candidate信息给用户B
						await this.$ownerInstance.callMethod("sendCandidate", {
							type: "candidate",
							userId: this.userId,
							rid: this.rid,
							msg: event.candidate,
							formUserId: data.userId,
						})
					}
				}
 
				pear.setRemoteDescription(new RTCSessionDescription(data.offer))
 
				// 接收端创建answer并发送给发起端
				pear.createAnswer().then(answer => {
					pear.setLocalDescription(answer);
					// 通知serve层给房间用户发送answer
					this.$ownerInstance.callMethod("sendAnswer", {
						type: "answer",
						userId: this.userId,
						rid: this.rid,
						msg: answer,
						formUserId: data.userId,
					})
				})
			},
			// 发起连接申请,offier的创建
			changeRTCjoin(RTCjoin) {
				if (!RTCjoin) {
					return
				}
				RTCjoin.forEach((item, index) => {
					let peer = null;
					try {
						peer = new RTCPeerConnection(iceServers);
					} catch (e) {
						console.log("实例化RTC失败", e);
					}
 
					this.localStream.getAudioTracks()[0].enabled = this.isTrue;
					this.localStream.getTracks().forEach(
						(track) => peer.addTrack(track, this.localStream)
					);
 
					peer.ontrack = (event) => {
						console.log("发起连接申请,offier的创建:peer.ontrack");
						const track = event.track || event.streams[0]?.getTracks()[0];
						if (track && track.kind === 'audio') {
							console.log("存在音轨2", event.streams[0]);
							this.renderAudio(item.userId, event.streams[0]);
						} else {
							console.warn("No audio track found in the received stream.");
						}
					};
 
					// 通过监听onicecandidate事件获取candidate信息
					peer.onicecandidate = (event) => {
						if (event.candidate) {
							// 通过信令服务器发送candidate信息给用户B
							this.$ownerInstance.callMethod("sendCandidate", {
								type: "candidate",
								userId: this.userId,
								rid: this.rid,
								msg: event.candidate,
								formUserId: item.userId,
							})
						}
					}
					this.myPeerConnections.push({
						peer: peer,
						otherId: item.userId
					})
 
					peer.createOffer(this.offerOptions).then(offer => {
						peer.setLocalDescription(offer);
						// 通知serve层给房间用户发送offier
						this.$ownerInstance.callMethod("sendOffier", {
							type: "offer",
							userId: this.userId,
							rid: this.rid,
							msg: offer,
							formUserId: item.userId,
						})
					})
				})
			},
 
			renderAudio(uid, stream) {
				let audio2 = document.getElementById(`audio_${uid}`);
				console.log("audio_name", `audio_${uid}`);
				if (!audio2) {
					audio2 = document.createElement('audio');
					audio2.id = `audio_${uid}`;
					audio2.setAttribute("webkit-playsinline", "");
					audio2.setAttribute("autoplay", true);
					audio2.setAttribute("playsinline", "");
					audio2.onloadedmetadata = () => {
						if (this.isAudio == 1) {
							console.log("不自动播放");
							audio2.pause();
						} else {
							audio2.play();
						}
					};
 
					this.audioList.push(audio2)
				}
				if ("srcObject" in audio2) {
					console.log("使用了srcObject赋值");
					audio2.srcObject = stream;
				} else {
					console.log("找不到srcObject赋值");
					audio2.src = window.URL.createObjectURL(stream);
				}
			},
async initMedia() {
				const that = this;
 
				console.log("##########", this.isMedia);
				// #ifdef APP-PLUS
				if (this.isMedia == 'android') {
					console.log("androidandroidandroidandroid");
					await plus.android.requestPermissions(
						['android.permission.RECORD_AUDIO'],
						async (resultObj) => {
								var result = 0;
								for (var i = 0; i < resultObj.granted.length; i++) {
									var grantedPermission = resultObj.granted[i];
									result = 1
								}
								for (var i = 0; i < resultObj.deniedPresent.length; i++) {
									var deniedPresentPermission = resultObj.deniedPresent[i];
									result = 0
								}
								for (var i = 0; i < resultObj.deniedAlways.length; i++) {
									var deniedAlwaysPermission = resultObj.deniedAlways[i];
									result = -1
								}
								that.localStream = await that.getUserMedia();
								that.$ownerInstance.callMethod("sendJoin", {
									type: "join",
									userId: that.userId,
									rid: that.rid,
									name: that.name
								})
							},
							function(error) {
								console.log("导入android出现错误", error);
							}
					);
				} else {
					console.log("iosiosiosiosiosios");
					that.localStream = await that.getUserMedia().catch(err => {
						console.log("出错了", err);
					})
					that.$ownerInstance.callMethod("sendJoin", {
						type: "join",
						userId: that.userId,
						rid: that.rid,
						name: that.name
					})
				}
 
				// #endif
				// #ifdef H5
				that.localStream = await that.getUserMedia();
				// 通知serve层加入成功
				this.$ownerInstance.callMethod("sendJoin", {
					type: "join",
					userId: this.userId,
					rid: this.rid,
					name: this.name
				})
				// #endif
			},
			getUserMedia(then) {
				return new Promise((resolve, reject) => {
					navigator.mediaDevices.getUserMedia(this.mediaConstraints).then((stream) => {
						return resolve(stream);
					}).catch(err => {
						if (err.name === 'NotAllowedError' || err.name === 'PermissionDeniedError') {
							// 用户拒绝了授权
							reject(new Error('用户拒绝了访问摄像头和麦克风的请求'));
						} else if (err.name === 'NotFoundError' || err.name === 'DevicesNotFoundError') {
							// 没有找到摄像头或麦克风
							reject(new Error('没有找到摄像头或麦克风'));
						} else if (err.name === 'NotReadableError' || err.name === 'TrackStartError') {
							// 摄像头或麦克风不可读
							reject(new Error('摄像头或麦克风不可读'));
						} else if (err.name === 'OverconstrainedError' || err.name ===
							'ConstraintNotSatisfiedError') {
							// 由于媒体流的约束条件无法满足，请求被拒绝
							reject(new Error('请求被拒绝，由于媒体流的约束条件无法满足'));
						} else if (err.name === 'TypeError' || err.name === 'TypeError') {
							// 发生了类型错误
							reject(new Error('发生了类型错误'));
						} else {
							// 其他未知错误
							reject(new Error('发生了未知错误'));
						}
					})
				});
			},
</script>

4.信令服务器

略（就是socket，里面写swich，不会私信，小额收费）

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