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trtc-calling-js-roomid

v1.0.0

Published

腾讯云 TRTC CALLING SDK

Downloads

2

Readme

腾讯云实时音视频通话 TRTC CALLING SDK

TRTCCalling 组件,是我们基于腾讯云 Web 版的 TRTC SDK信令(TSignalling) SDK 组合而成的一个功能组件,用于支持双人和多人场景下的音视频通话通能。

接入方式

从v0.6.0起,需要手动安装依赖 trtc-js-sdktim-js-sdk 以及 tsignaling

  • 为了减小 trtc-calling-js.js 的体积,避免和接入侧已使用的 trtc-js-sdk 和 tim-js-sdk 以及 tsignaling 发生版本冲突,trtc-js-sdk 和 tim-js-sdk 以及 tsignaling 不再被打包到 trtc-calling-js.js,在使用前您需要手动安装依赖。
  npm i trtc-js-sdk --save
  npm i tim-js-sdk --save
  npm i tsignaling --save
  npm i trtc-calling-js --save
  // 如果是通过node下载的依赖,则使用 import 引入
  import TRTCCalling from 'trtc-calling-js';
 
  // 如果您通过 script 方式使用 trtc-calling-js,需要按顺序
  // 手动引入 trtc.js
  <script src="./trtc.js"></script>
  // 接着手动引入 tim-js.js
  <script src="./tim-js.js"></script>
  // 然后再手动引入 tsignaling.js
  <script src="./tsignaling.js"></script>
  // 最后再手动引入 trtc-calling-js.js
  <script src="./trtc-calling-js.js"></script>

  let options = {
  SDKAppID: 0, // 接入时需要将0替换为您的云通信应用的 SDKAppID
    // 从v0.10.2起,新增 tim 参数
    // tim 参数适用于业务中已存在 TIM 实例,为保证 TIM 实例唯一性
    tim: tim
  };
  let trtcCalling = new TRTCCalling(options);

API list

| API | 含义 | | :-------------------------------- | :----------------- | | new TRTCCalling(params) | 初始化 SDK | | setLogLevel(level) | 设置日志级别 | | on(eventName, callback, context) | 监听事件 | | off(eventName, callback, context) | 取消监听事件 | | login(params) | 登录 | | logout() | 登出 | | call(params) | 邀请通话 | | groupCall(params) | 邀请群通话 | | accept(params) | 接受通话邀请 | | reject(params) | 拒绝通话邀请 | | hangup() | 挂断 | | startRemoteView(params) | 启动远端画面渲染 | | stopRemoteView(params) | 停止远端画面渲染 | | startLocalView(params) | 启动本地画面渲染 | | stopLocalView(params) | 停止本地画面渲染 | | openCamera() | 启动摄像头 | | closeCamera() | 关闭摄像头 | | setMicMute(isMute) | 设备麦克风是否静音 | | setVideoQuality(profile) | 设置视频质量 | | switchToAudioCall() | 视频通话切换语音通话 | | switchToVideoCall() | 语音通话切换视频通话 | | getCameras() | 获取摄像头设备列表 | | getMicrophones() | 获取麦克风设备列表 | | switchDevice() | 切换摄像头或麦克风设备 |

event list

| event | 含义 | | :----------------------------------------------- | :------------------------ | | TRTCCalling.EVENT.INVITED, | 收到邀请通知 | | TRTCCalling.EVENT.REJECT, | 被邀用户拒绝通话 | | TRTCCalling.EVENT.NO_RESP, | 被邀用户超时无应答 | | TRTCCalling.EVENT.LINE_BUSY, | 被邀用户正在通话中,忙线 | | TRTCCalling.EVENT.CALLING_CANCEL, | 本次通话被取消了 | | TRTCCalling.EVENT.CALLING_TIMEOUT, | 本次通话超时未应答 | | TRTCCalling.EVENT.CALLING_END, | 本次通话结束 | | TRTCCalling.EVENT.USER_ENTER, | 用户进入通话 | | TRTCCalling.EVENT.USER_LEAVE, | 用户离开通话 | | TRTCCalling.EVENT.KICKED_OUT, | 重复登录,被踢出 | | TRTCCalling.EVENT.USER_VIDEO_AVAILABLE, | 远端用户开启/关闭了摄像头 | | TRTCCalling.EVENT.USER_AUDIO_AVAILABLE, | 远端用户开启/关闭了麦克风 | | TRTCCalling.EVENT.SDK_READY, | SDK 进入 ready 状态 | | TRTCCalling.EVENT.SDK_NOT_READY, | SDK 没有 ready 状态 | | TRTCCalling.EVENT.GROUP_CALL_INVITEE_LIST_UPDATE | 群聊更新邀请列表 |

Error code

| code | 错误类型 | 含义 | | :----------------- | :---------------- | :------------------------------ | | 60001 | 方法调用失败 | switchToAudioCall 调用失败 | | 60002 | 方法调用失败 | switchToVideoCall 调用失败 | | 60003 | 权限获取失败 | 没有可用的麦克风设备 | | 60004 | 权限获取失败 | 没有可用的摄像头设备 | | 60005 | 权限获取失败 | 用户禁止使用设备 | | 60006 | 环境检测失败 | 当前环境不支持webRTC |

参考文档