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融合通信 VoIP SDK","keywords":["ARTC","WebRTC","call"],"contributors":[{"name":"pianzuo.zj"},{"name":"jason.yf"},{"name":"weimin.jwm"}],"license":"MIT","readme":"# Aliyun VoIP Web SDK\n\n阿里云融合通信voip-web-sdk，提供基于WebRTC适用于浏览器VoIP2PSTN或VoIP2VoIP通话场景的SDK。\n\n接入可参考demo: https://github.com/pianzuozj/voip-web-demo\n\n为保证稳定性，建议使用最新版chrome使用该功能\n\n## Status\n[![NPM version][npm-image]][npm-url]\n\n[npm-image]: https://img.shields.io/npm/v/aliyun-voip-web-sdk.svg?style=flat-square\n[npm-url]: https://npmjs.org/package/aliyun-voip-web-sdk\n\n## Installation\n\nYou can install it as dependency with npm/cnpm or yarn.\n\n```sh\n$ # save into package.json dependencies with -S\n$ npm install aliyun-voip-web-sdk -S\n\n$ # you can use cnpm for fast install\n$ cnpm install aliyun-voip-web-sdk -S\n\n$ # use yarn for install\n$ yarn add aliyun-voip-web-sdk\n```\n## 快速开始\n### 一、接入前准备\n\n客户端需要从阿里云申请融合通信账号后才能接入，为了保证安全性，融合通信SDK通过服务端下发的临时token作为身份标识与服务进行交互。临时token可通过融合通信服务pop接口从阿里云获得（详见服务端接入文档），并且具有时效性。SDK在token即将失效或其他需求而需要更新token时，会通过回调通知接入方，接入方需自行实现接口获取最新token并传递给SDK。\n\n![WebSDK交互流程](http://docs-aliyun.cn-hangzhou.oss.aliyun-inc.com/assets/pic/111505/cn_zh/1553053168482/WebSDK%E4%BA%A4%E4%BA%92%E6%B5%81%E7%A8%8B.jpg)\n\n### 二、快速开始\n\n- 创建实例\n```\nimport {\n    AlicomRTC, // AlicomRTC实例\n    Errors, // 错误码\n} from 'aliyun-voip-web-sdk'\n\n/**\n * 创始实例\n */\nvar rtc = new AlicomRTC()\n```\n```\n/**\n * 是否允许日志上传 默认不上传\n * @param enable true 上传日志  false 取消上传日志\n */\n\nsetUploadEnable(enable)\n```\n```\n/**\n * 设置服务监听\n */\nrtc.setServiceListener({\n  /**\n   * 服务连接成功，当前处于可用状态\n   */\n  onServiceAvailable: function () {\n    // 此时可以开始呼叫操作\n  },\n\n  /**\n   * 服务连接失败、或者连接断开了，当前处于不可用状态，服务正在销毁。\n   * @param errCode {number} 错误码\n   * @param errMsg {string} 错误描述\n   */\n  onServiceUnavailable: function (errCode, errMsg) {\n  },\n\n  /**\n   * 服务闲时\n   * @param errCode {number} 错误码\n   * @param errMsg {string} 错误描述\n   */\n  onServiceIdle: function (errCode, errMsg) {\n  },\n\n  /**\n   * 作为被叫，收到点对点音频来电\n   * @param call {Call} 表示这通来电的call实例\n   */\n  onReceivingAudioCall: function(call) {\n    // todo: 已有通话的情况下拒绝新呼入，或是挂断当前通话接听\n  },\n})\n\n/**\n * 可选，设置默认的呼叫超时时间\n */\n// rtc.setDefaultCallTimeout(30)\n```\n\n- 连接服务\n```\n/**\n * 使用RtcId初始化AlicomRTC实例\n */\nrtc.initWithRtcId('RTCID', {\n  updateToken: function (tokenHandler) {\n    var token = {} // todo: 通过服务端接口获取Token\n    tokenHandler.setToken(token)\n  }\n})\n```\n\n- 始初化通话实例\n当`ServiceListener`中的`onServiceAvailable`回调被唤起时，表示服务已经连接成功，此时可以开始创建VoIP2PSTN通话实例，设置并开始通话：\n```\n/**\n * 创建VoIP2Pstn的Call实例，每一次通话都需要生成一个Call实例\n */\nvar pstnCall = rtc.createVoIP2PstnCall('被叫号显', '被叫号码')\n\n/**\n * 创建VoIP2VoIP的Call实例，每一次通话都需要生成一个Call实例\n */\nvar voipCall = rtc.createVoIP2VoIPCall('被叫RtcID')\n\n/**\n * 开启服务端录音（可选）\n */\npstnCall.setServerRecordEnabled(true)\n\n/**\n * 本地禁音（可选）\n */\npstnCall.muteLocalAudio()\n\n/**\n * 本地取消禁音（可选）\n */\npstnCall.unmuteLocalAudio()\n\n/**\n * 发送dtmf (只能是0-9、*、#组成的字符串，最大长度不超过32位)\n */\npstnCall.sendDtmfData(dtmf)\n\n/**\n* 返回当前通话的channelId\n*/\ngetChannelId(): string\n\n```\n\n- 通话状态管理\n\n```\n/**\n * 设置Call状态监听\n */\npstnCall.setCallListener({\n  /**\n   * 电话已通知到被叫方，被叫振铃中\n   * @param talk {Talk} 通话对象\n   */\n  onCalleeRinging: function (talk) {},\n\n  /**\n   * 被叫已接听电话，正在连接中\n   * @param talk {Talk} 通话对象\n   */\n  onCalleeConnecting: function (talk) {},\n\n  /**\n   * 电话拨通了可以正常通话了。被叫已接听\n   * @param talk {Talk} 通话对象\n   */\n  onActive: function (talk) {},\n\n  /**\n   * 对端传来DTMF信息，每次回调只传入一个字符，如果是一连串输入的话则会多次回调，需要接入方自行根据时间戳处理\n   * @param dtmf {string} 对端传入的DTMF信息\n   * @param timestamp {number} 时间戳\n   * @param talk {Talk} 通话对象\n   */\n  onDtmfData: function (dtmf, timestamp, talk) {},\n})\n```\n\n- 主叫开始呼叫 / 被叫开始接听\n\n```\npstnCall.start()\n```\n\n- 停止呼叫/通话\n```\npstnCall.stop()\n```\n\n- 销毁实例\n```\nrtc.destroy()\n```\n\n### 三、常见错误码\n\n| 错误码 | 数值 | 原因 |\n| --- | --- | --- |\n| ERROR_MQTT_CONNECT_FAIL | 1000100 | mqtt连接失败 |\n| ERROR_UPLOAD_TOKEN_FAIL | 1000104 | token上传失败 |\n| ERROR_LOCAL_STOP | 2000000 | 本地主动挂断 |\n| ERROR_REMOTE_HANGUP | 2000099 | 对端主动挂断 |\n| ERROR_CALL_SELF | 2000100 | 拨打自己 |\n| ERROR_CALL_EMPTY | 2000101 | 拨打电话为空 |\n| ERROR_CALLEE_ALERTING_TIMEOUT | 2000103 | 被叫时，振铃超时 |\n| ERROR_REMOTE_REFUSE | 2000106 | 主叫时，对端拒绝应答 |\n| ERROR_REMOTE_CANCEL | 2000107 | 被叫时，对端取消呼叫 |\n| ERROR_NO_ANSWER | 2000108 | 被叫无人接听 |\n| ERROR_SERVICE_UNAVAILABLE | 2000113 | AlicomRTC服务不可用 |\n| ERROR_LOCAL_DESTROY | 2000116 | 本地主动销毁服务 |\n| ERROR_TIME_OUT | 2000102 | joinChannel超时 |\n| ERROR_NETWORK_INVALID | 2000121 | 网络异常 |\n| ERROR_MICROPHONE_NO_PERMISSION | 2000130 | 麦克风无权限 |\n| ERROR_MICROPHONE_NO_DEVICE | 2000135 | 没有麦克风设备 |\n| ERROR_MICROPHONE_NOT_ALLOWED | 2000136 | 浏览器不允许使用麦克风 |\n| ERROR_MICROPHONE_NOT_READABLE | 2000137 | 系统禁用麦克风或者麦克风被占用 |\n| ERROR_SERVER_BASE | 3000000 | 服务端返回业务异常的错误基准 |\n| ERROR_SERVER_KICKED | 3110000 | 账号被登录或被踢 |\n| ERROR_SERVER_UNAVAILABLE | 3110001 | 服务异常 |\n| ERROR_MEDIA_BASE | 4000000 | 媒体sdk返回异常的错误基准 |\n\n\n## License\n\nThe MIT License\n","readmeFilename":"README.md"}