Huge News!Announcing our $40M Series B led by Abstract Ventures.Learn More →
plivo-browser-sdk
Advanced tools
plivo-browser-sdk - npm Package Compare versions
Comparing version 2.1.0-beta.2 to 2.1.0-beta.3
| { | ||
| "name": "plivo-browser-sdk", | ||
| "title": "plivo-browser-sdk", | ||
| "version": "2.1.0-beta.2", | ||
| "description": "Plivo WebRTC sdk for use in chrome/firefox ", | ||
| "version": "2.1.0-beta.3", | ||
| "description": "This library allows you to connect with plivo's voice enviroment from browser", | ||
| "main": "./dist/plivo.js", | ||
@@ -22,3 +22,3 @@ "scripts": { | ||
| }, | ||
| "author": "", | ||
| "author": "Plivo", | ||
| "license": "Apache-2.0", | ||
@@ -25,0 +25,0 @@ "bugs": { |
235
README.md
@@ -1,235 +0,26 @@ | ||
| # PlivoWebSDK | ||
| A pure javascript webRTC-SIP library | ||
| ## Setup | ||
| `npm install` | ||
| ## Development Environment | ||
| `npm start` | ||
| This builds the SDK file and attaches it to the port 9000 - `http://localhost:9000/plivowebsdk.js` | ||
| This uses [webpack-dev-server](https://github.com/webpack/webpack-dev-server) so, when there is a change in any file, the build file is auto build, | ||
| When this is used in [plivo-websdk-2.0-example](https://github.com/plivo/plivo-websdk-2.0-example), the app will auto reload with the build with latest changes | ||
| ## Build | ||
| `npm run build` | ||
| Minified, production-ready build is created at `dist/plivowebsdk.min.js` | ||
| `npm run build:nominfy` | ||
| A non-minified version of the build file is created `dist/plivowebsdk.js` | ||
| ## Contribution | ||
| Follow this [Git branching Strategy](https://docs.google.com/document/d/10D4hIgwRlxhPMn3xZzZVPEyeszGLw2DF1b1z0g3ymEs/edit?ts=5b616182&pli=1) | ||
| Use your Plivo's google account to view that document. | ||
| ## Deploy | ||
| To deploy to production (CDN77 and CloudFront), just push to master, the upload is handled by CircleCI. | ||
| ## Revert plan to revert a buggy deployment | ||
| 1) `git checkout master` | ||
| 2) `git pull origin master` | ||
| 3) `git reset --hard <last-stable-tag>` replace `<last-stable-tag>` with last stable release tag | ||
| 4) `git push -f origin master` this force pushes the tag to master and the CI will kick in and deploy the last stable release to production. | ||
| ## Run sample app | ||
| Refer [example app](https://github.com/plivo/plivo-websdk-2.0-example) | ||
| Getting Started | ||
| =============== | ||
| Including the javascript | ||
| ------------------------ | ||
| * [Documentation](https://s3.amazonaws.com/plivowebrtc/v2-0.html) | ||
| * [Example](https://github.com/plivo/plivo-websdk-2.0-example) | ||
| Add the javascript file to the example app or to the application that you are build to use the SDK. | ||
| To use the development build: | ||
| ```html | ||
| <script type="text/javascript" src="http://localhost:9000/plivowebsdk.js"></script> | ||
| ``` | ||
| To use the production build: | ||
| ```html | ||
| <script type="text/javascript" src="https://cdn.plivo.com/sdk/browser/v2/plivo.min.js"></script> | ||
| ``` | ||
| Initialising the Plivo JS Object | ||
| NPM Installation | ||
| -------------------------------- | ||
| You may specify a set of configuration parameters to initialise the object. The details are covered in the [configuration section] | ||
| ``` | ||
| var options = {"debug":"DEBUG","permOnClick":true,"codecs":["OPUS","PCMU"],"enableIPV6":false,"audioConstraints":{"optional":[{"googAutoGainControl":false}]},"enableTracking":false,"appId":null}; | ||
| var plivoWebSdk = new window.Plivo(options); | ||
| ``` | ||
| Event registration | ||
| ------------------ | ||
| ```npm install plivo-browser-sdk --save``` | ||
| Pass function references to the events produced from the sdk. This is where your UI manipulation should handle all the different call flows. | ||
| #### Usage: | ||
| ```js | ||
| plivoWebSdk.client.on('onWebrtcNotSupported', onWebrtcNotSupported); | ||
| plivoWebSdk.client.on('onLogin', onLogin); | ||
| plivoWebSdk.client.on('onLogout', onLogout); | ||
| plivoWebSdk.client.on('onLoginFailed', onLoginFailed); | ||
| plivoWebSdk.client.on('onCallRemoteRinging', onCallRemoteRinging); | ||
| plivoWebSdk.client.on('onIncomingCallCanceled', onIncomingCallCanceled); | ||
| plivoWebSdk.client.on('onCallFailed', onCallFailed); | ||
| plivoWebSdk.client.on('onCallAnswered', onCallAnswered); | ||
| plivoWebSdk.client.on('onCallTerminated', onCallTerminated); | ||
| plivoWebSdk.client.on('onCalling', onCalling); | ||
| plivoWebSdk.client.on('onIncomingCall', onIncomingCall); | ||
| plivoWebSdk.client.on('onMediaPermission', onMediaPermission); | ||
| plivoWebSdk.client.on('mediaMetrics',mediaMetrics); | ||
| plivoWebSdk.client.on('audioDeviceChange',audioDeviceChange); | ||
| plivoWebSdk.client.on('onConnectionChange',onConnectionChange); | ||
| ``` | ||
| ```const Plivo = require('plivo-browser-sdk');``` | ||
| Registering using your Plivo Endpoint | ||
| ------------------------------------- | ||
| or | ||
| Register using your Plivo Endpoints credentials | ||
| ```import Plivo from 'plivo-browser-sdk'``` | ||
| ```js | ||
| var username = “johndoe1234@phone.plivo.com”; | ||
| var password = “XXXXXXXX”; | ||
| plivoWebSdk.client.login(username, password); | ||
| CDN | ||
| ------------------ | ||
| ```html | ||
| <script type="text/javascript" src="https://cdn.plivo.com/sdk/browser/v2/plivo.min.js"></script> | ||
| ``` | ||
| Making a call | ||
| ------------- | ||
| Making a call is among the various methods exposed by the Plivo JS Object. The other methods are described under [Methods]. You can pass extra headers to be sent with the call, but is optional. | ||
| ```js | ||
| var dest = “<jane1234@phone.plivo.com>”; | ||
| var extraHeaders = {'X-PH-Test1': 'test1', 'X-PH-Test2': 'test2'}; | ||
| plivoWebSdk.client.call(dest, extraHeaders); | ||
| ``` | ||
| Accepting a call | ||
| ---------------- | ||
| This is how an incoming call is answered, which is notified by the event `onIncomingCall`. | ||
| ```js | ||
| plivoWebSdk.client.answer(); | ||
| ``` | ||
| Sending DTMF | ||
| ------------ | ||
| This is how DTMF is sent during a call. | ||
| ```js | ||
| plivoWebSdk.client.sendDtmf('1'); | ||
| ``` | ||
| Configuration | ||
| ============= | ||
| All of these options are optional, if not present their default value will be used. | ||
| | **Option** | **default** | **Description** | | ||
| |---------------------|-------------|--------------------------------------------------------------------------------------------------------| | ||
| | `debug` | `DEBUG` | Set logging level for sdk, valid values: OFF, ERROR, WARN, INFO, DEBUG, ALL | | ||
| | `permOnClick` | `false` | Set to true if you want to ask for mic permission before call. Otherwise it will be asked on page load | | ||
| | `enableIPV6` | `false` | Enables or disables IPv6 candidate collection. *Experimental | | ||
| | `codecs` | `["OPUS","PCMU"]` | Valid array values include `OPUS` or `PCMU`. Setting one or the other will force that codec and remove the other. If no codecs is provided it will default to OPUS only| | ||
| | `audioConstraints` | `{}` | Audio constraints object that will be passed to webRTC `getUserMedia()`. This is a browsers specific audio controls.| | ||
| |`enableTracking` | `false` | set `true` to collect call stats | ||
| | `appId` | `null` | if `enableTracking` is `true` set a valid callstats.io `application id`. If you don't have an `appId` and still wanna track call stats, Just set `enableTracking: true` and we'll use our default account.| | ||
| | `dscp` | `false` | Buy setting it to `true` you can enable QoS in voice traffic. Differentiated Services field in the packet headers for all WebRTC traffic. | ||
| | `registrationDomainSocket` | `null` | if you want to register with specific Plivo websocket server. | ||
| | `preDetectOwa` | `false` | set it to `true` if you want to detect one way audio before answering/sending the call. | ||
| Methods | ||
| ======= | ||
| These are all the methods supported on the `plivoWebSdk.client` | ||
| | **Option** | **params** | **Description** | | ||
| |-------------|--------------------------|------------------------------------------| | ||
| | `login` | `username`, `password` | Registering a Plivo endpoint | | ||
| | `logout` | | Logging out from the registered endpoint | | ||
| | `call` | `number`, `extraHeaders` | Call a number or sip address. |getCallUUID | ||
| | `answer` | | Answer an incoming call | | ||
| | `hangup` | | Hangup an ongoing call | | ||
| | `reject` | | Reject and incoming call | | ||
| | `sendDtmf` | `digit` | Send the digits as dtmf | | ||
| | `mute` | | Mute the mic | | ||
| | `unmute` | | Unmute the mic | | ||
| | `setRingTone`|`url to audio file` | Set the ring tone that plays when calling, default tone is [here](https://s3.amazonaws.com/plivosdk/audio/us-ring.mp3) | | ||
| | `setRingToneBack`| `url to audio file` | Set the ring tone that plays when being called, default tone is [here](https://s3.amazonaws.com/plivosdk/audio/us-ring.mp3) | | ||
| |`setConnectTone`| true/false| Dial beep will play till we get 18X response from server. setting `false` will not play beep tone.| | ||
| | `setDtmfTone`| `digit`, `url to audio tone` | Set the tone played when sending dtmf. Default tone is located [here](https://s3.amazonaws.com/plivosdk/audio/dtmf-2.mp3), where you need to replace digit between `dtmf-[digit].mp3`. | | ||
| | `setDebug`| `INFO` | will look for any of `[INFO, DEBUG, ERROR, WARN, ALL, OFF]` Set the log level of the sdk, `DEBUG` and `ALL` will show all logs, `OFF` will turn off all plivo logs | | ||
| | `sendQualityFeedback` | `(callUUID`,`score`, `comment`) | score is between 1-5. For score 1-4, comment must be one of the following values ['bad-audio', 'call-dropped', 'wrong-callerid', 'post-dial-delay', 'dtmf-not-captured', 'audio-latency', 'unsolicited-call', 'one-way-audio', 'no-audio', 'never-connected'], only valid if call stats integration setup. If invalid score/comment returns error| | ||
| | `getCallUUID` | | Returns call UUID if a call is active, else returns null| | ||
| | `getLastCallUUID`| | Returns last call UUID, Useful in the cases if you want to send feedback for last call | ||
| |`webRTC`| | Return `true` if webRTC is supported. `false` if there is no webrtc support | ||
| |`version`| | It is a variable that returns current version of Plivo SDK | ||
| #### Audio API | ||
| Methods allowed in audio API `plivoWebSdk.client.audio` | ||
| | Name | Param | Description | ||
| |---------------------|-------------|--------------------------------------------------------------------------------------------------------| | ||
| | `availableDevices` {Function}| `filter` {String} - Optional | Promise based callback! Pass `'input'` as filter by Input audio devices,`'output'` as filter by Output audio devices, `null` to get all audio devices | ||
| | `revealAudioDevices` {Function} | `arg` {String} - Optional |Promise based callback! This will force and ask for allow permission, On permisson allowed it will list available devices. Pass `'returnStream'` as arg to get local stream in Promise success | ||
| |`microphoneDevices` {Object}| `set(deviceId)` - deviceId {String} is Manditory, `get()`,`reset()` | `set` method will set the audioDevice ID as default Microphone device. `get` method will return the Microphone device ID which is set already!, `reset` method will remove any Microphone device ID which is already set! | ||
| |`speakerDevices` {Object}|`set(deviceId)` - deviceId {String} is Manditory, `get()`,`reset()`,`media(source)`- source {String} is Manditory| `set` method will set the audioDevice ID as default Speaker device for DTMF and Callee's/Remote audio. `get` method will return the Speaker device ID which is set already!, `reset` method will remove any Speaker device ID which is already set! `media` will take `'dtmf'` or `'ringback'` as source name and will return a HTML audio element for the source name you passed | ||
| |`ringtoneDevices` {Object}|`set(deviceId)` - deviceId {String} is Manditory, `get()`,`reset()`,`media()`| `set` method will set the audioDevice ID as default Ringtone device for incoming ringtone audio. `get` method will return the Ringtone device ID which is set already!, `reset` method will remove any Ringtone device ID which is already set! `media` method will return Ringtone HTML audio element. | ||
| Events | ||
| ====== | ||
| | **Event** | **Description** | | ||
| |----------------------------------------------|----------------------------------------------------------------------------------| | ||
| | `onLogin` | Occurs when a login is successful | | ||
| | `onLoginFailed` | Occurs when a login has failed | | ||
| | `onLogout` | Occurs when a logout is successful | | ||
| | `onCalling` | Occurs when a call is initiated | | ||
| | `onCallRemoteRinging` | Occurs when the remote end starts ringing during an outbound call | | ||
| | `onCallAnswered` | Occurs when the an outbound call is answered | | ||
| | `onCallTerminated` | Occurs when the an outbound call has ended | | ||
| | `onIncomingCall` | Occurs when there is an incoming call | | ||
| | `onIncomingCallCanceled` | Occurs when an incoming call is cancelled by the caller | | ||
| | `onCallFailed(cause)` | Occurs when an outbound call fails | | ||
| | `onMediaPermission` | Occurs when media permission has been granted | | ||
| | `onWebrtcNotSupported` | Occurs when browser does not support web rtc | | ||
| | `audioDeviceChange(deviceObj)` | Occurs when there is a change in USB audio device, Device added or removed | | ||
| | `onConnectionChange` | Occurs when the connection state is changed such as websocket abruptly closes | | ||
| | `mediaMetrics`| `Works only for Chrome` Occurs with below `type` `high_jitter` When the jitter is higher than 30 ms for 3 out of last 5 samples, `high_latency` When the RTT is higher than 400 ms for 3 out of last 5 samples, `high_packetloss` When the packet loss is > 10% for OPUS and loss > 20% PCMU, `low_mos` When sampled mos is < 3 for 3 out of last 5 samples, `no_microphone_access` When we detect one way audio (<80 bytes sent in 3 seconds), `no_audio_received` When the user is not able to hear the callee or When audio level is stable for last 3 samples. All the above will have `level`:`warning` its `group` : `network/audio` related params will have its own `value` and state `active`:`true/false` , `ice_timeout` will get triggered when ICE gathering takes more than 2 sec either for outgoing call invite or incoming call answer. Example => `{active : true, desc : "local_audio", group : "audio", level : "warning", type : "no_audio_received", value : 0}` | ||
| Tests | ||
| ===== | ||
| These are just behavioural in nature which just tests if a call flows as it should. The SDK code is not unit testable as it has no modularity(all the functions are defined in the callbacks itself). | ||
| # Pre requisites | ||
| + firefox(from version 56) and chrome(from version 59) supporting headless execution | ||
| + tmux | ||
| + [linphonec](http://www.linphone.org/) | ||
| + sip endpoints credentials in `test/spec.js` | ||
| Run `npm test` in the root directory to start executing tests on firefox and chrome in headless mode | ||
| Refer [Change log](https://github.com/plivo/plivo-websdk-2.0/blob/master/CHANGELOG.md) for version changes |
Fixed alerts
No contributors or author data
MaintenancePackage does not specify a list of contributors or an author in package.json.
Found 1 instance in 1 package
Improved metrics
- Number of package files
- increased by25%
5
- Number of low maintenance alerts
- decreased by-100%
0
Worsened metrics
- Total package byte prevSize
- decreased by-51.02%
13697
- Number of lines in readme file
- decreased by-88.56%
27
No dependency changes