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Refer to the API guide for detailed documentation on the APIs used.","readme":"# How to Write Aculab WebRTC Apps\n\nThis guide shows the basic steps needed to write WebRTC applications for Aculab Cloud. Refer to the API guide for detailed documentation on the APIs used.\n\nWebRTC Client instances can call, and be called from, [Aculab Cloud services]( https://www.aculab.com/cloud/guides/outbound-and-inbound-services/) and other WebRTC Client instances. They cannot call PSTN or SIP phones directly, although of course Aculab Cloud services can call those devices.\nRefer to interface.html for detailed documentation on the APIs used.\n\n## Integrating WebRTC\n\nInclude the JavaScript library by adding the [aculab-webrtc](https://www.npmjs.com/package/aculab-webrtc) npm package\n\n```\nyarn add aculab-webrtc\n```\n\n\n### Video calls between WebRTC Client instances\n\nThe default behaviour of the interface is to set up audio only calls. When the call is between two WebRTC Client instances, you can enable video in both or just one direction.\n\nTo make a call with video to a WebRTC Client instance you need to pass a suitably configured AculabCloudCallOptions object to AculabCloudClient.callClient(). The constraints property of the options object is passed to the browser's getUserMedia() function. For example:\n```\n    var options = {\n        constraints: {\n            audio: true,\n            video: true\n        }\n    };\n    call = acc.callClient(\"other_client_id\", token, options);\n```\nThis module is used for Android and iOS app development by Aculab’s [React Native](https://github.com/aculab-com/react-native-aculab-client) package, which has a dependency on the react-native-webrtc module. This mutes video by stopping the stream, rather than sending a stream of 0's in order to shut off camera light (see https://github.com/react-native-webrtc/react-native-webrtc/issues/643).\n\nWe have implemented a workaround that makes callbacks available for local and remote video mute/unmute.  \n\n```\n    aculab_cloud_call.onLocalVideoMuteCB\n    aculab_cloud_call.onLocalVideoUnMuteCB\n```\n\nThe local callbacks are triggered when video is muted/unmated and can place an image in the local view of the call window.\n\n```\n    aculab_cloud_call.onRemoteVideoMuteCB\n    aculab_cloud_call.onRemoteVideoUnMuteCB \n```\n\nThe remote callbacks are triggered when RTP data is detected to have started or stopped and can indicate the change in state to the user.\n\n\n\nFor incoming calls from WebRTC Client instances, you can determine the caller's media settings for the call from the object passed to the onIncoming callback. For a call with video in both directions, the offeringVideo and canReceiveVideo properties of that object will be true. To answer the call with video, you need to pass a suitably configured AculabCloudCallOptions object to AculabCloudIncomingCall.answer(). For example:\n```\n    var options = {\n        constraints: {\n            audio: true,\n            video: true\n        }\n    };\n    call.answer(options);\n```\n\nNote that calls will fail if the constraints require a local video stream but the user doesn't have a webcam, or refuses access to it. To receive video but not send it, you can use:\n```\n    var options = {\n        constraints: {\n            audio: true,\n            video: false\n        },\n        receiveVideo: true\n    };\n```\n\n## Creating a WebRTC Client Instance\n\nTo create a WebRTC Client instance, you will need the following information:\n\n\nCloud ID - [The Cloud ID]( https://www.aculab.com/cloud/guides/cloud-regions/) for your Aculab Cloud account, for example 1-2-0.\nWebRTC Access Key - [The WebRTC Access Key](https://cloud.aculab.com/home/webrtcsettings) for your Aculab Cloud account.\nClient Id -\tA text identifier which Aculab Cloud uses to identify a specific WebRTC Client instance.\n\nAll WebRTC Client instances use the AculabCloudClient class. Create a new instance of the class with:\n\n```\n    var cloudId = \"1-2-0\";\n    var webRtcAccessKey = \"3krt4sfjo...\";\n    var clientId = \"myClient-4369\";\n    var logLevel = 2;\n    acc = new AculabCloudClient(cloudId, webRtcAccessKey, clientId, logLevel);\n```\n\nwhere logLevel is between 0 and 6 inclusive. 0 disables logging.\n\n## Enabling Incoming Calls on a WebRTC Client Instance\n\nIf you will be accepting incoming WebRTC calls, then you need to set up the onIncoming and onIncomingState callbacks. \n\n```\n    acc.onIncoming = newCall;\n    acc.onIncomingState = incomingState;\n```\n\nSee [Accepting an Incoming WebRTC Call](#accepting-an-incoming-webrtc-call) for information on the newCall and incomingState functions.\n\nA WebRTC Client instance needs a token to be able to receive incoming WebRTC calls. This token is retrieved from the [webrtc_generate_token](https://www.aculab.com/cloud/web-services/webrtc-clients/ ) web service. It should be generated by server side code, so the client does not know your web services API access key. An example as follows:\n\n```\n    fetch('/get_webrtc_token_for_client', {\n        method: 'POST',\n        body: clientId\n      })\n      .then(response => response.json())\n      .then(data => acc.enableIncoming(data.token));\n```\n\nThe onIncomingState callback is used to notify you if incoming WebRTC calls are enabled or disabled.\n\n## Calling a WebRTC Client instance from WebRTC\n\nWebRTC Client instances can call other WebRTC Client instances. This is achieved by calling AculabCloudClient.callClient() then setting up the AculabCloudOutgoingCall callbacks to handle the call control events.\n\nA WebRTC Client instance needs a token to be able to call another WebRTC Client instance. This token is retrieved from the [webrtc_generate_token](https://www.aculab.com/cloud/web-services/webrtc-clients/ ) web service. It should be generated by server side code, so the client does not know your web services API access key.\n\nAn AculabCloudCallOptions object can be passed to configure the media settings. For example, this can be used to make video calls by requesting access to the user's webcam.\n\n```\n    var options = { constraints: { audio: true, video: true } };\n    call = acc.callClient(\"other_client_id\", data.token, options);\n    call.onDisconnect = callDisconnected;\n    call.onRinging = callRinging;\n    call.onMedia = gotMedia;\n    call.onConnecting = connecting;\n    call.onConnected = connected;\n```\n\nSee [Call Control Callbacks](#call-control-callbacks) for information on these callbacks.\n\n## Calling an Aculab Cloud Service from WebRTC\n\nWebRTC Client instances can call Aculab Cloud inbound services. This is achieved by calling AculabCloudClient.callService() then setting up the AculabCloudOutgoingCall callbacks to handle the call control events.\n\n```\n    call = acc.callService(encodeURIComponent(\"incoming_service_name\"));\n    call.onDisconnect = callDisconnected;\n    call.onRinging = callRinging;\n    call.onMedia = gotMedia;\n    call.onConnecting = connecting;\n    call.onConnected = connected;\n```\n\nSee [Call Control Callbacks](#call-control-callbacks) for information on these callbacks.\n\n## Calling a WebRTC Client Instance from Aculab Cloud\n\nUAS and REST applications can call a WebRTC Client instance by setting the call destination to webrtc:<clientId>.\n\nThe Caller ID specified is passed to the onIncoming callback.\n\n## Calling a WebRTC Client Instance from the REST API Version 2.0\n\nAn example of using the Connect action to call a WebRTC Client instance, written in Python.\n\n```\n    my_connect = Connect()\n    my_connect.set_next_page(\"/call_finished\")\n    my_connect.append_destination(\"webrtc:{0}\".format(clientId))\n    my_connect.set_call_from('441234567890')\n    my_connect.set_hold_media(Play(file_to_play='holdmusic.wav'))\n    my_connect.set_secondary_call(SecondaryCall(first_page=WebPage(url='secondary_first'),\n                                                final_page=WebPage(url='secondary_final'),\n                                                error_page=WebPage(url='error_page')))\n```\n\n## Calling a WebRTC Client Instance from the UAS API\n\nA Python example of using the UASCallChannel object to call a WebRTC Client instance.\n\n    if channel.start_call('webrtc:{0}'.format(clientId), call_from='441234567890') is True:\n        # we have started an outbound call\n        pass\n## Accepting an Incoming WebRTC Call\n\n### onIncomingState\n\nonIncomingState is used to inform you whether incoming WebRTC calls are enabled or disabled.\n```\n    function incomingState(obj) {\n        if (obj.ready) {\n            // We can receive incoming calls\n        } else {\n            // We cannot receive incoming calls\n            // Report obj.cause to the user\n        }\n    }\n```\n\n### onIncoming\n\nIf WebRTC Client has been configured to accept incoming calls, then the onIncoming callback is used to accept any incoming call.\n\nThe onIncoming callback should set up the AculabCloudIncomingCall callbacks, notify the user of the call, and send the ringing notification to the remote end.\n```\n    // newCall has been assigned to the onIncoming callback above\n    function newCall(obj) {\n        call = obj.call;\n        callerId = obj.from;\n        inbound = true;\n\n        // Set up callbacks\n        call.onDisconnect = callDisconnected;\n        call.onMedia = gotMedia;\n        call.onConnecting = connecting;\n        call.onConnected = connected;\n\n        // Play a ringing to notify user of the call\n        // get the HTMLMediaElement element\n        var player = document.getElementById('player');\n        if (player.canPlayType('audio/wav')) {\n            player.loop = 'loop';\n            player.src = 'audio/ringing.wav';\n            player.type = 'audio/wav';\n            player.load();\n            var p = player.play();\n            if (p !== undefined) {\n                p.catch(error => {\n                    // Handle error\n                });\n            }\n        } else {\n            // Browser can't play audio/wav\n        }\n\n        // Send ringing notification to the caller\n        call.ringing();\n    }\n```\n\nSee [Call Control Callbacks](#call-control-callbacks) for information on these callbacks.\n\n### Answering the call\n\nWhen the user accepts the incoming call the call's answer() function needs to be called. For calls from WebRTC Client instances, an AculabCloudCallOptions object can be passed to configure the media that is sent.\n\n### Call Control Callbacks\n\nThe various Call Control Callbacks have been set up in the code snippets above. Example code showing how these callbacks are implemented is shown in this section.\n\n### onDisconnect\n\nThis callback is called when the call has been disconnected. The player element should be reset, and any resources cleaned up.\n```\n    function callDisconnected(obj) {\n        // Notify the user that the call has disconnected.\n\n        // Stop anything that is playing\n        stopPlayer();\n\n        // reset state\n        call = null;\n        inbound = false;\n        isconnected = false;\n    }\n\n    // Function to stop the HTMLMediaElement element playing.\n    // This is used by the callDisconnected and gotMedia callbacks.\n    function stopPlayer() {\n        // Get the HTMLMediaElement element\n        var player = document.getElementById('player');\n\n        // Reset the player state\n        player.pause();\n        player.loop = '';\n        player.src = '';\n        player.srcObject = null;\n        player.type = '';\n        player.load();\n        playing_ringing = false;\n        gotremotestream = false;\n    }\n```\n\n### onRinging\n\nIf you have not got an media stream from the remote end, then you can generate a ringtone locally.\n```\n    function callRinging(obj) {\n        // get the HTMLMediaElement element\n        var player = document.getElementById('player');\n        if (!playing_ringing) {\n            if (player.canPlayType('audio/wav')) {\n                // Play a ringback tone if we have not got an audio stream\n                // from the remote end\n                // gotremotestream is initialised in gotMedia()\n                if (!gotremotestream) {\n                    player.loop = 'loop';\n                    player.src = 'audio/ringback.wav';\n                    player.type = 'audio/wav';\n                    player.load();\n                    var p = player.play();\n                    if (p !== undefined) {\n                        p.catch(error => {\n                            // Handle error\n                        });\n                    }\n                }\n            } else {\n                // Browser can't play audio/wav\n            }\n            playing_ringing = true;\n        }\n    }\n```\n\n### onMedia\n\nThe onMedia callback is used to give you the remote media stream which you can connect to your HTMLMediaElement element.\n```\n    function gotMedia(obj) {\n        // Stop anything that is playing\n        stopPlayer();\n\n        var player = document.getElementById('player');\n        gotremotestream = true;\n        player.srcObject = obj.stream;\n        player.load();\n        var p = player.play();\n        if (p !== undefined) {\n            p.catch(error => {\n                // Handle error\n            });\n        }\n    }\n```\n\n### onConnecting\n\nonConnecting is called when the browser has the local media stream and is preparing the sockets needed to transport the call media. There is nothing that specifically needs doing in this callback, but it can be used to connect the local media stream to show the camera view.\n```\n    function connecting(obj) {\n        // Update a web page element to notify the user that the\n        // call is connecting\n    }\n```\n\n### onConnected\n\nWhen the call has been answered the onConnected callback will be called. You can use this callback to notify the user about call progress.\n```\n    function connected(obj) {\n        // Update a web page element to notify the user that the\n        // call is connected\n        isconnected = true;\n    }\n```\n\n### Disconnecting a Call\n\nDisconnecting a connected call is as simple as calling disconnect() on the call object. Inbound calls that have not connected should be rejected with a valid SIP cause, for example 486 busy here, instead.\n```\n    function stopCall() {\n        if (call) {\n            // If inbound call has not been accepted, then reject the call\n            if (inbound && !isconnected) {\n                call.reject(486);\n            } else {\n                call.disconnect();\n            }\n        }\n    }\n```\n\n\n### Troubleshooting\n\nA number of common issues can cause problems with media in a WebRTC call.\n\nIf the same username is registered from more than one browser instance then the most recently registered instance will receive incoming WebRTC calls. As WebRTC Client instances are automatically re-registered at various intervals, the most recently registered instance may change without warning. Therefore, in general it's best to register a given username with only one browser or application instance at a time.\n\n \n\n \n\n \n\n \n\n \n\n \n\n \n","readmeFilename":"README.md"}