ringcentral/ringcentral-web-phone

RingCentral WebPhone Library for JavaScript WebRTC

ringcentral-web-phone's Issues

Add links to information on Digital Lines

Since Digital Lines are a requirement for RingCentral WebRTC, it would be useful to link to some resources on how to create and manage digital lines. Of note, a limited number of digital lines are available for free for sandbox test accounts.

For sandbox account device selection, we should mention selecting the RingCentral for Desktop softphone option.

Here are some KB articles on Digital Lines:

1. DigitalLine Overview (KB 5862)
2. Purchasing an IP Phone or RingCentral Digital Line from the online account (KB 3136)
3. Assigning a Digital Line to a Different Extension (KB 3748)

Call Forwarding creates undefined error

When you forward incoming call - this error occurs

Uncaught TypeError: Cannot read property 'sendRequest' of undefined

Warm Transfer

Implement warm transfer feature:

1. client puts session on hold
2. client creates a new call
3. client sends REFER in session 1
4. client handles disconnect of session 1

Calling request to.displayName its the same value of from.displayName

When we make a call on the target user page this code

console.log('To', session.request.to.displayName, session.request.to.friendlyName);
console.log('From', session.request.from.displayName, session.request.from.friendlyName);

print the same value the name of From user

Also there is away to get the caller user extension Id ??

Call transfer and call forward doesnt work

1. line.transfer() . Just puts the call on hold
2. line.forward() doesnt forward the call. It puts the call on hold.

Is the widget working ?

Hi
I check out your code and I found the you have widget folder
so my question is finalized or under development ?

Remove PRACK special handler

The fix that morphs PRACK into ACK needs to be removed:

https://github.com/ringcentral/ringcentral-web-phone/blob/master/src/ringcentral-web-phone.js#L350

The options are:

1. Only remove the fix and verify that everything works OK (there might be 4xx errors in console)
2. Completely cancel the PRACK request -- this has never been tested, behavior is unpredictable

If option 1 will not produce errors, go with it, otherwise try option 2. If it will not work -- let @kirill-konshin know, so that we can find another solution.

Error when human formatted phone number is used

Human readable phone numbers are not usable in the demo and return and "Invalid resource owner credentials." error.

For example:

• 1-650-555-1212
• [phone]

It would be ideal to normalize numbers by removing non-digit characters.

Add a panel to show number of active calls

Needs a panel to show all the active calls in the demo. There is no way to track the active calls which are on hold and jump to another call.

Error when country code is omitted from Auth

If you leave out the "1" and you're on an American-based number, you receive an "Invalid credentials" error from the auth request.

Need to do one or more of the following:

• Disable the Register Sip Configurations button in demo until all fields are correct
• Change style settings to reflect error on the fromNumber field in the UI to be in error state (use Bootstrap)
• Provide example of a 'passing' fromNumber as a placeholder
• Provide notification message above fromNumber field with details on issue

Use session.refer() for transfers instead of custom handling

Currently we use custom handling of transfer process but SIP.js has own implementation of session.refer().

When using attended transfer of SIP.JS it adds extra "friendlyName" part to Refer-To header:

Refer-To: "friendlyName" <sip:...>

This part is rejected by RC telephony.

More info:

• http://sipjs.com/guides/transfer/
• http://stackoverflow.com/questions/34133947/make-an-attended-transfer-with-sip-js

Currently SIP.JS's refer method will not return a promise and it's hard to understand was the call successfully transferred.

Here's how target is assembled in SIP.JS:

      target = '"' + target.remoteIdentity.friendlyName + '" ' +
        '<' + target.dialog.remote_target.toString() +
        '?Replaces=' + target.dialog.id.call_id +
        '%3Bto-tag%3D' + target.dialog.id.remote_tag +
        '%3Bfrom-tag%3D' + target.dialog.id.local_tag + '>';

Add coveralls

Cannot hold/unhold the incoming call

Hold & unhold functionality works when call has been started from the app. When the call has been received by the app hold & unhold breaks.

Upgrade to SIPJS 0.7

404 on the favicon for the demo

I have a complete set of device-agnostic favicon and appicons we can use that have the RingCentral circle logo on transparent backgrounds.

Will submit a PR in a moment.

Promise usage on some methods

PhoneLine.prototype.cancel = function() {
    var session = this.getSession();
    return new Promise(function(resolve, reject) {
        session.terminate({statusCode: 486});
        return null;
    });
};

I am wondering if resolve() need to be called after actions finish? Such as:

PhoneLine.prototype.cancel = function() {
    var session = this.getSession();
    return new Promise(function(resolve, reject) {
        session.terminate({statusCode: 486});
        resolve();
        return null;
    });
};

Error when unhold

DOMException: Failed to set local answer sdp: Failed to push down transport description: Offerer must use actpass value for setup attribute.

Call Flip

When not on call and you try to flip the call
Call is disconnected by user when the user is put on hold

Uncaught (in promise) Error: Not on call
at PhoneLine.flip (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:965:31)
at callflip (http://localhost:63342/web-phone-1/demo/index.js:131:14)
at HTMLInputElement.onclick (http://localhost:63342/web-phone-1/demo/index.html:154:87)

Fix the error message

Documentation

Write a minimal viable documentation for classes and methods.

Improve error reporting when outbound calls flag is not enabled.

This is what you receive if a phone is not configured with the outbound call flags, need to add better error handling (we did this the other day, and I thought I pushed it up).

After receiving positive Registered notification and then trying to call outbound, I'm receiving this console.log() error in Firefox v42.0:

WebPhone.prototype.call@http://localhost:8080/build/ringcentral-web-phone.js:398:1
startCall/<@http://localhost:8080/demo/index.js:48:17
lib$es6$promise$$internal$$tryCatch@http://localhost:8080/demo/ringcentral-bundle.js:1412:16
lib$es6$promise$$internal$$invokeCallback@http://localhost:8080/demo/ringcentral-bundle.js:1424:17
lib$es6$promise$$internal$$publish@http://localhost:8080/demo/ringcentral-bundle.js:1395:11
lib$es6$promise$asap$$flush@http://localhost:8080/demo/ringcentral-bundle.js:1206:9

I see the console.log above line (47) with the data which will be provided to webCall() being logged as SIP call to 131760097331 from 16506429233, so the numbers going in appear correct.

Add "Prerequisites" to the README

The README.md file needs to have a section to address the "Prerequisites" for using RingCentral's WebRTC. Items such as:

• A valid RingCentral Account with Connect Platform access
• An extension configured with a Digital Line
• An application defined in the Developer Portal with the appropriate type and permissions

Provide better description for what the audioHelper is and what options are available

Right now, it just seems like a property I have to add, but I don't know why or what options I may have as a developer.

Could you provide some better information, or links to learn more, and to understand why it is part of the SDK.

Error: sip.parser | error parsing header "Contact"

When use web-phone with commonjs module bundler, SIP(v0.7.4) will throw errors when trying to register:
LoggerFactory.js:72 Mon Jun 27 2016 10:44:47 GMT+0800 (CST) | sip.parser | error parsing header "Contact"

Following steps reproduce the error:

1. Open a new project
2. npm i ringcentral ringcentral-web-phone
3. add following code and use a module bundler like webpack:
   https://gist.github.com/changbenny/74f99ee50eb714e60a6dd60dfa399d44

Errors occur when setting up the demo app

After clicking the Register Sip Configurations button, my RC credentials can be authorized, but then I receive the following error messages:

WebSocket connection to 'wss://webphone-sip.devtest.ringcentral.com:8083/' failed: Error in connection establishment: net::ERR_CONNECTION_TIMED_OUT

Unable to Call from WebPhone to Outside number

Not seeing any errors in the console or network tabs, so assuming all is working internally...but the call never is received from my cellular device.

On this call: https://platform.devtest.ringcentral.com/restapi/v1.0/client-info/sip-provision

Seeing this body returned (not capturing error and notifying developer):

  "sipFlags" : {
    "voipFeatureEnabled" : true,
    "voipCountryBlocked" : false,
    "outboundCallsEnabled" : false
  }
}

Step 1: Enable flags check in register function
Step 2: Check flag again when call is created (reject Promise with something like "outbound calling is disabled")

Error in peer.createDTMFSender

Hi
I try the demo and its working but when I try to send DTMF I got this error

TypeError: peer.createDTMFSender is not a function

In ringcentral-web-phone.js line 590

var dtmfSender = peer.createDTMFSender(stream.getAudioTracks()[0]);

Thanks

Add TravisCI

Hangup

When not on call you try to transfer the call, hang the call

Uncaught (in promise) m…e.e…s.I…r.exception {code: 2, name: "INVALID_STATE_ERROR", status: 9, message: "Invalid status: 9"}
InviteServerContext.reject @ Session.js:1153InviteServerContext.terminate @ Session.js:1235PhoneLine.cancel @ web-phone.js:857(anonymous function) @ VM1265:2InjectedScript._evaluateOn @ VM842:875InjectedScript._evaluateAndWrap @ VM842:808InjectedScript.evaluate @ VM842:664

Fix the error handling

Create test suite

The mentioned permission "VoIP Calling" in "Configuring your RingCentral app" does not exists

As my colleague and I are trying to explore how to work with the Web Phone, we stuck while configuring the RingCentral app for Web Phone according to the README file.

The issue is that when we try to set up a new browser-based app, there is no permission VoIP Calling. Should I use different permissions?

Start/Stop Recording

When not on call but sip registered.
When you muting or unmuting
When press start/stop record
Could be handled with in app

TypeError: Cannot read property 'getAudioTracks' of undefined
at setStreamMute (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:1167:24)
at PhoneLine.setMute (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:1157:9)
at Object.service.mute (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:1559:22)
at mute (http://localhost:63342/web-phone-1/demo/index.js:45:20)

Clean up the error msg

UI include pop up for incoming call

Call Mute and Unmute

Remove muted flag and use isOnMute

session.reject(); is not reflect on the caller side

Hi
When you create a call from browser to another
when the target browser reject the call its not reflect on the other side I mean no events triggered on the caller side so the caller can not know the call is rejected

Unhold is broken with SIP.js 0.7.3

VM215 sip.js:2885 DOMException: Failed to set local answer sdp: Failed to push down transport description: Offerer must use actpass value for setup attribute.

Hi Kirill, we have shown you this issue when you were in China office. Please help fix it. Thanks.

SIP.js needs to be packaged with the bundle for factory reference

Currently, looking at the demo code, the sip.js library is referenced externally from the tag. This means that if you do not add this JS src in your head tag, that you will receive a SIP undefined error in JS while trying to execute demo code.

Needs to be bundled within the library itself, especially for apps which hit the 3rd conditional statement of the UMD format being used by src/ringcentral-web-phone.js (lines 9-12)

Call Answering

When Call reaches voice mail You cannot disconnect the call
Call shows not connected, Cannot cancel/hangup the call, cannot be traces in active call array

The phone keeps ringing even after the call reaches voicemail

When pressed answer this error occurs
Uncaught (in promise) Error: The INFO response status code is: 481 (waiting for 200)(…)

Then when you disconnect the call this error occurs
Sun Feb 07 2016 22:50:17 GMT-0800 (PST) | sip.sipmessage | error parsing "From" header field with value "sip:@XXXXXXXXX;tag=434hg6583k"

Remote call doesn't end

Hangup from the phone . The current session in the local ends. The session in the remote doesn't end.

line.cancel() gives this error
Uncaught m…e.e…s.I…r.exception {code: 2, name: "INVALID_STATE_ERROR", status: 9, message: "Invalid status: 9"}

NPM & Bower

• Create a procedure to propagate version parameter to appropriate JSON files
• Add necessary fields to JSON files

Add Permission option to allow use microphone in the browser

Need to add an option to create a pop up to allow microphone permission .

Outbound caller ID is a company number instead of direct number

When making outbound call from RC Corp account even if user has direct number the company number is shown as caller ID on the mobile phone.

Answer Incoming Call

Making a call to a number and when the call is not picked if you try to disconnect this error is seen.
The call reaches voice mail

Uncaught INVALID_STATE_ERROR: Invalid status: 9InviteServerContext.reject @ Session.js:1153InviteServerContext.terminate @ Session.js:1235PhoneLine.cancel @ web-phone.js:857UserAgent.hangup @ web-phone.js:394service.hangup @ web-phone.js:1450disconnect @ index.js:84onclick @ index.html:139
index.js:15 emitting event newTransaction
index.js:15 emitting event stateChanged
index.js:15 emitting event stateChanged
index.js:15 emitting event accepted

Fix error handling

Demo not working on Firefox

Cannot run the demo on Firefox

Disconnecting incoming call thows error

If you hangup incoming call. It does hangup localstream but not the remote stream. It throws this error:

Thu Feb 11 2016 13:34:58 GMT-0800 (PST) | sip.inviteservercontext | ReferenceError: service is not defined
    at Object.<anonymous> (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:12453:23)
    at Object.<anonymous> (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:1523:27)
    at Array.forEach (native)
    at Object.EventEmitter.emit (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:1521:32)
    at Object.ServerContext.reject (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:5028:8)
    at Object.InviteServerContext.reject (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:6341:40)
    at Object.InviteServerContext.terminate (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:6418:12)
    at PhoneLine.cancel (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:12585:13)
    at UserAgent.hangup (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:12097:14)
    at WebPhone.hangup (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:436:21)

Add support for 3-legged OAuth

Please add support for 3-legged OAuth to the demo app.

Update readme for npm installation and bower

Propagate X-User-Agent and Client-Id headers

sendDTMFInfo

Need to check functionality of Info DTMF as it uses session.dtmf(tone, options{} ) - cannot send regular dtmf signals through this

Call Forward

If clicked for outgoing call session : this is the error you get-
Uncaught (in promise) TypeError: self.session.accept is not a function
at http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:1138:22
at PhoneLine.answer (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:1118:12)
at PhoneLine.forward (http://localhost:63342/web-phone-1/build/ringcentral-web-phone.js:1097:17)
at forward (http://localhost:63342/web-phone-1/demo/index.js:156:14)

Fix the error msg.

Call-duration

Include call-duration panel to show the call-duration for incoming call and out-going call.

Make changes to demo

1. Remove floating console window.
2. Divide the demo into panels and hide/unhide functionality panel depending on the sip registrations status
3. Clean up demo