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JavaScript 76.41% Shell 3.13% CSS 9.67% HTML 10.79%

jscommunicator's Introduction

Introduction

JSCommunicator is a WebRTC communication tool developed in HTML, CSS and JavaScript. It supports voice, video and chat messaging using the SIP protocol over WebSockets.

Please see http://jscommunicator.org for more details about features, examples, mailing lists and the latest source code.

Server requirements

To use JSCommunicator, you require a SIP proxy that supports the SIP over WebSockets transport.

Only recent versions of the SIP proxies support WebSockets.

repro from reSIProcate http://www.resiprocate.org

Kamailio http://www.kamailio.org

See the Real-Time Communication Quick Start Guide for details about setting up a SIP and TURN server to support WebRTC calls.

Dependencies

You need to have the following JavaScript libraries available in your web server or CMS:

JsSIP The latest code on the JSCommunicator master branch works with JsSIP version 0.6.x

jQuery (v1.4 or greater has been tested)

jQuery UI

jQuery.i18n.properties

ArbiterJS (v1.0 has been tested)

Font Awesome (v4.1 or greater has been tested, earlier versions do not work)

All dependencies can be easily fetched using either of the following methods:

on a Debian/Ubuntu system, use the "deb-setup.sh" script
on other systems, see the "code_grabber" script

Getting started

For integration in static or dynamically generated web sites, frameworks and Content Management Systems, please see INTEGRATION.md

Use cases

Standalone:
- include the HTML fragment into a static HTML page, possibly using Server Side Includes (SSI). Make sure the page references all the required JavaScript source files.
In a CMS, wiki or other publishing platform:
- include the HTML fragment into a block in a page managed by a CMS. Configure the CMS to send the JavaScript to the user or include <script> tags within the content body with the phone fragment. The DruCall module for Drupal provides an excellent example of how to do this, see http://drucall.org

Interaction with other JavaScript applications in a web page

ArbiterJS is used to provide a loosely-coupled mechanism for integration with other JavaScript modules used in the same page/site.

For example, you can:

send a signal to JSCommunicator telling it which destination to dial
receive notifications from JSCommunicator when calls are made or received and use this information to query or update an address book or CMS, display related information about the caller in another part of the screen, etc

For an example, see the files:

event-demo.shtml event-demo.js

Credits

JSCommunicator is based on the JsSIP library and inspired by the TryIt.jssip.net and RetroRTC demo applications produced by Versatica.

License

JSCommunicator http://jscommunicator.org

The JavaScript code in this page is free software: you can redistribute it and/or modify it under the terms of the GNU General Public License (GNU GPL) as published by the Free Software Foundation, either version 2 of the License, or (at your option) any later version. The code is distributed WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU GPL for more details.

You may distribute non-source (e.g., minimized or compacted) forms of that code without the full copy of the GNU GPL normally required provided you include this license notice and a URL through which recipients can access the Corresponding Source.

jscommunicator's People

Contributors

Stargazers

Watchers

jscommunicator's Issues

display more info about why a call failed

The call failure details are currently too brief

Display more details about what went wrong

Add buttons for sending DTMF symbols A, B, C and D

DTMF allows the symbols A, B, C and D.

We should have buttons for these on the dial pad

Better UI feedback when registration failed

Currently, if registration fails due to a bad password (401 or 407 error codes) the UI shows "SIP Registration failure" but it also shows the boxes for users to try and dial or chat.

Maybe the other functionality should be disabled when registration is in this state?

Maybe the error should give a link to log in again? Currently, the user can go back to the login form by clicking the logout link (provided in the fix for #23) but it would be nice to make it more obvious that the user should try to login again.

can't correct a bad password

If the user types a bad password, the UI displays a message saying that the registration failed but it doesn't give the user a way to return to the login form.

As a workaround, the user can refresh the page.

If the login form is not enabled in the configuration, then the password/registration failure message should be displayed as usual and it should not send the user to the login form as they never would have seen the login form anyway.

webcam: full screen

Show the full screen control by default in the webcam viewer

Does not show local video while dialing

When dialing a video calling and waiting for the other party to answer, local video is not displayed and there is no indication that we are making a video call other than the webcam light being illuminated.

can't use delete button in chat

When typing a chat message, I found the backspace button works but the delete button does not.

specify a timeout for connectivity checks

Currently, the connectivity checks keep running if UDP is blocked. Amongst other things, this means the callee continues to hear ringing for a long time after clicking to answer.

Usually, if UDP is not blocked, the checks will complete in less than 2 seconds. Probably need a default timeout value between 2 - 5 seconds.

event-demo.html: allow clicking URIs from previous calls/missed calls

The event-demo code creates a table of previous calls (including outgoing calls and missed calls)

Making the URIs in this table clickable could make it behave like a redial facility.

Login form, remember-me checkbox: advise that password not cached

It may be useful to display some visual clue to advise the user that only their username and display name will be cached but their password will never be cached if they click the remember-me checkbox.

disable rather than hide the phone when websocket link disconnects

Currently, when the websocket link is disconnected, the phone is hidden

Maybe it is better to simply disable it (make it all appear gray) so the user can still see what they were doing (e.g. the number they were dialing)

support other major languages

Make versions of the HTML for other major languages (Spanish, French, German, Italian, Chinese, ...)

How does jQuery best handle internationalization?

create Wordpress plugin

Create a Wordpress plugin using JSCommunicator, similar to the DruCall plugin for Drupal

chat: reject messages from peer when chat disabled

There is now a config.js option to enable/disable the chat area in the UI.

If the chat UI is disabled then there is no way to display any incoming chat messages. Therefore, they should be rejected with a SIP error code in the handler for the newMessage event.

Query about this sent to the JsSIP mailing list:
https://groups.google.com/forum/#!topic/jssip/4rzuI6A3amo

save username and display name in local storage

JSCommunicator provides a way for users to log in with arbitrary SIP credentials (see the way it is used at https://rtc.debian.org )

Each time a user visits the site, they have to type

their display name (e.g. Daniel Pocock)
their SIP address (e.g. [email protected])
their password

We should avoid keeping the user password in any cache or local storage. However, maybe the display name and SIP address can be saved in local storage. It may be a good idea to provide a checkbox where the user can confirm that they want those details remembered in the browser they are using.

add messaging support

Use the jQuery tabbed panel to create a chat messaging facility within JSCommunicator

See the way messages are sent and received in http://tryit.jssip.net for API details of the send message function and the received message callback.

Login form, remember-me checkbox: add configuration option

It would be useful to have a configuration option to enable/disable this checkbox

show the display name of the peer

When dialing or receiving a call, it would be useful to provide a way to show the display name of the peer.

Disable dialing buttons if destination is blank

Disable the dialing buttons if the destination box is blank and enable them when the user starts typing something in the box or when the application inserts a value there.

support for SIP instance-id, flush existing registrations

Investigate how we can support SIP instance ID and clear other registrations from the same user.

This was discussed on the JsSIP mailing list

make DTMF feedback tones optional

When the user presses a dialpad button, a DTMF tone is played.

We should make this feedback optional in the config.

Breaks with newer jQuery-UI v1.9+

The phone works with jQuery-UI 1.8.21 but it broke when trying jQuery-UI 1.9

require_relay_candidate not working

When require_relay_candidate is true and no relay candidate is found in the SDP, an error is logged but the call appears to try and proceed anyway. It then appears to fail with a 407 error.

clear dialing box when calling?

Offer a config option to automatically clear the dialing box when making a call

It should probably not clear the box if the edit permissions on the box are disabled and the user is locked into dialing a single number or if the box is invisible

Please enable content negotiation

In order to get the desired language, content negotiation should be used before the select menu.

Both Registered/Not Registered messages display

Apparently when a user is not registered (or if there is a SIP registration failure) both the messages Registered/Not Registered are displayed:

clear video frame after call finishes

When the user makes a second call, the video frame (both local and remote video frames) appear with the last images from the previous call.

integration with local audio settings

Find out if there is some way to integrate with the local audio settings / control panel so that the user can see if their microphone is disabled and see if the gain is set to a very low value. It would be even better if the user could be given a control to change these settings or open the control panel provided by their OS.

create a MediaWiki plugin

Create a MediaWiki plugin using JSCommunicator, similar to the DruCall plugin for Drupal

Login form, username/display name: provide default values

It may be useful to provide default values to appear in these boxes that have a lower priority than any remembered values.

The values specified by display_name and uri are not default values as such as they always override any value that the user has remembered.

Default values, however, should not override remembered values, they should only be used when remembered values don't exist at all.

Does not perform ICE connectivity checks before ringing

JsSIP is not sending back a 183 with ICE candidates in response to an INVITE.

It just sends back 180 and ringing starts.

It only does ICE checks after answering, this is not consistent with the ICE protocol.

Consequently, people sometimes experience missing audio/video streams. This should never happen.

The issue needs to be addressed with JsSIP rather than JSCommunicator.

create a Joomla! plugin

Create a Joomla! plugin using JSCommunicator, similar to the DruCall plugin for Drupal

provide visual indication of local audio level

There is currently no visual indication if the local microphone is detecting audio.

Some other softphones display a little meter or gauge that moves up and down to indicate when the local user is speaking. This is very helpful for detecting situations where the microphone is broken, not connected to the right socket or the microphone input is disabled in the OS.

TypeMismatchError when making a call

Hello,

I tried to contact the list, but I'm getting the email returned, so trying here...

I'm testing JSCommunicator in an Ubuntu 13.10 machine, and I'm getting the account registered in a kamailio server. But when I try to make an audio call, I get an error, a JS exception, and I'm kind of stuck with it.

Firefox and Chromium show different errors, but for the same script (JSComm.js) and line (29).
Moreover, I always get this message in the web: "ERROR: failed to start call, check that microphone/webcam are connected, check browser security settings, peer may not support compatible codecs", but no popup or browser message appears requesting the access to the microphone.

Thanks in advance.

When using Firefox, I get this:

"make_call failed: Error: RTCPeerConnection constructor passed invalid RTCConfiguration - malformed URI: " JSComm.js:29

When using Chromium, I get this error:

make_call failed: TypeMismatchError: The type of an object was incompatible with the expected type of the parameter associated to the object. JSComm.js:29
DOMException {message: "The type of an object was incompatible with the ex…d type of the parameter associated to the object.", name: "TypeMismatchError", code: 17, stack: "Error: The type of an object was incompatible with…ttp://localhost/JSCommunicator/jquery.js:5:10866)", INDEX_SIZE_ERR: 1…}
code: 17
message: "The type of an object was incompatible with the expected type of the parameter associated to the object."
name: "TypeMismatchError"
stack: "Error: The type of an object was incompatible with the expected type of the parameter associated to the object.↵ at Object.RTCMediaHandler.init (http://localhost/JSCommunicator/JsSIP.js:3389:27)↵ at Object.RTCMediaHandler (http://localhost/JSCommunicator/JsSIP.js:3283:8)↵ at Object.RTCSession.connect (http://localhost/JSCommunicator/JsSIP.js:4192:26)↵ at Object.UA.call (http://localhost/JSCommunicator/JsSIP.js:5170:11)↵ at Object.window.JSCommManager.make_call (http://localhost/JSCommunicator/JSComm.js:29:250)↵ at Object.window.JSCommUI.make_call (http://localhost/JSCommunicator/JSComm.js:37:280)↵ at HTMLInputElement. (http://localhost/JSCommunicator/JSComm.js:32:479)↵ at HTMLInputElement.x.event.dispatch (http://localhost/JSCommunicator/jquery.js:5:14129)↵ at HTMLInputElement.v.handle (http://localhost/JSCommunicator/jquery.js:5:10866)

Regards,
David N.

ERROR: SIP Registration failure.

Hi,

I spent the day trying to get webrtc going, but now I am stuck.

my setup:
Basically I followed http://www.rtcquickstart.org/ with small
variations. I skipped however the DNS setup stuff.
I am using repro as sip proxy. I can make calls with jitsi and empathy.

jsscommunicator does not seem to work for me. I always get the
SIP Registration on the phone.shtml pagefailure.

I have increased the log level of repro and it complains too:

REGISTER sip:spirou.hh.de SIP/2.0
Via: SIP/2.0/WS c4nuv9hulu9f.invalid;branch=z9hG4bK9564909;received=192.168.168.97
Max-Forwards: 69
Contact: <sip:[email protected];transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:722f588d-e0de-43a4-a72b-2c9973011dc7>";expires=3600
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=qivdl4evkp
Call-ID: 1mopiqpdgdrucf1td56cq6
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, UPDATE, MESSAGE, OPTIONS
Supported: path, gruu, outbound
User-Agent: JsSIP 0.5.0
Content-Length: 0


DEBUG | 20141218-165938.945 | repro | RESIP:TRANSACTION | 140341293410048 | TransactionUser.cxx:87 | Match!
DEBUG | 20141218-165938.945 | repro | RESIP | 140341293410048 | Helper.cxx:374 | Helper::makeResponse(SipReq:  REGISTER spirou.hh.de tid=9564909 cseq=1 REGISTER [email protected] / 1 from(wire) code=100 reason=
DEBUG | 20141218-165938.945 | repro | RESIP:TRANSACTION | 140341293410048 | TimerQueue.cxx:50 | Adding timer: Timer Trying tid=9564909 ms=3500
DEBUG | 20141218-165938.945 | repro | RESIP:TRANSACTION | 140341293410048 | TuSelector.cxx:70 | Send to TU: TU: DialogUsageManager size=0

REGISTER sip:spirou.hh.de SIP/2.0
Via: SIP/2.0/WS c4nuv9hulu9f.invalid;branch=z9hG4bK9564909;received=192.168.168.97
Max-Forwards: 69
Contact: <sip:[email protected];transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:722f588d-e0de-43a4-a72b-2c9973011dc7>";expires=3600
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=qivdl4evkp
Call-ID: 1mopiqpdgdrucf1td56cq6
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, UPDATE, MESSAGE, OPTIONS
Supported: path, gruu, outbound
User-Agent: JsSIP 0.5.0
Content-Length: 0


INFO | 20141218-165938.945 | repro | REPRO:APP | 140341268231936 | AclStore.cxx:558 | AclStore - source address NOT trusted: 192.168.168.97:47721 WS
DEBUG | 20141218-165938.945 | repro | RESIP | 140341268231936 | Helper.cxx:374 | Helper::makeResponse(SipReq:  REGISTER spirou.hh.de tid=9564909 cseq=1 REGISTER [email protected] / 1 from(wire) code=407 reason=
INFO | 20141218-165938.945 | repro | RESIP:DUM | 140341268231936 | ServerAuthManager.cxx:460 | Sending challenge to SipReq:  REGISTER spirou.hh.de tid=9564909 cseq=1 REGISTER [email protected] / 1 from(wire)
DEBUG | 20141218-165938.945 | repro | RESIP:DUM | 140341268231936 | DialogUsageManager.cxx:983 | SEND:

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/WS c4nuv9hulu9f.invalid;branch=z9hG4bK9564909;received=192.168.168.97
Proxy-Authenticate: Digest nonce="1418918378:24df2c666f01f1bccd6c9d526229c961",algorithm=MD5,realm="spirou.hh.de",qop="auth,auth-int"
To: <sip:[email protected]>;tag=8310bf65
From: <sip:[email protected]>;tag=qivdl4evkp
Call-ID: 1mopiqpdgdrucf1td56cq6
CSeq: 1 REGISTER
User-Agent: repro 1.9.7
Content-Length: 0


DEBUG | 20141218-165938.945 | repro | RESIP | 140341268231936 | SipStack.cxx:594 | SEND: SipResp: 407 tid=9564909 cseq=1 REGISTER / 1 from(tu)
INFO | 20141218-165938.945 | repro | RESIP:DUM | 140341268231936 | ServerAuthManager.cxx:47 | ServerAuth challenged request SipReq:  REGISTER spirou.hh.de tid=9564909 cseq=1 REGISTER [email protected] / 1 from(wire)
DEBUG | 20141218-165938.945 | repro | RESIP:TRANSPORT | 140341293410048 | TransportSelector.cxx:1178 | Transmitting to [ V4 192.168.168.97:47721 WS target domain=unspecified mFlowKey=32 ] tlsDomain= via [ V4 192.168.168.21:5062 WS target domain=unspecified mFlowKey=0 ]

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/WS c4nuv9hulu9f.invalid;branch=z9hG4bK9564909;received=192.168.168.97
Proxy-Authenticate: Digest nonce="1418918378:24df2c666f01f1bccd6c9d526229c961",algorithm=MD5,realm="spirou.hh.de",qop="auth,auth-int"
To: <sip:[email protected]>;tag=8310bf65
From: <sip:[email protected]>;tag=qivdl4evkp
Call-ID: 1mopiqpdgdrucf1td56cq6
CSeq: 1 REGISTER
User-Agent: repro 1.9.7
Content-Length: 0

sigcomp id=
DEBUG | 20141218-165938.946 | repro | RESIP:TRANSPORT | 140341285017344 | TcpBaseTransport.cxx:283 | Processing write for [ V4 192.168.168.97:47721 WS target domain=unspecified mFlowKey=32 ]
DEBUG | 20141218-165938.946 | repro | RESIP:TRANSPORT | 140341285017344 | ConnectionManager.cxx:65 | Found fd 32
DEBUG | 20141218-165938.961 | repro | RESIP:TRANSPORT | 140341285017344 | Transport.cxx:382 | incoming from: [ V4 192.168.168.97:47721 WS target domain=unspecified mFlowKey=32 ]
DEBUG | 20141218-165938.961 | repro | RESIP:TRANSPORT | 140341285017344 | Connection.cxx:387 | Connection::performReads()  read=812
DEBUG | 20141218-165938.961 | repro | RESIP:TRANSACTION | 140341293410048 | TransactionUser.cxx:79 | Checking if SipReq:  REGISTER spirou.hh.de tid=5578936 cseq=2 REGISTER [email protected] / 2 from(wire) is for me
DEBUG | 20141218-165938.962 | repro | RESIP:TRANSACTION | 140341293410048 | TransactionUser.cxx:84 | Checking rule...
DEBUG | 20141218-165938.962 | repro | RESIP:TRANSACTION | 140341293410048 | MessageFilterRule.cxx:45 | Matching rule for:

REGISTER sip:spirou.hh.de SIP/2.0
Via: SIP/2.0/WS c4nuv9hulu9f.invalid;branch=z9hG4bK5578936;received=192.168.168.97
Max-Forwards: 69
Contact: <sip:[email protected];transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:722f588d-e0de-43a4-a72b-2c9973011dc7>";expires=3600
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=qivdl4evkp
Call-ID: 1mopiqpdgdrucf1td56cq6
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, UPDATE, MESSAGE, OPTIONS
Proxy-Authorization: Digest algorithm=MD5, username="[email protected]", realm="spirou.hh.de", nonce="1418918378:24df2c666f01f1bccd6c9d526229c961", uri="sip:spirou.hh.de", response="106f41f6aee674371a89c9b6b6571e34", qop=auth, cnonce="hqrhaad85lcv", nc=00000001
Supported: path, gruu, outbound
User-Agent: JsSIP 0.5.0
Content-Length: 0


DEBUG | 20141218-165938.962 | repro | RESIP:TRANSACTION | 140341293410048 | TransactionUser.cxx:87 | Match!
DEBUG | 20141218-165938.962 | repro | RESIP | 140341293410048 | Helper.cxx:374 | Helper::makeResponse(SipReq:  REGISTER spirou.hh.de tid=5578936 cseq=2 REGISTER [email protected] / 2 from(wire) code=100 reason=
DEBUG | 20141218-165938.962 | repro | RESIP:TRANSACTION | 140341293410048 | TimerQueue.cxx:50 | Adding timer: Timer Trying tid=5578936 ms=3500
DEBUG | 20141218-165938.962 | repro | RESIP:TRANSACTION | 140341293410048 | TuSelector.cxx:70 | Send to TU: TU: DialogUsageManager size=0

REGISTER sip:spirou.hh.de SIP/2.0
Via: SIP/2.0/WS c4nuv9hulu9f.invalid;branch=z9hG4bK5578936;received=192.168.168.97
Max-Forwards: 69
Contact: <sip:[email protected];transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:722f588d-e0de-43a4-a72b-2c9973011dc7>";expires=3600
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=qivdl4evkp
Call-ID: 1mopiqpdgdrucf1td56cq6
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, UPDATE, MESSAGE, OPTIONS
Proxy-Authorization: Digest algorithm=MD5, username="[email protected]", realm="spirou.hh.de", nonce="1418918378:24df2c666f01f1bccd6c9d526229c961", uri="sip:spirou.hh.de", response="106f41f6aee674371a89c9b6b6571e34", qop=auth, cnonce="hqrhaad85lcv", nc=00000001
Supported: path, gruu, outbound
User-Agent: JsSIP 0.5.0
Content-Length: 0


INFO | 20141218-165938.962 | repro | RESIP:DUM | 140341268231936 | ServerAuthManager.cxx:402 | Requesting credential for [email protected] @ spirou.hh.de
INFO | 20141218-165938.962 | repro | RESIP:DUM | 140341268231936 | ServerAuthManager.cxx:53 | ServerAuth requested credentials SipReq:  REGISTER spirou.hh.de tid=5578936 cseq=2 REGISTER [email protected] / 2 from(wire)
DEBUG | 20141218-165938.962 | repro | REPRO:APP | 140341335373568 | UserAuthGrabber.hxx:48 | Grabbed user info for [email protected]@spirou.hh.de :
DEBUG | 20141218-165938.962 | repro | RESIP:TRANSACTION | 140341335373568 | TuSelector.cxx:70 | Send to TU: TU: DialogUsageManager size=0

UserAuthInfo [email protected] @ spirou.hh.de A1=
INFO | 20141218-165938.962 | repro | RESIP:DUM | 140341268231936 | ServerAuthManager.cxx:131 | Checking for auth result in realm=spirou.hh.de A1=
INFO | 20141218-165938.962 | repro | RESIP:DUM | 140341268231936 | ServerAuthManager.cxx:136 | User unknown [email protected] in spirou.hh.de
DEBUG | 20141218-165938.962 | repro | RESIP | 140341268231936 | Helper.cxx:374 | Helper::makeResponse(SipReq:  REGISTER spirou.hh.de tid=5578936 cseq=2 REGISTER [email protected] / 2 from(wire) code=404 reason=User unknown.
DEBUG | 20141218-165938.962 | repro | RESIP:DUM | 140341268231936 | DialogUsageManager.cxx:983 | SEND:

SIP/2.0 404 User unknown.
Via: SIP/2.0/WS c4nuv9hulu9f.invalid;branch=z9hG4bK5578936;received=192.168.168.97
To: <sip:[email protected]>;tag=a53ad358
From: <sip:[email protected]>;tag=qivdl4evkp
Call-ID: 1mopiqpdgdrucf1td56cq6
CSeq: 2 REGISTER
User-Agent: repro 1.9.7
Content-Length: 0


DEBUG | 20141218-165938.962 | repro | RESIP | 140341268231936 | SipStack.cxx:594 | SEND: SipResp: 404 tid=5578936 cseq=2 REGISTER / 2 from(tu)
INFO | 20141218-165938.962 | repro | RESIP:DUM | 140341268231936 | ServerAuthManager.cxx:112 | ServerAuth rejected request UserAuthInfo [email protected] @ spirou.hh.de A1=
DEBUG | 20141218-165938.962 | repro | RESIP:TRANSPORT | 140341293410048 | TransportSelector.cxx:1178 | Transmitting to [ V4 192.168.168.97:47721 WS target domain=unspecified mFlowKey=32 ] tlsDomain= via [ V4 192.168.168.21:5062 WS target domain=unspecified mFlowKey=0 ]

SIP/2.0 404 User unknown.
Via: SIP/2.0/WS c4nuv9hulu9f.invalid;branch=z9hG4bK5578936;received=192.168.168.97
To: <sip:[email protected]>;tag=a53ad358
From: <sip:[email protected]>;tag=qivdl4evkp
Call-ID: 1mopiqpdgdrucf1td56cq6
CSeq: 2 REGISTER
User-Agent: repro 1.9.7
Content-Length: 0

sigcomp id=
DEBUG | 20141218-165938.962 | repro | RESIP:TRANSPORT | 140341285017344 | TcpBaseTransport.cxx:283 | Processing write for [ V4 192.168.168.97:47721 WS target domain=unspecified mFlowKey=32 ]
DEBUG | 20141218-165938.962 | repro | RESIP:TRANSPORT | 140341285017344 | ConnectionManager.cxx:65 | Found fd 32

´´´


However I have a user "web" added to the repro-webinterface. Users "win" and "lin" work in jitsi and empathy. The line UserAuthInfo [email protected] @ spirou.hh.de looks funny...

but I have no idea howto fix this.

start sending DTMF as soon as user presses button

At present,

DTMF is sent after the button click is released
the DTMF duration is a fixed value (not the length of the button press)

As it is a fixed duration, we should send the DTMF indication as soon as the user presses the button

Chat window too small

I can barely see 3 lines of chat. Please make it a way bigger. Also, best would be if it was resizeable, or automatically matching the window size.

display an alert if other active registrations exist

SIP allows multiple simultaneous registrations

Stateful proxies can fork INVITEs to all registered UAs but this is not always desirable.

It would be useful to show the user if other registrations are still active, this may alert them to the fact that they are still logged in from another PC, another browser or another tab.

webcam: customize size

Make the size of the webcam display customizable

add captcha support to prevent spam calls

Provide some example of how to implement captcha

This probably involves using cookies, the cookie support in the repro SIP proxy could help:

http://www.resiprocate.org/SIP_Over_WebSocket_Cookies

UPDATE TRANSLATIONS - French, Spanish and German plz

With new additions to JSCommunicator, some labels are missing translations. Specifically the last 10 or so highlighted: https://github.com/opentelecoms-org/jscommunicator/blob/master/internationalization/Messages.properties#L36-L45