l1kmakes/sipml5-ng

A modernised version of the SIPml5 WebRTC library

Our Implementation - Interactive Web Phone Application

Our organization is adopting this project as part of a transition to an open-source and maintainable communications stack. Our changes in this repository intend to take this reference implementaion to a usable product for our organization and other organizations. In support of this the API is being updated to utilize more modern APIs, for example code to support HTML5 screen sharing has come a long way since this was originally written and plugins/WebKit API's are no longer required. While it is being built to be targeted for our environment and stack, we are keeping modularity in mind so other organizations could use this with their own requirements.

Our environment uses Asterisk as a SIP back end and has many web resources protected by SAML (specifically simplesamlphp), which are key integration points for this implementation. The SAML login pre-populates SIP attributes which are defined in Active Directory, and if the passcode is stored in the optional MySQL databse, automatic SIP registration happens by providing credentials in the background, so all other SIP parameters (realm, websocket server, proxy) will be provided by a configuration file and will NOT be user configurable. Effectively, a user can register with SIP using their AD Credentials. Where possible we made everything configurable; settings are hidden if they're set in the config but available in a pop-out menu if they are not set.

While moving settings to a single static config file, we will also attempt to modularize theming and metadata with the hope that most configuration can be placed in just the config.php file, which will make implementation by other groups MUCH easier.

We have added sms chat and threaded conversations, chat persistence (if configured with the optional MySQL DB), custom shortcut buttons, user-editable custom shortcuts (if configured with the optional MySQL DB), pop-up notifications indicating incoming chats or calls, searchable AD contact lookups, and a more responsive mobile UI.

Stretch goals for our organization are to implement presence indication and group SMS support.

At this time we cannot provide a public demo of our fork as it is protected behind ActiveDirectory logins through SAML, however the demo below from duobango still works for external SIP testing. The video below is a demonstration of our implementation.

Installation Requirements

Our implementation requires the following:

1. Linux Server (Tested on Ubuntu 20.04.1)
2. apache2 web server (Tested on 2.4.41)
3. php (Tested on 7.4.3)
4. php-mysqli (Optional for shortcuts, persistent chat and auto-registration via SAML)
5. mysql-server (Tested on 8.0.23, optional for shortcuts, persistent chat and auto-registration via SAML)
6. php-curl and php-dom (PHP modules for optional simplesamlphp integration)

Optional - MySQL setup

Optionally, if a MySQL server is configured, several features become enabled:

1. If AD/LDAP login is enabled through simpleSAMLPHP, the passcode-extension pairs can be stored for automatic SIP registration; this way a user only needs to log in once (through AD/LDAP) and not a second time at SIP registration.
2. Chat/SMS persistence between sessions, chats are written to the database and can be recalled in subsequent sessions
3. Shortcut storage, users can configure custom shortcut buttons

A mysql server can be set up on the server hosting this app, or on another system. All DB configuration is defined in config.php. To set up the tables and columns, please run the sql commands in mysql_db_setup.sql. After that, to add a user to the system, run the following in mysql:

INSERT INTO sipml5_web.extensions (extension,passcode) VALUES ('1234','1234');

SIPml5-NG

On May 14th, 2012 SIPml5, the world's first open Source HTML SIP client was released. SIPml5 had captivated the mind of RTC pioneers in the open source communities. However, as time pregressed, its creator Doubango Telecom had abandoned the project. On Feb 8th, 2018 Doubango Telecom had released their final version of SIPml5 (version 2.1.4) - and since then the code base had remained unmaintained.

On June 17th, 2020 Cloudonix released its fork of the original SIPml5 project - SIPml5-NG. The new project picks up the project from that point and merges back to the project various patches and updates, provided by the Open Source community and the various SIPml5 developer community.

SIPml5-NG is an open source (BSD license) HTML5 SIP client entirely written in javascript for integration in social networks (FaceBook, Twitter, Google+), online games, e-commerce websites, email signatures... No extension, plugin or gateway is needed. The media stack relies on WebRTC.

The client can be used to connect to any SIP or IMS network from your preferred browser to make and receive audio/video calls and instant messages.

Javascript SIP/SDP stack

The SIP and SDP stacks (~1 Mo) are entirely written in javascript and the network transport uses WebSockets as per rfc7118. The live demo doesn't require any installation and can be used to connect to any SIP server using UDP, TCP or TLS transports.

Short but not exhaustive list of supported features:

• Works on Chrome, Firefox, IE, Safari, Opera and Bowser
• Audio / Video call
• Screen/Desktop sharing from Chrome to any SIP client
• Instant messaging
• Presence
• Call Hold / Resume
• Explicit Call transfer
• Multi-line and multi-account
• Dual-tone multi-frequency signaling (DTMF) using SIP INFO
• Click-to-Call
• SIP TelePresence (Video Group chat)
• 3GPP IMS standards

Media Stack

The media stack depends on WebRTC (Web Real Time Communication) which is natively supported by the following browsers:

• Google Chrome
• Firefox
• Safari
• Microsoft Edge

Interoperability

Using SIPml5-NG and cloudonix.io you can call any SIP-legacy endpoint or connected with any SIP compatible network.

Our testing and verification process includes testing using the following WebRTC/VoIP tools:

• SIP over WebSocket Servers

  • Asterisk
  • Freeswitch
  • Kamailio
  • OpenSIPS

• SIP over WebSocket Endpoints

  • webrtc.cloudonix.io

• Desktop Browsers

  • Google Chrome
  • Firefox
  • Safari
  • Edge

• Mobile Browsers

  • Google Chrome
  • Firefox
  • Safari

The following browsers are known to be non-compatible or do not support WebRTC natively:

• Opera
• Facebook Mobile Browser
• Mobile browsers provided by low end mobile vendors

Where did the Asterisk patches go?

As of version 12 of the Asterisk project, support for SIP over WebSocket is native to the project. As the patches were required for older versions, that are no longer supported or available - we had decided to remove that directory.

License

The code is released under BSD terms. For more information: https://github.com/cloudonix/sipml5-ng/blob/wiki/License.md

How to contribute to the project

If you have a patch for this project, please submit a pull request. Pull requests will be tested and verified for proper functionality and stability. Once a patch had been accepted - it will be merged back into the upstream repository.

Project home page: github.com/L1kMakes/sipml5-ng
Download JS API: SIPml-api.js

Credits/Acknowledgements

Duobango Telecom, original source for this project, the API, and the demo: Duobango Telecom Duobango Telecom GitHub

Cloudonix, who forked SIPml5 into SIPML5-ng, the fork we are using: Cloudonix Cloudonix Github

WebRTC Experiment by Muaz Khan, whose screen-sharing code was referenced adapted into this: Muaz Khan Github WebRTC-Experiment Plugin-free WebRTC Screen Sharing

sipml5-ng's Issues

Documentation: Unable to hold after completing an initial hold/resume

Creating this issue as documentation in our repository of what is an upstream issue.

If you make an audio call, you can place it on hold, then resume it. Afterwards, if you try to put the call on hold again, it does NOT go on hold and the hold button grays out.

The expected result is that you should be able to hold/resume as many times as you want in a call.

We discovered this when implementing features for multi-call switching. Worried that our modifications to the sipml5 api caused this so we reverted to the upstream doubango api js and experienced the issue as well, then we fully implemented doubango's reference UI and api and again experienced the issue.

We have submitted upstream bugs to the 3 parent repositories (one of which has issues disabled, so I emailed the dev:

• DoubangoTelecom#360
• cloudonix#4

We are also attempting to fix this ourselves, however the nuance of the sip api is slightly outside our expertise.

Waiting call and conference call

the waiting call keep response busy,
Is the multiline ability is not available or something configuration missing ?

No audio incoming Firefox 103.0

Hello, the incoming audio of the call does not work in Firefox 103.0