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https://github.com/altanai/unifiedcommunicator

Rich Communication services (RCS) integration with Enterprise Unified Communicator on sipml5 (webRTC)
https://github.com/altanai/unifiedcommunicator

call-analytics callscreening communication conferencing cpaas enterprise geolocation javaee phonebook sipml5 spring-mvc tomcat turn ucc voicemail voip webrtc webrtc-analytucs webrtc-javascript-library webrtc-signaling

Last synced: 13 days ago
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Rich Communication services (RCS) integration with Enterprise Unified Communicator on sipml5 (webRTC)

Host: GitHub
URL: https://github.com/altanai/unifiedcommunicator
Owner: altanai
Created: 2014-01-17T15:20:52.000Z (almost 11 years ago)
Default Branch: master
Last Pushed: 2019-12-16T11:46:48.000Z (almost 5 years ago)
Last Synced: 2024-10-25T13:42:41.719Z (21 days ago)
Topics: call-analytics, callscreening, communication, conferencing, cpaas, enterprise, geolocation, javaee, phonebook, sipml5, spring-mvc, tomcat, turn, ucc, voicemail, voip, webrtc, webrtc-analytucs, webrtc-javascript-library, webrtc-signaling
Language: JavaScript
Homepage: https://telecom.altanai.com/2013/10/02/webrtc-solution/
Size: 34 MB
Stars: 29
Watchers: 12
Forks: 7
Open Issues: 0
Metadata Files:
- Readme: README.md

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README

# Unified Communicator and Collaborator for Enterprise

Modular enterprise communicator solution for enterprise based communication and collaboration . Use sipml5 client side libarary to provide webRTC based media stream capture and propagration from client side without external plugins.

Current modules include

* Conferencing
* Geolocation
* WebRTC_presentation
* Emoticons
* Importcontacts
* UploadPicsAudioVideo
* apache-tomcat-7.0.42
* presentation_server_config.txt
* EnhancedCallLogs&Analytics
* Notifications
* ViewPicsAudioVideo
* callscreening
* experimental features
* Offlinemail
* Voicemail
* database_server_config.txt
* workspace

![ucc component diagram ](img/ucc_component_diagram.png)

Technologies used :
* Java EE ( Enterprise Edition )
* Apaache WebSerer
* Front end technologies ( Javascript , CSS , Html)
* Google Maps API
* SQL backend for Database
* mysql JSDB driver
* Directory Integration with Google contacts

## Installation and setup

1. Install Mysql databse

2. Install Apache tomcat

3. Install SIP serveltes Server

4. Install Apache Server

5. Host webproject on apache server

## Getting started

-- tbd --

## Release Notes

### Version 1 :

- Single Sign On
- Login with id and password to access all services
- Audio / Video Call
- Call Hold / Call Transfer
- Messaging:
- SIP Instant Messaging
- Message to Facebook Messenger
- Message delivered as Email
- Chatroom
- group chat between multiple users . Room is created for set of users .
- Video Conferencing
- video chat between multiple parties . Room is created for set of users .
- File Transfer
- Sharing of files from local to remote , in peer-to-peer and broadcasting fashion .
- Third party Webservices
- Widgets like calendar , weather , stocks , twitter are embedded.
- Visual Voice Mail
- Record and deliver voice message to recipients voice mail inbox which can be accessed/ played from web client .
- Phonebook
- cloud integration
- add new entries
- add photos to contacts identity
- import contacts from google account
- Click to Call :
- Drop down list of contacts form mail call console
- 2 step Click to call from Phonebook
- Presence :
- Publish online / offline status
- Use Subscribe / notify requests of SIP
- Web Ssocket to SIP Gateway
- Conversion between the signal coming from the WebRTC and SIP client to the IMS core
- Conversion of “voice/video " media between sRTP and RTP
- Conversion of other media (data channel) towards MSRP and Transcoding.
- Support of ICE procedure
- Implementation of a STUN server
- QoS Support

### Version 2 :

- Logs
- calls logs
- Message logs
- User Profile
- user details like address , email and social networking accounts
- Phonenumber for GSM integration through SMS
- User's Media storage like Pictures , profile picture , Audio , video
- File sharing documents storage for future access in the same format
- Real Time and Offline Analytics
- service usage with graphical and tabular history trends
- Session Management
- Single Sign-on
- Forgot password regeneration using secure question
- Registration of new user account
- Logout and clearance of session parameters
- Security
- No redirection to any page through url entry without valid session
- No going back to home page after logout by back button on browser
- No data vulnerability
- Multiple login through different devices handled
- OAuth
- Login via IMAP / token through facebook and Google
- Phonebook with Presence functionality inbuilt
- Directory Service based on country / region
- Geolocation of approximate location detection of device logged in and visibility to others

### Version 3 :

- Integration with new age CSP deployments like VoLTE, ViLTE, VoWiFi
- Multi vendor support
- Interactive webrtc services
- Media Services
- Automated Natural language Speech recognition
- Semantic processing via ML
- Enhanced incall services replacing IVR ( touch -tone)
- VQE (voice Quality Enhancements)
- Encoding and Decoding - Multiple Codec Support
- Transcoding
- Silence Suppression
- Security via TLS, encryption and AAA
- Http, NFS caching
- NAT using Xirsys TURN
- Recording, playback and media file compression
- active frame selection
- DTMF (Dual Tone Multi Frequency)
- SIP info messages (out-of-band)
- SIP notify messages (out-of-band)
- Inband DTMF not supported yet
- Audio
- mixing
- announcements ( VXML, MSML )
- filters
- gain control ( AGC using webrtc stack)
- noise suppresesion ( webrtc stack)
- speakers notification
- Narrowband, Wideband, and Super Wideband
- dynamic sample rate
- Video
- continuous presence ( Face detetion )
- floor control
- video lipsync (sync)
- speaker tile selection
- VQE (Voice Quality Enhancement )
- Acoustic Echo Cancelation
- noise reduction
- noise line detection
- noise gating
- Packet Loss concealment
- Call analyics
- progress analysis
- MOS , R-factor ( derived from latency , jitter , packet loss )
- CDR (Call detail records ) and accounting
- Lawful interception