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

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