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

1. Install Mysql databse

2. Install Apache tomcat

3. Install SIP serveltes Server

4. Install Apache Server

5. Host webproject on apache server

-- tbd --

• 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

• 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

• 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