{"id":15725938,"url":"https://github.com/altanai/unifiedcommunicator","last_synced_at":"2025-05-13T01:35:17.751Z","repository":{"id":13316419,"uuid":"16003021","full_name":"altanai/unifiedCommunicator","owner":"altanai","description":"Rich Communication services (RCS) integration with Enterprise Unified Communicator on sipml5  (webRTC)  ","archived":false,"fork":false,"pushed_at":"2019-12-16T11:46:48.000Z","size":35640,"stargazers_count":29,"open_issues_count":0,"forks_count":8,"subscribers_count":11,"default_branch":"master","last_synced_at":"2025-04-01T05:41:21.009Z","etag":null,"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"],"latest_commit_sha":null,"homepage":"https://telecom.altanai.com/2013/10/02/webrtc-solution/","language":"JavaScript","has_issues":true,"has_wiki":null,"has_pages":null,"mirror_url":null,"source_name":null,"license":null,"status":null,"scm":"git","pull_requests_enabled":true,"icon_url":"https://github.com/altanai.png","metadata":{"files":{"readme":"README.md","changelog":null,"contributing":null,"funding":null,"license":null,"code_of_conduct":null,"threat_model":null,"audit":null,"citation":null,"codeowners":null,"security":null,"support":null}},"created_at":"2014-01-17T15:20:52.000Z","updated_at":"2024-10-10T08:15:43.000Z","dependencies_parsed_at":"2022-08-29T10:31:15.551Z","dependency_job_id":null,"html_url":"https://github.com/altanai/unifiedCommunicator","commit_stats":null,"previous_names":[],"tags_count":0,"template":false,"template_full_name":null,"repository_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/altanai%2FunifiedCommunicator","tags_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/altanai%2FunifiedCommunicator/tags","releases_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/altanai%2FunifiedCommunicator/releases","manifests_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/altanai%2FunifiedCommunicator/manifests","owner_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/owners/altanai","download_url":"https://codeload.github.com/altanai/unifiedCommunicator/tar.gz/refs/heads/master","host":{"name":"GitHub","url":"https://github.com","kind":"github","repositories_count":253855029,"owners_count":21974399,"icon_url":"https://github.com/github.png","version":null,"created_at":"2022-05-30T11:31:42.601Z","updated_at":"2022-07-04T15:15:14.044Z","host_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub","repositories_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories","repository_names_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repository_names","owners_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/owners"}},"keywords":["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"],"created_at":"2024-10-03T22:25:05.852Z","updated_at":"2025-05-13T01:35:17.701Z","avatar_url":"https://github.com/altanai.png","language":"JavaScript","readme":"# Unified Communicator and Collaborator for Enterprise\n\nModular 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.\n\nCurrent modules include \n\n* Conferencing                   \n* Geolocation\n* WebRTC_presentation\n* Emoticons                      \n* Importcontacts                 \n* UploadPicsAudioVideo           \n* apache-tomcat-7.0.42           \n* presentation_server_config.txt\n* EnhancedCallLogs\u0026Analytics     \n* Notifications                  \n* ViewPicsAudioVideo             \n* callscreening                  \n* experimental features\n* Offlinemail                    \n* Voicemail                      \n* database_server_config.txt     \n* workspace\n\n![ucc component diagram ](img/ucc_component_diagram.png)\n\nTechnologies used :\n* Java EE ( Enterprise Edition )\n* Apaache WebSerer\n* Front end technologies ( Javascript , CSS , Html)\n* Google Maps API\n* SQL backend for Database\n* mysql JSDB driver \n* Directory Integration with Google contacts \n\n## Installation and setup\n\n1. Install Mysql databse \n\n2. Install Apache tomcat \n\n3. Install SIP serveltes Server\n\n4. Install Apache Server \n\n5. Host webproject on apache server\n\n## Getting started \n\n-- tbd --\n\n## Release Notes \n\n### Version 1 :\n\n- Single Sign On\n- Login with id and password to access all services\n- Audio / Video Call\n    - Call Hold / Call Transfer\n- Messaging:\n    - SIP Instant Messaging\n    - Message to Facebook Messenger\n    - Message delivered as Email\n- Chatroom\n    - group chat between multiple users . Room is created for set of users .\n- Video Conferencing\n    - video chat between multiple parties . Room is created for set of users .\n- File Transfer\n    - Sharing of files from local to remote , in peer-to-peer and broadcasting fashion .\n- Third party Webservices\n    - Widgets like calendar , weather , stocks , twitter are embedded.\n- Visual Voice Mail\n    - Record and deliver voice message to recipients voice mail inbox which can be accessed/ played from web client .\n- Phonebook \n    - cloud integration\n    - add new entries\n    - add photos to contacts identity\n    - import contacts from google account\n- Click to Call :\n    - Drop down list of contacts form mail call console\n    - 2 step Click to call from Phonebook\n- Presence :\n    - Publish online / offline status\n    - Use Subscribe / notify requests of SIP\n- Web Ssocket  to SIP Gateway\n    - Conversion between the signal coming from the  WebRTC  and SIP client  to the  IMS core\n    - Conversion of  “voice/video \" media  between sRTP and RTP\n    - Conversion of other media (data channel) towards MSRP and Transcoding.\n    - Support of ICE procedure\n    - Implementation of a STUN server\n- QoS Support  \n\n\n### Version 2 :\n\n- Logs\n    - calls logs\n    - Message logs\n- User Profile\n    - user details like address , email and social networking accounts\n    - Phonenumber for GSM integration through SMS\n    - User's Media storage like Pictures , profile picture , Audio , video\n    - File sharing documents storage for future access in the same format\n- Real Time and Offline Analytics\n- service usage with graphical and tabular history  trends\n- Session Management\n    - Single Sign-on\n    - Forgot password regeneration using secure question\n    - Registration of new user account\n    - Logout and clearance of session parameters\n- Security\n    - No redirection to any page through url entry without valid session\n    - No going back to home page after logout by back button on browser\n    - No data vulnerability\n    - Multiple login through different devices handled\n- OAuth\n    - Login via IMAP / token through facebook and Google\n- Phonebook with Presence functionality inbuilt\n- Directory Service based on country / region\n- Geolocation of approximate location detection of device logged in and visibility to  others\n\n### Version 3 :\n\n- Integration with new age CSP deployments like VoLTE, ViLTE, VoWiFi \n- Multi vendor support\n- Interactive webrtc services \n- Media Services \n    - Automated Natural language Speech recognition\n    - Semantic processing via ML \n    - Enhanced incall services replacing IVR ( touch -tone) \n    - VQE (voice Quality Enhancements) \n    - Encoding and Decoding - Multiple Codec Support\n    - Transcoding\n    - Silence Suppression\n- Security via TLS, encryption and AAA\n- Http, NFS caching \n- NAT using Xirsys TURN  \n- Recording, playback and media file compression  \n- active frame selection\n- DTMF (Dual Tone Multi Frequency)\n    - SIP info messages (out-of-band)\n    - SIP notify messages (out-of-band)\n    - Inband DTMF not supported yet \n- Audio \n    - mixing \n    - announcements ( VXML, MSML )\n    - filters \n    - gain control ( AGC using webrtc stack)\n    - noise suppresesion ( webrtc stack)\n    - speakers notification\n    - Narrowband, Wideband, and Super Wideband\n    - dynamic sample rate\n- Video  \n    - continuous presence ( Face detetion ) \n    - floor control\n    - video lipsync (sync)\n    - speaker tile selection \n- VQE (Voice Quality Enhancement )\n    - Acoustic Echo Cancelation\n    - noise reduction\n    - noise line detection\n    - noise gating\n    - Packet Loss concealment \n- Call analyics \n    - progress analysis   \n    - MOS , R-factor ( derived from latency , jitter , packet loss )\n- CDR (Call detail records ) and accounting \n- Lawful interception\n","funding_links":[],"categories":[],"sub_categories":[],"project_url":"https://awesome.ecosyste.ms/api/v1/projects/github.com%2Faltanai%2Funifiedcommunicator","html_url":"https://awesome.ecosyste.ms/projects/github.com%2Faltanai%2Funifiedcommunicator","lists_url":"https://awesome.ecosyste.ms/api/v1/projects/github.com%2Faltanai%2Funifiedcommunicator/lists"}