VoIP, conferencing and messaging

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Asterisk

http://www.asterisk.org FLOSSMETRICS link: http://melquiades.flossmetrics.org/projects/asterisk

Asterisk is the world’s leading open source telephony engine and tool kit. Asterisk can be configured as the core of an IP or hybrid PBX, switching calls, managing routes, enabling features, and connecting callers with the outside world over IP, analog (POTS), and digital (T1/E1) connections. Asterisk runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris; it can also be built out as the heart of a media gateway, bridging the legacy PSTN to the expanding world of IP telephony. Asterisk’s modular architecture allows it to convert between a wide range of communications protocols and media codecs. Several projects add a web interface to provide simplified administration, for example FreePBX and TrixBox.

Among the features:

  • ADSI On-Screen Menu System
  • Alarm Receiver
  • Append Message
  • Authentication
  • Automated Attendant
  • Blacklists
  • Blind Transfer
  • Call Detail Records
  • Call Forward on Busy
  • Call Forward on No Answer
  • Call Forward Variable
  • Call Monitoring
  • Call Parking
  • Call Queuing
  • Call Recording
  • Call Retrieval
  • Call Routing (DID & ANI)
  • Call Snooping
  • Call Transfer
  • Call Waiting
  • Caller ID
  • Caller ID Blocking
  • Caller ID on Call Waiting
  • Calling Cards
  • Conference Bridging
  • Database Store / Retrieve
  • Database Integration
  • Dial by Name
  • Direct Inward System Access
  • Distinctive Ring
  • Distributed Universal Number
  • Predictive Dialer
  • Privacy
  • Open Settlement Protocol (OSP)
  • Overhead Paging
  • Protocol Conversion
  • Remote Call Pickup
  • Remote Office Support
  • Roaming Extensions
  • Route by Caller ID
  • SMS Messaging
  • Spell / Say
  • Streaming Media Access
  • Supervised Transfer
  • Talk Detection
  • Text-to-Speech (via Festival)
  • Three-way Calling
  • Time and Date
  • Transcoding
  • Trunking
  • VoIP Gateways
  • TDMoE (Time Division Multiplex over Ethernet)
  • Allows for integration of physically separate installations
  • Uses commonly deployed data connections
  • Allows a unified dialplan across multiple offices
  • supports FXS, FXO, MF and DTMF signalling, 4ESS and BRI.


DimDim

http://www.dimdim.com FLOSSMETRICS link: http://melquiades.flossmetrics.org/projects/dimdim FLOSSMETRICS quality evaluation: http://melquiades.flossmetrics.org/projects/dimdim/quality

DimDim is a web-based conferencing solution. Among the features:

  • Presentation and Document sharing: Interactive realtime collaboration over documents and presentations allows enhanced expression and exchange of ideas.
  • Audio and video sharing: High-quality multi-party video and audio sharing can be used to personalize meetings with a face-to-face approach.
  • Application sharing: Full screen as well as specific application sharing from a Presenter’s computer can be used to show and educate even a novice audience.
  • White board and Annotations: Realistic interactive collaboration which involves a lot of annotations, corrections, group drawing (and doodles in the margin) is enabled through digital whiteboard and annotations feature of Dimdim.
  • Chat: As in any real world meeting there will be sub-groups of people engaged in conversation and exchange of ideas (some serious ones others more like “kicks under the table”) within the larger assembly in a web conference. This is facilitated through the multi-user chat feature.
  • Polls: Polls enable the presenter to gauge the mood of the participants and to take decisions considering the opinions of many.
  • Question manager: Question manager which is like a moderated Q&A setup enables the presenter to better manage the interaction.
  • Record and Archive: All the interaction is recorded and archived for sharing with non-attendees and to reach a broader audience.

Screenshots


Ekiga

http://www.ekiga.org/ FLOSSMETRICS link: http://melquiades.flossmetrics.org/projects/ekiga FLOSSMETRICS quality evaluation: http://melquiades.flossmetrics.org/projects/ekiga/quality

Ekiga (formely known as GnomeMeeting) is an open source VoIP and video conferencing application for GNOME. Ekiga uses both the H.323 and SIP protocols. It supports many audio and video codecs, and is interoperable with other SIP compliant software and also with Microsoft NetMeeting. Among the features:

  • SIP Compliant
  • Registrar Support
  • Possibility to Simultaneously Register to Several Accounts
  • Proxy Support
  • Outbound Proxy Support
  • Call Hold
  • Call Transfer
  • Call Forwarding on No Answer, on Busy, Always
  • Configurable Port Ranges
  • Instant Messaging
  • DTMFs Support
  • Message Waiting Indications Support
  • ENUM Support
  • Transparent NAT Support, Assisted NAT Support (STUN, IP Translation)
  • SIP re-INVITE support
  • iLBC, GSM-06.10, MS-GSM, G.711-Alaw, G.711-uLaw, G.726, G.721 and Speex Audio Codecs
  • H.261 QCIF Video Code
  • Dynamic Jitter Buffer
  • Dynamic Threshold Algorithm for Silence Detection
  • Echo Cancellation
  • Wideband Codec Support
  • Automatic Video Bandwidth Limitation
  • Video Transmission/Reception Control

Screenshots Screenshots


OpenFire

http://www.jivesoftware.com/products/openfire/index.jsp

Openfire is an instant messaging (IM) and groupchat server that uses the XMPP protocol. It provides:

  • Web-based administration panel
  • Plugin interface
  • Customizable
  • SSL/TLS support
  • User-friendly web interface and guided installation
  • Database connectivity (i.e. embedded Apache Derby or other DBMS with JDBC 3 driver) for storing messages and user details
  • LDAP connectivity
  • Platform independent, pure Java
  • Full integration with Spark Jabber client

The companion Spark client provides an easy and efficient interface to OpenFire, with drag&drop file transfer, VoIP, multi-protocol chat and IM.

Screenshots


OpenMeetings

http://code.google.com/p/openmeetings/

Multi-Language Customizable Video-Conferencing and Collaboration. It features:

  • Video/Audio
  • See Desktop of any participant
  • Multi-Language and Customizable
  • Whiteboard with drawing, write & edit, dragNDrop, Resizeing, Images (DragNDrop from Library), Symbol(s)
  • Conference while drawing (4x4 or 1xn modus)
  • Safe Drawings / whiteboard and load it next time, edit and resave
  • ImportDocuments: .tga, .xcf, .wpg, .txt, .ico, .ttf, .pcd, .pcds, .ps, .psd, .tiff, .bmp, .svg, .dpx, .exr, .jpg, .jpeg, .gif, .jpg, .ppt, .odp, .odt, .sxw, .wpd, .doc, .rtf, .txt, .ods, .sxc, .xls, .sxi, .pdf
  • Send invitation and direct Links into a meeting
  • Moderating System
  • User-/Organisation System
  • Private and Public (Organisation only) Conference-Rooms

It works through a AJAX/Flash client that requires no client download in most situations.


OpenWengo

http://www.openwengo.com/

A SIP-based multiplatform VoIP client, similar to Skype.

Screenshots


SIPX

http://www.sipfoundry.org/

A complete SIP-based VoIP PBX system. It features:

  • Transfer (consultative & blind)
  • Call coverage
  • Call hold / retrieve
  • Consultation hold
  • Music on Hold for IETF standards compliant phones
  • Uploadable music file
  • 3-way conference
  • Call pickup (global and directed call pickup)
  • Call park & retrieve
  • Hunt groups
  • SIP URI dialing
  • CLID (Calling Line Identification)
  • CNIP (Calling party Name Identification Presentation)
  • CLIP (Call Line Identification Presentation)
  • CLIR (Call Line Identification Restriction)
  • Per gateway CLIP manipulation
  • Call waiting / retrieve
  • Do not Disturb (DnD)
  • Forward on busy, no answer, do not disturb
  • Multiple line appearances
  • Multiple calls per line
  • Multiple station appearance
  • Outbound call blocking
  • Click-to-dial (Windows XP)
  • Redial
  • Call history (dialed, received, missed)
  • Auto off-hook / ring down
  • Incoming only
  • Peer-to-peer media routing for best quality (media not routed through the sipX server)
  • Unmatched voice quality with lowest delay and jitter
  • Support for any codec supported by the phone (including video)
  • Support for Polycom HD Voice
  • Codec negotiation (no transcoding required)
  • Numeric or alpha-numeric User ID
  • User PIN management (UI or TUI)
  • Aliasing facility (numeric and alpha-numeric aliases)
  • Extension and alias uniqueness assurance
  • Granular per user permissions
  • Call permissions
  • Dial plans
  • directory services
  • softkeys
  • speed dial
  • PSTN trunking
  • least-cost routing
  • SIP trunking
  • analog lines support
  • unlimited numer of simultaneous calls
  • fully redundant
  • Call Detail Records collection and reporting
  • Plug & Play Device Management
  • voicemail
  • presence features
  • hunt groups
  • call center server


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