MyConnection Server is broadband testing software which measures connections for bandwidth speeds and connection quality for time critical applications such as VoIP, Video conferencing, and IPTV. It helps organizations assess networks for deployment of new/additional services and identify and resolve last mile customer connectivity problems with little need for the customer to assist in the resolution process. A network route testing component details the path of the connections and where packet loss and latency occur, including discovery of multiple routes to a destination. Remote Test Agents enable technical staff to customize and interactively manage the bandwidth testing process and perform extended quality testing over hours or days to address and resolve intermittent problems as required. Satellite Servers establish additional connection testing points at the application edge to accurately test actual application network paths.
Oreka is an enterprise telephony recording and retrieval system with a Web-based user interface. It supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP, and audio sound devices, and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems, such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc. It is being used in call centers and contact centers for quality monitoring (QM) purposes.
Sofia-SIP is a SIP user agent library, compliant with the IETF RFC3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services. The primary target platform is GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center.
VOCP is a complete messaging solution for voice modems, with voicemail, fax, email pager, DTMF command shell and Text-to-Speech support, 4 graphical interfaces, and a Web interface. Callers navigate the system using a touch-tone phone and may send and receive faxes, voice mail, and pager messages, listen to text/HTML email messages, or execute configured programs on the host and hear the resulting output.
chan-sccp-b is an extension of the original chan-sccp implementation for the Asterisk soft-PBX. It lets you hook up a Cisco/SCCP Phone to your Asterisk server using the SCCP protocol, which works a lot better than the SIP firmware on the same phone. It provides full phone functionality instead of just a simple SIP channel provider. It offers functionality like shared lines, hotline functionality, guest login, dynamic speeddials, private line automatic ring-down (PLAR), personal softkey configurations, Dundi support, SCCP extended dialplan functions, manager support, and custom device state buttons.
WengoPhone is a multi-platform VOIP client sponsored and developed by WENGO and MBDSYS. The GUI part is Qt-based, and the Video-over-IP engine is based on the eXosip, oSIP, oRTP, and ffmpeg projects. The eXosip module is extended by a phApi module, which implements a high-level, easy-to-use call control API. WengoPhone supports PC-to-PC voice, video, and chat. One can use a standard SIP service provider such as Wengo to be assigned an incoming number, make calls to PSTN/cell phones, get voice mail, and more.
Mediastreamer is a portable C library that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM, and AMR), video codecs (MPEG4, H263, H264, and Theora), sound card I/O, wav file streaming, webcam video capture, echo-cancellation, conferencing, parametric equalization, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.
Gizmo Project is a program that uses advanced VoIP technology to allow you to talk with your friends around the world. Gizmo Project uses your Internet connection to make calls to people using other computers. It is simple, and allows you to talk clearly and for as long as you want.
Twinkle is a software phone for voice over IP communications using the SIP protocol. You can use it for direct IP phone to IP phone communication, or in a network using a SIP proxy to route your calls. Some of the features offered are call waiting, call hold, 3-way conference call, call transfer, and call reject. It supports STUN or a statically configured public IP address for NAT traversal. When using STUN, it will send keep-alive packets to keep NAT bindings alive. It supports ZRTP for secure voice communication.