Yate is a next-generation telephony engine. While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, robbed bit, ISDN PRI, BRI, and SS7. YateClient is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.
Ekiga (formely known as GnomeMeeting) is a soft phone, video conferencing, and instant messenger application for use over the Internet. It supports HD sound quality and video up to DVD size and quality. It is interoperable with many other standards compliant software, hardware, and service providers as it uses both of the major telephony standards, SIP and H.323.
The GNU Gatekeeper is a free H.323 gatekeeper based on the OpenH323 project. You can use it to manage a Voice-over-IP network and let endpoints (e.g., Netmeeting) communicate through symbolic names. It also has an external interface for billing and other applications. It runs on a number of Unix versions (including Linux and Solaris) and Windows.
Bayonne is the telephony server of the GNU project. It offers a script-driven threaded multi-line state event telephony service on GNU/Linux, xBSD, and Microsoft Windows for building voice response systems, and uses telephony plugins for runtime driver configuration. It also features "TGI" for making Perl applications "telephony aware". It may be used to build telephony-based system administration, home automation, automated attendant, v-commerce, and voice messaging systems.
GNU SIP Witch is a secure peer-to-peer VoIP server. Calls can be made even behind NAT firewalls, and without requiring service providers. SIP Witch can be used on the desktop to create bottom-up secure calling networks and as a free software alternative to Skype. It can also be used as a stand-alone SIP-based office telephone server, or to create secure VoIP networks for an existing IP-PBX such as Asterisk, FreeSWITCH, or Yate.
PhoNetInfo retrieves detailed phone and network information. It runs on Symbian "Belle", S^3/"Anna", and S60 5th and 3rd edition mobile phones. Information about the following topics can be retrieved: Phone manufacturer and model, firmware version, battery level, WLAN and bluetooth MAC, bluetooth device class, IMSI, IMEI, subscriber ID, charger status, running tasks, active profile settings, network mode, network signal strength and cell ID, network name and ID, network country code and registration status, CPU speed, CPU type and architecture, size of RAM and ROM, time since last reboot, and more. All information can be saved to a file.
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.
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.
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.