Jingle Nodes is a relay auto-discovery service that provides Jingle Relay Type Candidates that can be used in ICE-UDP and also on RAW-UDP Jingle sessions. Relay candidates can provide NAT traversal for users who don't have STUN/TURN Support, but also for users with STUN/TURN support for whom negotiation failed. It is designed to allow you to communicate freely with your friends without being attached to closed service providers like Skype or telecommunications carriers.
Billion PBX is IP phone system that uses the SIP protocol. In addition to voice calls it supports call forwarding, voicemail, callback, and much more, and features simple installation and configuration, reliability, low hardware requirements, and a flexible pricing policy. Its main features are subscriber registration by SIP Registrar and RAS protocols, call routing by SIP and H.323, call authorization, subscriber self-service, Call Data Records (CDR), embedded rating and billing, integration with external billing systems, NAT traversal, IVR, report generation, and a convenient administrative Web interface allowing remote management.
Billion Softswitch works as a switchboard in SIP/H.323 VoIP networks, proxying both signalling and media streams while performing cross-protocol conversion and even media stream transcoding. It integrates advanced call routing features as well as embedded compact billing system. It supports subscriber registration by SIP Registrar and RAS, call authorization, call data record (CDR) list maintenance, external billing system interfaces, NAT/firewall support, and report generation, and is easy to install and maintain.
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.
HOMER is a robust, carrier-grade, scalable SIP capturing system and monitoring application with hEP, IP Proto4 (IPIP) encapsulation, and port mirroring/monitoring support right out of the box, ready to process and store large amounts of signaling with instant searches, end-to-end analysis, and drill-down capabilities for ITSPs, VoIP providers, and trunk suppliers using SIP signaling.
baresip is a bare-bones SIP user agent. It supports SIP, SDP, RTP/RTCP, and STUN/TURN/ICE, and IPv4 and IPv6, and is RFC-compliant and has portable C89 and C99 source code. A modular plugin architecture provides stdio, cons, and evdev user interfaces, celt, g711, g722, gsm, ilbc, l16, and speex audio codecs, alsa, coreaudio, gst, portaudio, oss, winwav, and mda audio drivers, speex_pp, speex_aec, speex_resamp, and sndfile audio filters, the avcodec video codec, avformat, quicktime, qtcapture, v4l, and v4l2 video sources, sdl, opengl, and x11 video display drivers, and srtp media encoding.
SEMS is a media and application server for SIP based VoIP services. It shows good performance doing basic services like announcements and conference for combination with external application servers. Thanks to its easy-to-use and flexible application development framework and back-to-back user agent support, application logic and media serving can be combined in the same process. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox, and lots of example applications are available. Scripting can be done in Python and a simple state machine description language. Support All commonly used free codecs (including g711, gsm, iLBC, speex, adpcm, and l16) are supported. Other features include wideband, ZRTP encryption, a SIP registrar client, an XMLRPC server/client, and a DIAMETER client.