MCS MyVoIP very accurately measures the quality and performance of Internet connections for Voice over IP (VoIP) usage by simulating UDP voice data traffic between a server and browser clients. Connections are tested for jitter and packet loss and rated for the supported level of sound quality. The VoIP test can be set to various codecs or customized by packet size, packet rate, and test length. The test can further be combined with a bandwidth speed test or network route diagnostics for more in-depth connection analysis.
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.
Sipp is a performance testing tool for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC & UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It also reads XML scenario files describing any performance testing configuration. It features the dynamic display of statistics about running tests, periodic CSV statistics dumps, TCP, UDP, or TLS over IPv4 or IPv6 over multiple sockets or multiplexed with retransmission management, regular expressions and variables in scenario files, conditional branching, and dynamically-adjustable call rates. RTP play (voice, video, and RFC2833 DTMFs) is also supported.
asterisk-oh323 adds H.323 support to the ASTERISK soft PBX. It does this by interfacing the OpenH323 library to ASTERISK through a loadable module. The package provides the channel driver as well as a wrapper in a shared library form. It is able to initiate and receive calls to and from H.323 endpoints, and has been successfully tested with the H.323 terminals on the OpenH323 site (ohphone, openphone), GnomeMeeting, Microsoft NetMeeting, Cisco routers, H.323 Snom phones, and other hardware and software H.323 IP phones.
Jiplet Container (Java SIP Servlet) is a servlet-like development and runtime environment for SIP applications. The SIP protocol is widely used for voice services over IP networks. This product enables developers to create server-side SIP applications using a component-based model similar to that envisioned by the J2EE architecture. The Jiplet container runs as a standalone server as well as a JBOSS service.