Linphone is an audio and video Internet phone with GTK+ and console interfaces. It uses the SIP protocol, and is compatible with most SIP clients and gateways. It can use various audio and video codecs such as Speex, GSM, G711, G722, ilbc, amr, Theora, H263-1998, MPEG4, H264, VP8, and snow.
oRTP is a library implementing the Real-time Transport Protocol (RFC3550), written in C. It is easy to use and provides a packet scheduler for sending and receiving packets on time, adaptive jitter compensation, automatic sending of RTCP compound packets, and the RTCP parser API. It works with IPv6.
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.
bcg729 is an Open Source implementation of both the encoder and decoder of the ITU G729 Annex A speech codec. It is written in C 99, is fully portable, and can be executed on many platforms, including both ARM and x86. libbcg729 supports concurrent channel encoding/decoding for multi-call applications such as conferencing. This project was initially developed as part of Mediastreamer2, Linphone's media processing engine, so it contains the glue to be integrated into Linphone/Mediastreamer2.
Re: WEB ? Phone
This is exaclty what I've understood.
I will change the presentaion lines on freshmeat one
of these days.
Re: WEB ? Phone
> why is it called web phone? Does it rely
> on any HTTP
Linphone uses the Session Initiation Protocol (rfc
2543), and Real Time Protcol (rfc 1889). Http is
inadequat. I agree that "internet phone" is better than