SIP (Session Initiation Protocol) is the IETF (Internet Engineering Task Force) signaling standard for managing multimedia session initiation defined in RFC 3261 commonly used in VOIP communication. SIP can be used to initiate voice, video and multimedia sessions, for both interactive applications (e.g. an VOIP phone calls or a video conference applications) and non-interactive ones (e.g. video streaming).
The mjSIP stack has been used in research activities by Dept. of Engineering and Architecture at University of Parma and by DIE - University of Roma "Tor Vergata" and several commercial products.
mjSIP includes all classes and methods for creating SIP-based applications. It implements the complete layered stack architecture as defined in RFC 3261 (Transport, Transaction, and Dialog layers), and is fully compliant with RFC 3261 and successive standard RFCs. Moreover it includes higher level interfaces for Call Control and User Agent implementations. mjSIP comes with a core package implementation that includes:
sip
, server
, ua
, and supporting modules net
, sound
, and util
. Extracted
examples into modules examples
and phone
. args4j
, applied Java naming conventions, encapsulated
fields. Access configuration from production code through read-only interfaces.slf4j
over tinylog
.DTMF
info messages.RTP
media streams.SDP
messages with well-known formats that are not explained in rtpmap
fields. mjSIP is available open source under the terms of the GNU GPL license (General Public Licence) as published by the Free Software Foundation.
The project's original home page is at: http://mjsip.org/
There are several independent forks of the project on Github: