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Java SIP softphone
This project forked from ymartineau/peers
Java SIP softphone
============================================================ Peers: java sip softphone http://peers.sourceforge.net/ ============================================================ LICENSE This software is released under GPL License version 3 or any later version. Please read this license in gpl.txt if not already done. SPECIFICATION Peers is a software phone (softphone) compatible with the following specifications: - RFC 3261 (SIP), - RFC 4566 (SDP), - RFC 3550 (RTP), - RFC 3551 (RTP Audio/Video profile), - RFC 2617 (Digest Authentication), - RFC 4733 (DTMF), - ITU-T G.711 (PCMU, PCMA) You should easilly retrieve those specifications on internet with your favorite search engine. PREREQUISITES This software has been developed using Oracle Java Development Kit version 7. You should install the latest Java Runtime Environment on your computer if you just want to run the application. If you want to compile the sources yourself, you should use the JDK. In both cases, you can download the installation files here: http://www.oracle.com/technetwork/java/javase/downloads/index.html CONFIGURATION Your SIP account credentials can be configured in conf/peers.xml. Please read comments in this file for configuration details. This configuration file is ruled by a grammar file: peers.xsd. Thus to modify this file, you can use jEdit with XML and Error list plugins. You can download jEdit here: http://www.jedit.org/index.php?page=download This provides xml completion and grammar checking which can be very useful to avoid simple configuration errors. You can also configure your SIP account using graphical user interface. RUNNING If you are a Windows user you can use the .bat batch script in root directory, if you use any Unix compatible sytem, you can use the .sh script. You can also double click .jar file. You can then call any IP address using SIP protocol, if the remote host does not listen on the default SIP port (5060), you can use the following example URI: sip:192.168.1.2:6060 For some softphones, it is necessary to add a userpart to the called sip uri, for example: sip:[email protected]:6060 If you configured a sip account in configuration file or using gui, you can also place calls with usual sip uris: sip:[email protected] Advanced users can run several peers instances on the same computer. In this case a folder should be created in peers root directory for each peers instance. This folder should contain three directories: conf, logs and media. conf should contain peers.xml and peers.xsd for this instance. peers.xml will need to be updated with this instance parameters, peers.xsd can be copied from root conf directory. You will need to do this for each instance. <media> parameter in configuration file should be activated for at most one peers instance, this avoids comfusion in microphone capture and sound playback. You should also check that SIP and RTP ports are not the same in each configuration file. Here is an example configuration: peers/ user1/ conf/ peers.xml peers.xsd logs/ media/ user2/ conf/ peers.xml peers.xsd logs/ media/ Once all those files have been created and updated, each instance can be run providing a java system property giving peers home directory: java -classpath build/classes -Dpeers.home=user1 net.sourceforge.peers.gui.MainFrame As a Main-Class has been defined in jar manifest, you can also use the following command line: java -jar build/peers.jar -Dpeers.home=user1 net.sourceforge.peers.gui.MainFrame HISTORY 2007-11-25 0.1 First release minimalist UAC and UAS. 2007-12-09 0.1.1 First release update moved startup scripts in root directory 2008-03-29 0.2 Second release New features: - provisional responses (UAC and UAS), - CANCEL management (UAC and UAS), updated GUI with provisional stuff - new Logger which enables network traffic tracing and classical log4j-like logging in two separate files Bugs fixed: - 1900810 MTU too small management 2008-06-08 0.3 Third release New features: - register management (initial register, register refresh, unregister) - authentication using message digest (RFC2617) Improved features: - media capture/sending using pipes and three threads - using TestNG for tests - no singleton is used anymore - xxxRequestManagers and xxxMethodHandlers are instanciated only once for uas and uac - provisional responses can create or update dialog info (remote target, etc.) Bugs fixed: - 1994625 provisional responses with to-tag 2009-09-23 0.3.1 Peers resurrection New features: - Running: - peers.home system property to run peers in several environments - Configuration: - an outbound proxy can now be configured in configuration file - media can now be activated or deactivated in configuration file - SIP: - support "sent-by" and "received" Via parameters - support 407 Proxy-Authenticate on REGISTER - support 401 and 407 on INVITE - support re-INVITEs (refresh target) - manage challenges on INVITEs - RTP: - support remote party update (ip address and port) Improved features: - transport log file now contains real remote ip address and port - fixed media sending issue (replaced encoder with mobicents media server g711 encoder) 2010-12-13 0.4 Peers GUI New features: - GUI: - update sip account settings in account frame with no modification in files - SIP: - 3102136 keep-alives sent and answered - 3109472 added rport management based on RFC 3581 - 3031364 added OPTIONS method management - RTP: - 2934223 support RFC 4733 outgoing telephone-events (DTMF) - 3107231 added PCMA (G711 A-law) codec Improved features: - gui complete refactoring, now using clean events between sip core and gui Bugs fixed: - 3032080 no media on pickup - 2864885 multiple challenged calls - 3098214 start CSeq at 1 instead of 0 2011-05-19 0.4.1 Bug fixes Bugs fixed: - GUI: - 3155571 tuijldert: update dialog state before gui notification - SIP: - 3137803 tuijldert: client transaction synchronization on responses reception - 3285107 added authentication on bye request 2011-07-04 0.4.2 Bug fixes Bugs fixed: - SIP: - 3325451 cannot hangup with asterisk 1.8.4.2 - 3324115 ACK does not contain Authorization header 2011-09-06 0.4.3 Various improvements New features: - SIP: - manage opaque parameter in challenge - use same tag in From header between original and authenticated INVITE - support incoming INVITEs and reINVITEs with empty body Bugs fixed: - SIP: - 3392342 bad CANCEL sip request when other party doesn't picking up 2014-05-01 0.5 Maven release - moved source code from sourceforge subversion repository to git hosted on github - now using maven with multiple modules to build peers - updated doc - simplified SIP API AUTHOR Yohann Martineau [email protected]
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