Free sip software for linux

SIPp is a free Open Source test tool / traffic generator for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods.
SIPp project can also reads custom XML scenario files describing from very simple to complex call flows. It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management and dynamically adjustable call rates.
Other advanced features include support of IPv6, TLS, SIP authentication, conditional scenarios, UDP retransmissions, error robustness (call timeout, protocol defense), call specific variable, Posix regular expression to extract and re-inject any protocol fields, custom actions (log, system command exec, call stop) on message receive, field injection from external CSV file to emulate live users.
While optimized for traffic, stress and performance testing, SIPp can be used to run one single call and exit, providing a passed/failed verdict.
Last, but not least, SIPp has a comprehensive documentation available both in HTML and PDF format.
SIPp can be used to test many real SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, SIP PBX, ... It is also very useful to emulate thousands of user agents calling your SIP system.
Enhancements:
- New: Statistical (conditional) branching feature. See
- http://sipp.sf.net/doc/reference.html#Randomness+in+conditional+branching.
- New: 3PCC extended mode - see
- http://sipp.sourceforge.net/doc/reference.html#3PCC+Extended
- Tool: monitor remote SIP servers through SNMP - see
- http://sipp.sourceforge.net/wiki/index.php/Getting_feedback_from_the_server
- Enh: extends the -aa option to UPDATE messages
- Enh: changes in the compilation with external libs - useful for the package
- generation system
- Enh: Allow sampling from a Weibull distribution for pause duration
- Enh: use stat_delimiter for the trace_rtt option and include number that is
- being reported.
- Enh: Add repeat_rtd keyword for repeated RTD calculations
- Enh: Handle stripping Control-M characters from multi-valued headers in
- get_header
- Enh: Update the clock_tick more frequently so that we have a higher timer and
- statistics resolution
- Enh: for option that take a time as argument - allow them to be specified using
- seconds or milliseconds
- Enh: fail when parsing a scenario that has pcap if pcap is not enabled
- Enh: print the actual location of the error log file and the error condition
- (if any) on creation
- Enh: fail when parsing a scenario that enables authentication if SSL is not
- enabled
- Enh: Makefile - include EXTRAENDLIBS keyword so that libraries can be appended
- to the list after SSL Enh: Add regexp_match argument to the receive command for
- universal catching
- Enh: remote control: increase the allowed number of control sockets.
- Enh: 3pcc extended: clean the reach of the allowed number of local twin sockets
- Enh: amelioration of statistic computing
- Enh regexp: add case_indep, occurence and start_line options for the hdr
- matching case
- Enh: Stats: Use RTDs that are precise to the microsecond in -trace_rtt, and
- improve the consistency between trace_rtt and the averages
- Enh: Only include RTDs that are actually used in the CSV output
- Enh: Allow loss percentages less than 1 and also a global command line option to
- specify that packets should be lost at a given percentage
- Fix: fix for -t un error for non-IPv6 platform like win32 - Unable to bind UDP
- socket, errno = 106 (Address family not supported by protocol)
- Fix: Do not initialize the screen library in background mode
- Fix: allow having an optional recv before a recvCmd
- Fix: Authentication: allow [fieldn] values for the [authentication] field
- Fix: updated support of short header forms
- Fix: small bug fix to the micrortt.diff which is required for the initial call
- rate to work properly
- Fix:Allow the code to compile with -Wall -Werror on Linux
- Fix: empty line was generated when [routes] keyword was used and proxy did not
- record-route
- Fix: pcap on HPUX; Fix: Simple fixes identified with valgrind
- Fix: 3pcc extended: problem when quitting
- Fix: regexp: add a warning when the specified header is not present in the
- received message
- Fix: pcap: destroy the send packets thread properly even if the sendto failed
- Fix: bug in regexp due to an incomplete commit (rev 172) - remove some debugging
- traces in message log file
- Fix: bugs on retransmission counters and on cookies for optional messages
- Fix: Incorrect branch in automatic ACK answering to unexpected >= 400 responses,
- as well as automatic CANCEL - in such cases, branch must be identical to the
- branch of the initial INVITE request
- Fix: 3pcc extended: clean up when screen exit. Fix: stop logging when the
- maximum allowed file size is reached (avoid core dump)
- Fix: incomplete Via header in automatic responses when aborting calls
- Fix: -h: -key parameter
- Fix: 3pcc/3pcc extended modes: closes twin sockets properly when twin instance
- exits, to break the poll() loop and avoid the print of empty messages
- Fix: in 3pcc/3pcc extended modes: send BYE/CANCEL before exit due to other twin
- instance exit
- Fix: force exit when pressing q twice (Q press can still be used)
- Fix: Aka authentication for synchro case, also added password len as password
- for authentication might contain char Fix: possible core dump in SDP parser
- Fix: accept up to the 5 defined RTDs (previously would only accept one less)
- Fix: Fail if there is an invalid repartition table specification - previous
- behavior was a core dump
- Fix: added -users option to the parameter table
- Fix: change option description to match timed options
- Fix: trace_err did not work in background mode
- Fix: bug when testing the presence of the 3PCC compilation flag
- Fix: bug in -r -rp option

One of the features of Python that makes it so powerful is the ability to take existing libraries, written in C or C++, and make them available as Python extension modules. Such extension modules are often called bindings for the library.
SIP is a tool that makes it very easy to create Python bindings for C and C++ libraries. Python-SIP was originally developed to create PyQt, the Python bindings for the Qt toolkit, but can be used to create bindings for any C or C++ library.
SIP comprises a code generator and a Python module. The code generator processes a set of specification files and generates C or C++ code which is then compiled to create the bindings extension module. The SIP Python module provides support functions to the automatically generated code.
The specification files contains a description of the interface of the C or C++ library, i.e. the classes, methods, functions and variables. The format of a specification file is almost identical to a C or C++ header file, so much so that the easiest way of creating a specification file is to edit the corresponding header file.
SIP makes it easy to exploit existing C or C++ libraries in a productive interpretive programming environment. SIP also makes it easy to take a Python application (maybe a prototype) and selectively implement parts of the application (maybe for performance reasons) in C or C++.
Enhancements:
- This release adds support for consolidated and composite modules.
- It adds support for pickling classes and enums.

Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification.
Sofia-SIP project can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services.
The primary target platform for Sofia-SIP is GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center. Sofia-SIP is licensed under the LGPL.
Main features:
SIP features
- Sofia-SIP implementation follows RFC3261 and related key RFCs. INFO, UPDATE and REFER methods are supported. Also supported is SIMPLE presence and instant messaging, with the MESSAGE, SUBSCRIBE/NOTIFY and PUBLISH methods. Features such as early sessions, provisional responses, early media, caller preferences and session timers are included. Full set of transports, including both TCP and UDP over either IPv4 or IPv6, are supported.
SIP Offer-Answer module
- Sofia-SIP provides an implementation of the SDP offer-answer negotiation as specified in RFC3264. This is an essential component in using SIP to establish media sessions such as VoIP and video conferencing.
NAT traversal support
- Support for STUN as specified in RFC3489. STUN functionality is available via a separate module, so it can also be used independently from the base SIP stack. SIP extensions such as symmetric response routing (RFC3581/rport) are supported as well.
SIP security support
- Signaling can be secured by use of SSL/TLS. Also HTTP basic and digest authentication methods are supported.

Minisip project is a SIP User Agent ("Internet telephone") developed at KTH currently running on Linux. Keywords: Secure VoIP; SIP; MIKEY; RTP; SRTP; SDP; Video Telephony; Push-to-talk. You can download it for free from the download page.

Minisip is developed by Ph.D and Master students at the Royal Institute of Technology, KTH, Stockholm, Sweden.

The source code is available as a number of libraries under the GNU Lesser General Public License (LGPL) and applications under the GNU General Public Licence (GPL).

SSIP-GST Gaim plugin is an open-source SIP/SIMPLE plugin library,
compliant with the IETF RFC3261 specification. SSIP-GST plugin serves as an example GUI client for the Sofia-SIP library.

It can be used with Gaim as a SIP client software for uses such as VoIP, IM and presence. Media support is integrated using GStreamer, and is merged from sofsip-cli command line example client for Sofia-SIP user agent library.

SSIP-GST is developed on top of Sofia-SIP, which is based on a SIP stack developed at the Nokia Research Center. SSIP-GST plugin is licensed under the GPL.

SSIP-GST aims to leverage features of the cool Gaim UI for Sofia-SIP library usage.

Siproxd is a proxy/masquerading daemon for the SIP protocol. It handles registrations of SIP clients on a private IP network and performs rewriting of the SIP message bodies to make SIP connections work via an masquerading firewall (NAT).
Siproxd project allows SIP software clients (like kphone, linphone) or SIP hardware clients (Voice over IP phones which are SIP-compatible, such as those from Cisco, Grandstream or Snom) to work behind an IP masquerading firewall or NAT router.
SIP (Session Initiation Protocol, RFC3261) is the protocol of choice for most VoIP (Voice over IP) phones to initiate communication. By itself, SIP does not work via masquerading firewalls as the transfered data contains IP addresses and port numbers.
There do exist other solutions to traverse NAT existing (like STUN, or SIP aware NAT routers), but such a solutions has its disadvantages or may not be applied to a given situation. Siproxd does not aim to be a replacement for these solutions, however in some situations siproxd may bring advantages.
HOW TO GET STARTED
make sure libosip2 is installed
If your libposip2 libraries are installed in /usr/local/lib, be sure to include this library path to /etc/ld.so.conf
$ ./configure
$ make
$ make install
edit /usr/etc/siproxd.conf according to your situation.
At least configure if_inbound and if_outbound. They must represent the interface names (e.g. on Linux: ppp0, eth1) for the inbound and outbound interface.
edit /usr/etc/siproxd_passwd.cfg if you enable client authentication in siproxd.conf
start siproxd (siproxd does not require root privileges)
$ siproxd
Enhancements:
- Several issues related to 64 bit architectures have been fixed and several minor bugfixes.

sipscreen project is a Linux iptables QUEUE target handler written in perl for screening incoming SIP phone calls.

If you have a network configuration similar to mine, with a Vonage or other Voice-over-IP adapter located behind a Linux gateway, you may find sipscreen useful for programmatically accepting or rejecting inbound calls, based on the caller ID information, the time of day, or any other clever algorithm you can think of.

pcapsipdump project is a tool for dumping (recording) SIP sessions (and RTP traffic, if available) to disk in a fashion similar to "tcpdump -w" (the format is exactly the same).

The difference is that the data is saved with one file per SIP session. Even if there are thousands of concurrect SIP sessions, each goes to separate file.