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OpenSER 1.2.2
OpenSER aims to be a collaborative project of its users, to develop a scalable and robust SIP server. more>>
OpenSER is a collaborative project of its users, to develop a scalable and robust SIP server.
Anyone can contribute to one of next items:
- code development - OpenSER core, modules and adjacent applications
- documentation- writing or enriching documentation
- miscellaneous - different management tasks (e.g., web site maintenance)
- ideas - new ideas bring brilliant solutions
Main features:
- robust and performant SIP (RFC3261) Registrar server, Location server, Proxy server and Redirect server
- stateless and transactional statefull SIP Proxy processing
- support for UDP/TCP/TLS transport layers
- scripting language for configurations file. With a syntax similar to sripting languages, the configuration offers a powerful and flexible way to deploy custom SIP services.
- management interface via FIFO file and unix sockets
- pseudo-variables to access and manage parts of the SIP messages and attributes specific to users and server
- authentication, authorization and accounting (AAA) via database (MySQL, Postgress, text files), RADIUS and DIAMETER
- CPL - Call Processing Language (RFC3880)
- NAT traversal support for SIP and RTP traffic
- ENUM support
- load balancing and least cost routing extensions
- support for replication - REGISTER offer new functions for replicating client information (real source and received socket).
- logging capabilities - can log custom messages including any header or pseudo-variable and parts of SIP message structure.
- modular architecture - plug-and-play module interface to extend the servers functionality
Enhancements:
- This is the second patch release in 1.2.x series, including minor enhancements and bugfixes done since 1.2.1 was released.
<<lessAnyone can contribute to one of next items:
- code development - OpenSER core, modules and adjacent applications
- documentation- writing or enriching documentation
- miscellaneous - different management tasks (e.g., web site maintenance)
- ideas - new ideas bring brilliant solutions
Main features:
- robust and performant SIP (RFC3261) Registrar server, Location server, Proxy server and Redirect server
- stateless and transactional statefull SIP Proxy processing
- support for UDP/TCP/TLS transport layers
- scripting language for configurations file. With a syntax similar to sripting languages, the configuration offers a powerful and flexible way to deploy custom SIP services.
- management interface via FIFO file and unix sockets
- pseudo-variables to access and manage parts of the SIP messages and attributes specific to users and server
- authentication, authorization and accounting (AAA) via database (MySQL, Postgress, text files), RADIUS and DIAMETER
- CPL - Call Processing Language (RFC3880)
- NAT traversal support for SIP and RTP traffic
- ENUM support
- load balancing and least cost routing extensions
- support for replication - REGISTER offer new functions for replicating client information (real source and received socket).
- logging capabilities - can log custom messages including any header or pseudo-variable and parts of SIP message structure.
- modular architecture - plug-and-play module interface to extend the servers functionality
Enhancements:
- This is the second patch release in 1.2.x series, including minor enhancements and bugfixes done since 1.2.1 was released.
Download (1.5MB)
Added: 2007-08-17 License: GPL (GNU General Public License) Price:
498 downloads
Siproxd 0.5.13
Siproxd is a SIP proxy for SIP-based softphones hidden behind an IP masquerading firewall. more>>
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.
<<lessSiproxd 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.
Download (0.21MB)
Added: 2006-06-20 License: GPL (GNU General Public License) Price:
702 downloads
Sofia-SIP 1.12.6
Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. more>>
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.
<<lessSofia-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.
Download (2.5MB)
Added: 2007-04-26 License: LGPL (GNU Lesser General Public License) Price:
920 downloads
SIP Express Router 0.9.6
SIP Express Router is a very fast and flexible SIP (RFC3261) server. more>>
SIP Express Router (ser) is a high-performance, configurable, free SIP ( RFC3261 ) server .
SER features an application-server interface, presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, FCP security, etc. Web-based user provisioning, serweb, available.
Its performance allows it to deal with operational burdens, such as broken network components, attacks, power-up reboots and rapidly growing user population.
SERs configuration ability meets needs of a whole range of scenarios including small-office use, enterprise PBX replacements and carrier services.
Main features:
- accounting
- digest authentication
- CPL scripts
- ENUM support
- instant messaging
- MySQL support
- PostgreSQL support
- a presence agent
- Radius authentication and accounting
- Diameter authentication
- record routing
- SMS gateway
- Jabber gateway
- NAT traversal support transaction module
- registrar
- user location
SER has been extensively and successfuly tested with many SIP products from other vendors (Microsoft, Cisco, Mitel, snom, Pingtel, Siemens, and many others). It has been powering our SIP services continuously for more than two years.
<<lessSER features an application-server interface, presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, FCP security, etc. Web-based user provisioning, serweb, available.
Its performance allows it to deal with operational burdens, such as broken network components, attacks, power-up reboots and rapidly growing user population.
SERs configuration ability meets needs of a whole range of scenarios including small-office use, enterprise PBX replacements and carrier services.
Main features:
- accounting
- digest authentication
- CPL scripts
- ENUM support
- instant messaging
- MySQL support
- PostgreSQL support
- a presence agent
- Radius authentication and accounting
- Diameter authentication
- record routing
- SMS gateway
- Jabber gateway
- NAT traversal support transaction module
- registrar
- user location
SER has been extensively and successfuly tested with many SIP products from other vendors (Microsoft, Cisco, Mitel, snom, Pingtel, Siemens, and many others). It has been powering our SIP services continuously for more than two years.
Download (2.0MB)
Added: 2006-01-11 License: GPL (GNU General Public License) Price:
1393 downloads
Other version of SIP Express Router
License:Freeware
Yxa 1.0 RC1
Yxa is a SIP stack and a set of SIP server applications written in Erlang/OTP. more>>
Yxa is a SIP stack and a set of SIP server applications written in Erlang/OTP. The SIP stack is RFC3261 compliant.
Among the features implemented are SIP registrar, SIP router, forking, CPL, IPv6, TLS, ENUM, PSTN gateway access control and modular user database backends.
The main goal of the project is to create a robust SIP server platform that can scale to tens of thousands of users, be interoperable through standards compliance, and still have short time-to-market for new features due to the use of a high level language.
Main features:
- RFC3261 compliant SIP-server, capable of everything a generic domain needs :
- Registrar that keeps track of your users
- Handles incoming SIP requests to your domain
- Handles routing of requests from your users to remote domains
- TCP, UDP and TLS (including SIPS) support
- Automatically maps e-mail addresses of your users to their SIP addresses, if you have the e-mail addresses in LDAP
- Handles multiple domains using a single server instance
- ENUM support for PSTN-bypass whenever possible
- IPv6 support
- Forking, both parallel and sequential
- CPL (RFC3880) support for advanced user-control of events (currently incoming calls only)
- Modular user database, currently with LDAP, Mnesia, MySQL and text-file backends
- PSTN destination access control (per user or for anonymous users
Enhancements:
- Various bugs in the draft-Outbound implementations have been fixed.
- The pstnproxys application logic has been updated to suit some real deployment scenarios.
- The test framework has been greatly enhanced.
<<lessAmong the features implemented are SIP registrar, SIP router, forking, CPL, IPv6, TLS, ENUM, PSTN gateway access control and modular user database backends.
The main goal of the project is to create a robust SIP server platform that can scale to tens of thousands of users, be interoperable through standards compliance, and still have short time-to-market for new features due to the use of a high level language.
Main features:
- RFC3261 compliant SIP-server, capable of everything a generic domain needs :
- Registrar that keeps track of your users
- Handles incoming SIP requests to your domain
- Handles routing of requests from your users to remote domains
- TCP, UDP and TLS (including SIPS) support
- Automatically maps e-mail addresses of your users to their SIP addresses, if you have the e-mail addresses in LDAP
- Handles multiple domains using a single server instance
- ENUM support for PSTN-bypass whenever possible
- IPv6 support
- Forking, both parallel and sequential
- CPL (RFC3880) support for advanced user-control of events (currently incoming calls only)
- Modular user database, currently with LDAP, Mnesia, MySQL and text-file backends
- PSTN destination access control (per user or for anonymous users
Enhancements:
- Various bugs in the draft-Outbound implementations have been fixed.
- The pstnproxys application logic has been updated to suit some real deployment scenarios.
- The test framework has been greatly enhanced.
Download (0.62MB)
Added: 2007-04-12 License: BSD License Price:
925 downloads
sipsak 0.9.6
sipsak is a command line tool for performing various tests on Session Initiation Protocol (SIP) applications and devices. more>>
sipsak is a small comand line tool for developers and administrators of Session Initiation Protocol (SIP) applications. sipsak can be used for some simple tests on SIP applications and devices.
Main features:
- sending OPTIONS request
- sending text files (which should contain SIP requests)
- traceroute (see section 11 in RFC3261)
- user location test
- flooding test
- random character trashed test
- interpret and react on response
- authentication with qop supported
- short notation supported for receiving (not for sending)
- string replacement in files
- can simulate calls in usrloc mode
- uses symmetric signaling and thus should work behind NAT
- can upload any given contact to a registrar
- send messages to any SIP destination
- Nagios compliant return codes
- search for strings in reply with regluar expression
- use multiple processes to create more server load
- read SIP message from STDIN (e.g. from a pipe |)
- supports DNS SRV through libruli
Version restrictions:
- The hostname is used in the Via line, which is not correct in all cases (e.g. if the loopback interface is used, or if the host has several interfaces). The rport parameter should fix problmes with the incorrect hostname, but for backward compatibility whith implementations which do not support rport this should be fixed.
- The DNS responses are not parsed compeltly which can result in strange output during hostname detection.
- TCP is not supported as transport protocol.
- IPv6 is not supported as transport protocol.
- Missing support for the Record-Route and Route header.
- Not fully RFC3261 compatible.
- Some smaller problems are listed in the TODO file.
Enhancements:
- A new option allows to add any header to the outgoing requests.
- The variable replacement option now accepts any number of attribute value pairs.
- Besides MD5 now SHA1 is support as digest authentication algorithm.
- The password for authentication can be read from stdin to prevent password disclosure in the process list.
- Fixed problems when executed as user root and compiles fine again under cygwin.
<<lessMain features:
- sending OPTIONS request
- sending text files (which should contain SIP requests)
- traceroute (see section 11 in RFC3261)
- user location test
- flooding test
- random character trashed test
- interpret and react on response
- authentication with qop supported
- short notation supported for receiving (not for sending)
- string replacement in files
- can simulate calls in usrloc mode
- uses symmetric signaling and thus should work behind NAT
- can upload any given contact to a registrar
- send messages to any SIP destination
- Nagios compliant return codes
- search for strings in reply with regluar expression
- use multiple processes to create more server load
- read SIP message from STDIN (e.g. from a pipe |)
- supports DNS SRV through libruli
Version restrictions:
- The hostname is used in the Via line, which is not correct in all cases (e.g. if the loopback interface is used, or if the host has several interfaces). The rport parameter should fix problmes with the incorrect hostname, but for backward compatibility whith implementations which do not support rport this should be fixed.
- The DNS responses are not parsed compeltly which can result in strange output during hostname detection.
- TCP is not supported as transport protocol.
- IPv6 is not supported as transport protocol.
- Missing support for the Record-Route and Route header.
- Not fully RFC3261 compatible.
- Some smaller problems are listed in the TODO file.
Enhancements:
- A new option allows to add any header to the outgoing requests.
- The variable replacement option now accepts any number of attribute value pairs.
- Besides MD5 now SHA1 is support as digest authentication algorithm.
- The password for authentication can be read from stdin to prevent password disclosure in the process list.
- Fixed problems when executed as user root and compiles fine again under cygwin.
Download (0.14MB)
Added: 2006-01-29 License: GPL (GNU General Public License) Price:
1367 downloads
SSIP-GST 1.0.0
SSIP-GST is yet another SIP/SIMPLE Gaim plugin more>>
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.
<<lesscompliant 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.
Download (0.29MB)
Added: 2006-01-18 License: GPL (GNU General Public License) Price:
1376 downloads
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