Partysip

Partysip is a SIP proxy server. It is a plugin oriented program with registration, authentication and routing capabilities.

Partysip is a modular application where capabilities are added and removed through plugins. The program comes with several GPL plugins. At this step, partysip and its plugins could be used as a 'SIP registrar', a 'SIP redirect server' and a 'SIP stateful proxy server'.
siproxd - SIP proxy/masquerading daemon

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 possible via an masquerading firewall. It allows SIP clients (like kphone, linphone) to work behind an IP masquerading firewall or router.

SIP (Session Initiation Protocol) is used by Softphones (Voice over IP) to initiate communication. By itself, SIP does not work via masquerading firewalls as the transfered data contains IP addresses and port numbers.
Load Balancer Proxy

The Load Balancer is a very simple proxy that is useful in SIP-based VoIP installations where there are multiple ingress proxy servers. The Load Balancer permits pooling these servers, thereby eliminating the need to balance user demands for connectivity through a complicated provisioning algorithm.

All users can send their INVITEs and REGISTERs to the same SIP URI and the Load Balancer will assign ingress proxy servers dynamically to each transaction. In this way, the traffic load is balanced over a pool of proxy servers based on the real-time demand for services.
STUN Server

The STUN (Simple Traversal of UDP through NATs (Network Address Translation)) server is an implementation of the STUN protocol that enables STUN functionality in SIP-based systems. The STUN server tar ball also include a client API to enable STUN functionality in SIP endpoints. In addition there is a command line UNIX client and a graphical windows client that check what type of NAT the user is using.

STUN is an application-layer protocol that can determine the public IP and nature of a NAT device that sits between the STUN client and STUN server.

The current version of the code supports most of RFC 3489 except the ability to get OTPs from the server.
Free VoIP Software Development Libraries
Yate

Yate (Yet Another Telephony Engine) is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.

Yate can be used to build a:

* VoIP server
* VoIP client
* VoIP to PSTN gateway
* PC2Phone and Phone2PC gateway
* H.323 gatekeeper
* H.323 multiple endpoint server
* H.323<->SIP Proxy
* SIP session border controller
* SIP router
* SIP registration server
* IAX server and/or client
* IP Telephony server and/or client
* Call center server
* IVR engine
* Prepaid and/or postpaid cards system

The software is written in C++ and it supports scripting in various programming languages (such as those supported by the currently implemented embedded PHP, Python and Perl interpreters) and even any Unix shell. The PHP, Python and Perl libraries have been developed and made available in order to ease development of external functionalities for Yate.

Yate is production-ready software and is easily extensible.

Yate is licensed under the GPL with an exception for linking with OpenH323 and PWlib (licensed under MPL).
PJSIP

PJSIP is an open source SIP stack supporting many SIP extensions/features, with the following key benefits:
Extremely portable

Write the application once, and it would run on many many platforms (all Windows flavors, Windows Mobile, Linux, all Unix flavors, MacOS X, RTEMS, Symbian OS, etc.)

Very small footprint

With less than 150KB for complete SIP features, PJSIP is ideal not only for embedded development where space is costly but also for general applications where smaller size means shorter download time for users.

High performance

...which means less CPU power requirement and more SIP transactions/calls can be handled per second.

Many features

Many SIP features/extensions such as multiple usages in dialog, event subscription framework, presence, instant messaging, call transfer, etc. have been implemented in the library.

Extensive SIP documentation

There can never be enough documentation, so we try to provide fellow developers with hundreds of pages worth of documentation.

PJSIP also features extensions, such as:
PJMEDIA

PJMEDIA is a complementary library for PJSIP to build a complete, full-featured SIP user agent applications such as softphones/hardphones, gateways, or B2BUA.

PJLIB-UTIL

PJLIB-UTIL is an auxiliary library providing supports for PJMEDIA and PJSIP. Some of the functions/components in this library: small footprint XML parsing, STUN client library, asynchronous/caching DNS resolver, hashing/encryption functions, etc.

PJLIB

A small footprint, high performance, ultra portable abstraction library and framework, used by PJSIP and PJMEDIA.

PJLIB is about the only library that PJLIB-UTIL, PJMEDIA, and PJSIP should depend, as it provides complete abstraction not only to Operating System dependent features, but it is also designed to abstract LIBC and provides some useful data structures too.

Vovida Open Communication Application Library (VOCAL)

The Vovida Open Communication Application Library (VOCAL) is an open source project targeted at facilitating the adoption of VoIP in the marketplace. VOCAL provides the development community with software and tools needed to build new and exciting VoIP features, applications and services. The software in VOCAL includes a SIP based Redirect Server, Feature Server, Provisioning Server, Policy Server and Marshal Proxy along with protocol translators from SIP to H.323 and SIP to MGCP. Our hope is that these modules will act as building blocks to help you create better, faster and stronger VoIP systems.
The GNU oSIP Library

oSIP is an implementation of SIP.

SIP stands for the Session Initiation Protocol and is described by the RFC3261. This library aims to provide multimedia and telecom software developers an easy and powerful interface to initiate and control SIP based sessions in their applications. SIP is a open standard replacement from IETF for H.323.
JVOIPLIB (Jori's Voice over IP library)

JVOIPLIB is an object-oriented Voice over IP (VoIP) library written in C++.