PJSIP

PJSIP
Name	PJSIP
Developer	Teluu (now part of Sippy), community
Released	2006
Operating system	Cross-platform
Genre	VoIP, SIP, multimedia
License	GNU Lesser General Public License

PJSIP is an open-source multimedia communication library implementing SIP, RTP, SDP and related protocols for Voice over IP. It is designed for real-time audio, video, and presence applications and is widely used in telephony gateways, softphones, and embedded devices. The project emphasizes portability, small footprint, and integration with telephony projects and standards bodies.

Overview

PJSIP provides a protocol stack and higher-level APIs for session initiation and media handling, enabling interoperability with projects and organizations such as Asterisk (PBX), FreeSWITCH, OpenSIPS, Kamailio, and commercial vendors like Cisco Systems, Avaya, Ericsson, and Huawei. It implements standards from the Internet Engineering Task Force through RFCs related to Session Initiation Protocol, Real-time Transport Protocol, and Session Description Protocol, and is adopted by applications ranging from mobile clients on platforms like Android and iOS to server-side gateways used by carriers like Verizon and Vodafone.

Architecture and Components

The library is structured into modular components: a SIP stack, a media stack using RTP/RTCP, codec management, NAT traversal modules, and a high-level user agent API. These components interact with networking subsystems and OS services including Linux kernel, Windows NT, macOS, and embedded RTOSes found in devices from Qualcomm and Broadcom. The SIP stack interoperates with proxy and registrar servers such as SIP Express Router and session border controllers from vendors like Ribbon Communications.

Features and Functionality

PJSIP supports SIP features including registration, invitation, forking, and presence, along with advanced call control features used by platforms like Microsoft Lync and Zoom Video Communications. Media features include adaptive jitter buffering, echo cancellation, and codec support for standards and implementations like G.711, G.722, Opus (audio codec), and H.264 interworking used by Polycom and Cisco TelePresence. For NAT traversal it integrates techniques similar to those standardized by STUN, TURN, and ICE used by projects like WebRTC.

Usage and APIs

Developers interact with PJSIP through C APIs and language bindings for Python (programming language), Java (programming language), and C++ wrappers, enabling integration in client applications such as softphones, PBX plugins, and call center software from vendors like Avaya Aura and Genesys. The library exposes APIs for account management, call control, media transport, and event callbacks used in systems developed by organizations like Twilio and Skype (application) derivatives. Example integrations include backend telephony services connecting to services like Amazon Web Services and Microsoft Azure.

Implementation and Platforms

PJSIP is implemented in portable C with optional assembly optimizations for DSPs from ARM Holdings and instruction sets used by Intel Corporation. It is compiled with toolchains such as GCC, Clang, and MSVC and packaged for distributions including Debian (operating system), Ubuntu, and Red Hat Enterprise Linux. Mobile deployments target platforms distributed by Samsung Electronics and Apple Inc., while embedded uses appear in network appliances by Cisco Systems and IoT devices from vendors like Texas Instruments.

Security and Performance

Security features include support for Transport Layer Security and SIP over TLS, SRTP encryption from standards bodies like IETF and cipher suites implemented in libraries such as OpenSSL and mbed TLS. PJSIP integrates authentication methods compatible with identity systems deployed by enterprises like Google LLC and Facebook for secure VoIP. Performance engineering addresses latency and jitter for carrier-grade deployments used by operators like AT&T and Deutsche Telekom, with benchmarking against media servers such as FreeSWITCH and Asterisk (PBX), and profiling using tools originating from projects like Valgrind and perf (Linux).

History and Development

The project began in the mid-2000s and grew through contributions from independent developers and companies, collaborating with ecosystem participants including SIP Forum, standards contributors at the IETF, and open-source communities around Asterisk (PBX) and OpenSIPS. Over time it received adoption across commercial vendors such as Cisco Systems, Huawei, and service providers like Verizon and community projects including Linphone and Ekiga. Development continues in public repositories and mailing lists with influence from conferences like ICCCN and SIGCOMM that shape real-time communications research.

Category:Telecommunications software