Multimedia communication library written in C language
# pkg install pjsipinstalls 2.14.1 from latest; from quarterly (2026Q4) 2.14.1# cd /usr/ports/net/pjsip && make install clean# make -C /usr/ports/net/pjsip config # change options first
| FreeBSD version | amd64 | aarch64 | i386 |
|---|---|---|---|
| FreeBSD 16 | 2.14.1 | 2.14.1 | – |
| FreeBSD 15 | 2.14.1 | 2.14.1 | – |
| FreeBSD 14 | 2.14.1 | 2.14.1 | ✗ |
| FreeBSD version | amd64 | aarch64 | i386 |
|---|---|---|---|
| FreeBSD 15 | 2.14.1 | 2.14.1 | – |
| FreeBSD 14 | 2.14.1 | 2.14.1 | ✗ |
version as in the ports tree (latest) or the most common one (quarterly) · different version · ✗ not built · – no packages for this platform. Package sets that have not been rebuilt for 60 days are left out.
The CVE record lists the packaged version 2.14.1 as affected, and no VuXML entry names the CVE. What does this mean?
| CVE | CVSS | Published | Description |
|---|---|---|---|
| CVE-2026-57161 | 8.8 | 2026-09-04 | PJSIP is a free and open source multimedia communication library written in C. Prior to commit acc03b5, a stack buffer overflow exists in PJSUA when processing Service-R… |
| CVE-2026-57162 | 8.8 | 2026-09-04 | PJSIP is a free and open source multimedia communication library written in C. Prior to commit a1b707c, a stack buffer overflow exists in the SRTP/SDES media transport w… |
| CVE-2026-57163 | 8.8 | 2026-09-04 | PJSIP is a free and open source multimedia communication library written in C. Prior to commit c4a151a, a stack buffer overflow exists in the GnuTLS TLS backend when par… |
| CVE-2026-57159 | 8.4 | 2026-09-04 | PJSIP is a free and open source multimedia communication library written in C. Prior to commit 673b978, a remote out-of-bounds read and write can occur in the SDP negoti… |
| CVE-2026-57164 | 8.3 | 2026-09-04 | PJSIP is a free and open source multimedia communication library written in C. Prior to commit 8d5956a, a heap buffer overflow exists in the PJLIB-UTIL HTTP client (http… |
| CVE-2026-42225 | 8.2 | 2026-05-07 | PJSIP is a free and open source multimedia communication library written in C. Prior to version 2.17, on GnuTLS builds, the SIP TLS transport (sip_transport_tls) can acc… |
| CVE-2026-41416 | 8.1 | 2026-04-24 | PJSIP is a free and open source multimedia communication library written in C. In 2.16 and earlier, there is an integer overflow in media stream buffer size calculation … |
| CVE-2026-84975 | 7.4 | 2026-09-18 | PJSIP is a free and open source multimedia communication library written in C. In 2.17 and earlier, the OpenSSL and GnuTLS backends in pjlib/src/pj/ssl_sock_ossl.c and p… |
| CVE-2026-77396 | 6.9 | 2026-09-18 | PJSIP is a free and open source multimedia communication library written in C. In 2.17 and earlier, the PJSIP AVI parser in pjmedia/src/pjmedia/avi_player.c uses an inpu… |
| CVE-2026-57160 | 6.9 | 2026-09-04 | PJSIP is a free and open source multimedia communication library written in C. Prior to commit d6a0e7f, a buffer overflow can occur in pjsip_generic_array_hdr_print() in… |
| CVE-2026-34235 | 6.9 | 2026-03-31 | PJSIP is a free and open source multimedia communication library written in C. Prior to version 2.17, a heap out-of-bounds read vulnerability exists in PJSIP's VP9 RTP u… |
| CVE-2026-41415 | 6.7 | 2026-04-24 | PJSIP is a free and open source multimedia communication library written in C. In 2.16 and earlier, there is an out-of-bounds read when parsing a malformed Content-ID UR… |
| CVE-2026-57166 | 6.3 | 2026-09-04 | PJSIP is a free and open source multimedia communication library written in C. Prior to commit 4472a31, a stack buffer overflow exists in the PJLIB-UTIL telnet CLI front… |
| CVE-2026-57165 | 6.3 | 2026-09-04 | PJSIP is a free and open source multimedia communication library written in C. Prior to commit 628b716, a stack buffer overflow exists in the PJLIB-UTIL telnet CLI front… |
Listed in VuXML, the packaged version 2.14.1 is still in the affected range; pkg audit reports it. What does this mean?
no CVE referenced
no CVE referenced
no CVE referenced
no CVE referenced
- Update net/asterisk13 to 13.15.1 - Import patches to fix security issues in net/pjsip [1] - While here regenerate net/pjsip patches Obtained from: Asterisk 13.15.1 distribution file [1] MFH: 2017Q2 Security: 0537afa3-3ce0-11e7-bf9d-001999f8d30b Security: fab87bff-3ce5-11e7-bf9d-001999f8d30b46d742e1b1
Only strip shared libraries if we are actually install them. This fixes the build when the SHARED option is off. Reported by: Dmitry Arkhireev <che at teftel.ru>6b6c984849
Fix net/pjsip build with libressl. PR: 216898 Submitted by: OlivierW <olivierw1+bugzilla-freebsd@hotmail.com>e46c07f2b6
- Update pjsip to 2.6 - Add new WEBRTC option, disabled by default - Make audio/speexdsp a dependency of the SPEEX option, reported by poudriere - Regenerate some patches - Bump net/asterisk13 PORTREVISION, I observed crashed when updating the pjsip libraries "below" it24ef7da130
multimedia/ffmpeg: update to 3.2.2 Changes: https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/n3.2.2:/Changelog PR: 207547 Submitted by: riggs, ebirth@b0ss.net (libressl fix) Exp-run by: antoine (3 tries)0c612c4aa3
- Add PJSUA option (on by default) to install the pjsua client binary - Strip libraries - Reflowed OPTIONS_DEFINE PR: 211533 Submitted by: Joseph Mingrone <jrm at ftfl.ca>3f0427cf06
Update to 2.5.5. PR: 211533 Submitted by: Joseph Mingrone <jrm at ftfl.ca>c97af6449d
multimedia/openh264: update to 1.6.0 API/ABI has changed, so bump PORTREVISION in consumers. Some of those are marked BROKEN that may need a fix similar to libav. https://git.libav.org/?p=libav.git;a=commitdiff;h=d825b1a53065 Changes: https://github.com/cisco/openh264/releases/tag/v1.6.0f1071e550c
- Update to 2.5.1 - Remove duplicate IPV6 opption in OPTIONS_DEFINE951b80d7c8
- Update net/pjsip to 2.5 [1] - Disable unneeded ALSA support in pjsip [1] - Replace custom patch with USES=pathfix - Fix pjsip build system to allow building while previous version is installed in PREFIX/LOCALBASE - Bump dependent port asterisk13 PORTREVISION to avoid runtime crash (seen while testing) PR: 209477 [1] Submitted by: yuri at rawbw.comec0cd005ec
- Update net/asterisk13 to 13.8.1 - Change PJ_IOQUEUE_MAX_HANDLES build time limit in pjsip as suggested by asterisk project [1] to mitigate potential DoS [2] - Add DEBUG and IPV6 options to pjsip port Obtained from: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject [1] Security: ee50726e-0319-11e6-aa86-001999f8d30b e21474c6-031a-11e6-aa86-001999f8d30b [2] MFH: 2016Q271121e8047
Remove ${PORTSDIR}/ from dependencies, categories m, n, o, and p.
With hat: portmgr
Sponsored by: Absolight8d6597e0bb- Update pjsip to 2.4.5 - Convert to USES=localbase - Bump asterisk PORTREVISION, it needs to be rebuild after this update.97ee0a0282
Update portaudio to v19/Remove portaudio2 [1] Chase portaudio change Add patches from debian for games/cultivation Add patches from upsteam for audio/rezound Mark py-fastaudio as broken Approved by: maintainer3fef6c52f0
Add a slave port to net/pjsip to force installing pjsip with external SRTP library. Make the www/asterisk13 depend on this slave port when both SRTP and PJSIP options in it are enabled, this allows enabling SRTP support in asterisk13 without the need to manually reconfigure other ports. Reported by: mat@ and a few others0bff8d28e5
Modify asterisk UPDATING entry to reflect the present situation. While here also modify misleading and stale comment in the net/pjsip port EXTSRTP option. Thanks to mat@ for making me notice these.f0bc5e8dd9
Due to recent changes in OpenSSL, disable SRTP support for asterisk13 by default and avoid pjsip pulling in libsrtp, otherwise a not working package would be generated. Add note to UPDATING to keep users informed.c20cc7662e
- Make pjsip use ports provided libsrtp instead of boudled one - Fix asterisk13 SRTP support - Fix asterisk13 SPEEX_LIB_DEPENDS - While here make SRTP option default for asterisk13 since it does not add dependenciesced7c37fcb
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging from desktops, embedded systems, to mobile WWW: http://www.pjsip.org/ Please note that default options are tailored for use by the upcoming asterisk13 port.1305d65ff1