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
handsets.
===> The following configuration options are available for pjsip-2.14.1_1:
APPS=off: Install apps & samples from the pjsip-apps directory
DEBUG=off: Build with debugging support
FFMPEG=off: FFmpeg support (WMA, AIFF, AC3, APE...)
KQUEUE=on: Use kqueue ioqueue on macos/BSD (experimental)
PA=on: Use external PortAudio development files
PJSUA2=on: Enable pjsua2 library
RESAMPLEDLL=off: Build libresample as a shared library
SHARED=on: Build shared libraries (other ports may depend on this)
SOUND=on: Include sound in the build
SRTP=on: Secure Real-Time Transport Protocol – SRTP support
UPNP=off: UPnP protocol support
VIDEO=on: Video support
WEBRTC=on: Include libwebrtc in the build
YUV=on: Include libyuv in the build
====> Codecs & Echo Cancellation
AMR=off: AMR 3GPP speech codec support (opencore)
BCG729=off: bcg729 codec support
G711=on: G.711 codec support
G722=on: G.722 codec support
G7221=off: G.722.1 codec support
GSM=off: GSM codec support
ILBC=off: iLBC codec support
L16=off: Linear/L16 codec support
OPUS=on: Opus audio codec support
SILK=off: Enable SILK support
SPEEX=on: Speex audio format support
SPEEXAEC=off: Speex Acoustic Echo Canceller/AEC
====> Resample Implementations: you have to select exactly one of them
RESAMPLE=on: Enable resampling implementations
SAMPLERATE=off: Sample rate conversion support
SPEEXRESAMPLE=off: Enable Speex resample
NONE=off: Disable resampling implementations
===> Use 'make config' to modify these settings
Templates/config.(guess|sub): Update to 2026-05-17
Also remove our local changes that replaced x86_64 with amd64 because
these complicate porting by forcing port maintainers to replace x86_64
with amd64 in configure scripts.
Additional changes:
bsd.port.mk: Apply S/amd64/x86_64/ to CONFIGURE_TARGET. This is not
required to run configure but some ports need to use this in pkg-plist.
comms/qico, news/fidogate:
CONFIGURE_TARGET shouldn't contain "--build". Just remove it altogether
because the default value is fine.
(Only the first 15 lines of the commit message are shown above )
net/pjsip: Update to 2.14.1 and adopt port
Major rework of port for better compliance with Porters Handbook and
general guidelines. Pass maintainership to submitter
Changelog:
https://github.com/pjsip/pjproject/releases/tag/2.14.1
PR: 277445
Remove WWW entries moved into port Makefiles
Commit b7f05445c00f has added WWW entries to port Makefiles based on
WWW: lines in pkg-descr files.
This commit removes the WWW: lines of moved-over URLs from these
pkg-descr files.
Approved by: portmgr (tcberner)
Add WWW entries to port Makefiles
It has been common practice to have one or more URLs at the end of the
ports' pkg-descr files, one per line and prefixed with "WWW:". These
URLs should point at a project website or other relevant resources.
Access to these URLs required processing of the pkg-descr files, and
they have often become stale over time. If more than one such URL was
present in a pkg-descr file, only the first one was tarnsfered into
the port INDEX, but for many ports only the last line did contain the
port specific URL to further information.
There have been several proposals to make a project URL available as
a macro in the ports' Makefiles, over time.
(Only the first 15 lines of the commit message are shown above )
Clean up some things
- Remove duplicate variables
- Remove nop variables
- Sort categories
- Remove redundant option descriptions that match the default ones
Reported by: portscan
- Update pjsip t 2.10
- Get distfile from github
- Add OPUS and VPX options, enabled by default
- Make VIDEO and WEBRTC options enabled by default
- Fix typo in WEBRTC option description
- Fix pkgconfig patch to respect LOCALBASE
PR: 245607
Submitted by: yuri
net/pjsip: Unbreak IPV6 option
- Do not set IPV6_CFLAGS twice
- Turn off IPv6 support when the option is off
Differential Revision: https://reviews.freebsd.org/D22693
- Add to pjsip a customized config_site.h file with values suggested
by the asterisk project. This allows WebRTC to work correctly in
asterisk out of the box [1]
- Also import some patches to pjsip from the asterisk project. These
patches have already been integrated in upstream pjsip development
sources [2]
Reported by: zhenya1993lzn@yandex.ru [1]
Obtained
from: https://github.com/asterisk/asterisk/tree/16.3/third-party/pjproject/patches
[2]
MFH: 2019Q2
Update pjsip patch to work with libressl after the latest update
to 2.7, which broke it.
PR: 216898
Submitted by: OlivierW <olivierw1+bugzilla-freebsd@hotmail.com>
Register conflict with libsrtp.
Reported by: rodrigo
> PR: If and which Problem Report is related.
> Submitted by: If someone else sent in the change.
> Reported by: If someone else reported the issue.
> Reviewed by: If someone else reviewed your modification.
> Approved by: If you needed approval for this commit.
> Obtained from: If the change is from a third party.
> MFC after: N [day[s]|week[s]|month[s]]. Request a reminder
email.
> MFH: Ports tree branch name. Request approval for merge.
> Relnotes: Set to 'yes' for mention in release notes.
> Security: Vulnerability reference (one per line) or
description.
> Sponsored by: If the change was sponsored by an organization.
> Differential Revision: https://reviews.freebsd.org/D### (*full* phabric URL
needed).
> Empty fields above will be automatically removed.
M pjsip/Makefile
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>
- 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" it
- 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>
- 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.com
- 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: 2016Q2
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: maintainer
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 others
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.
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.
- 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 dependencies
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
handsets.
WWW: http://www.pjsip.org/
Please note that default options are tailored for use by the upcoming
asterisk13 port.
