| Port details |
- pjsip-extsrtp Multimedia communication library written in C language
- 2.9_4 net
 =0       2.9_4Version of this port present on the latest quarterly branch. - Maintainer: madpilot@FreeBSD.org
- Port Added: 2015-05-06 20:10:26
- Last Update: 2020-04-18 10:10:16
- SVN Revision: 532016
- License: GPLv2+
- WWW:
- https://www.pjsip.org/
- Description:
- 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: https://www.pjsip.org/
 cgit ¦ GitHub ¦ GitHub ¦ GitLab ¦ 
- Manual pages:
- FreshPorts has no man page information for this port.
- pkg-plist: as obtained via:
make generate-plist - Dependency lines:
-
- pjsip-extsrtp>0:net/pjsip-extsrtp
- Conflicts:
- CONFLICTS:
- No installation instructions:
- This port has been deleted.
- PKGNAME: pjsip-extsrtp
- Flavors: there is no flavor information for this port.
- distinfo:
- TIMESTAMP = 1560947683
SHA256 (pjproject-2.9.tar.bz2) = d185ef7855c8ec07191dde92f54b65a7a4b7a6f7bf8c46f7af35ceeb1da2a636
SIZE (pjproject-2.9.tar.bz2) = 5009546
No package information for this port in our database- Sometimes this happens. Not all ports have packages. Perhaps there is a build error. Check the fallout link:
- Master port: net/pjsip
- Dependencies
- NOTE: FreshPorts displays only information on required and default dependencies. Optional dependencies are not covered.
- Build dependencies:
-
- gmake : devel/gmake
- Library dependencies:
-
- libportaudio.so : audio/portaudio
- libsrtp2.so : net/libsrtp2
- libgsm.so : audio/gsm
- libspeex.so : audio/speex
- libspeexdsp.so : audio/speexdsp
- NOTE: dependencies for deleted ports are notoriously suspect
- This port is required by:
- for Libraries
-
Deleted ports which required this port:
- * - deleted ports are only shown under the This port is required by section. It was harder to do for the Required section. Perhaps later...
Configuration Options:- ===> The following configuration options are available for pjsip-extsrtp-2.9_4:
AMR=off: AMR 3GPP speech codec support (opencore)
DEBUG=off: Build with debugging support
FFMPEG=off: FFmpeg support (WMA, AIFF, AC3, APE...)
G711=on: G.711 codec support
G722=on: G.722 codec support
G7221=on: G.722.1 codec support
GSM=on: GSM codec support
ILBC=on: iLBC codec support
IPV6=on: IPv6 protocol support
L16=on: Linear/L16 codec support
OPENH264=off: OpenH264 support
PJSUA=off: Command line SIP agent
RESAMPLE=off: Enable resampling implementations
RESAMPLEDLL=off: Build libresample as shared library
SAMPLERATE=off: Sample rate conversion support
SDL=off: Simple Direct Media Layer support
SHARED=on: Build shared libraries (other ports may depend on this)
SOUND=off: Sound (audio) support
SPEEX=on: Speex audio format support
SPEEXAEC=on: Speex Acoustic Echo Canceller/AEC
V4L=off: Video4Linux2 support
VIDEO=off: Video support
WEBRTC=off: Build linwebrtc
===> Use 'make config' to modify these settings
- Options name:
- N/A
- USES:
- gmake localbase pathfix ssl tar:bz2
- FreshPorts was unable to extract/find any pkg message
- Master Sites:
|
Number of commits found: 3
| Commit History - (may be incomplete: for full details, see links to repositories near top of page) |
| This is a slave port. You may also want to view the commits to the master port: net/pjsip | | Commit | Credits | Log message |
|---|
2.9_4
18 Apr 2020 10:10:16
  |
madpilot |
Remove net/pjsip-extsrtp slave port. It was created for use by
asterisk ports, which don't use it anymore after r532013. |
2.7.2_2
11 Jun 2018 17:19:29
|
madpilot |
Make asterisk and pjsip ports use the new net/libsrtp2 port as a dependency.
Reported by: tijl (thanks!) |
2.3_3
06 May 2015 20:10:10
 |
madpilot |
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 |
Number of commits found: 3
|
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