19:05 Daniel Engberg (diizzy)  Author: Oliver Epper

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

    c83600f

07:58 Yuri Victorovich (yuri) 

net/pjsip: Update 2.12.1 -> 2.13

    760e074

14:05 madpilot 

Update pjsip to 2.11

 

10:40 madpilot 

- 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

 

17:18 madpilot 

- Update pjsip to 2.9
- Remove patches already applied upstream
- Bump PORTREVISION on asterisk ports to avoid crashes

 

20:48 madpilot 

- Update asterisk13 to 13.24.0
- Update asterisk15 to 15.7.0
- Update asterisk16 to 16.1.0
- Update pjsip to 2.8
- Remove upstreamed patches from pjsip port
- Add USES=gnome to asterisk ports
- Silence portlint warnings

 

18:27 madpilot 

- Update pjsip to 2.7.1
- Import patches from the asterisk project
- Regenrate patches

Security:	19b052c9-c533-11e7-8da5-001999f8d30b
MFH:		2017Q4

 

12:25 madpilot 

- Update net/pjsip to 2.7
- Bump PORTVERSION on asterisk13 port to avoid ABI problems
- While here, sort the plist

 

17:52 madpilot 

- 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

 

15:26 madpilot 

- 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>

 

14:12 madpilot 

Update to 2.5.5.

PR:		211533
Submitted by:	Joseph Mingrone <jrm at ftfl.ca>

 

14:25 madpilot 

- 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

 

15:46 madpilot 

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.

 

11:54 madpilot 

- 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

 

14:42 madpilot 

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.