Reported by: rodrigo
> PR: If and which Problem Report is related.
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M pjsip/Makefile
- 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
- 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
- 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
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
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
- Fix asterisk13 SRTP support
- Fix asterisk13 SPEEX_LIB_DEPENDS
- While here make SRTP option default for asterisk13 since it does
not add dependencies
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.