Linphone is an audio and video internet phone using the SIP protocol.
The main features of linphone are:
- a nice graphical interface;
- includes a large variety of codecs with different quality / bandwidths;
- uses the well-known and standardised SIP protocol.
This package includes the Qt/QML based graphical desktop client.
- Import includes the libs shipped in linphone SDK 4.5.15.
- Mediastreamer2 sndio backend by yours truly, with help from ratchov@.
- Audio calls works but sound needs to be improved,
- Video calls between two linphone instances also working fine.
- AI_V4MAPPED support disabled (unsupported on OpenBSD), IPv6 untested
- bcunit not linked because of a circular dependency with bctoolbox, but
can be useful for debugging.
not yet linked to the build, but importing so that it can be polished
in-tree.
ok/tweaks sthen@
In libssl SSL_CTX and other structs will be made opaque. Take the
code path using accessors instead of reaching directly into some
structs to fix the resulting build breakage.
ok feinerer (maintainer)
SSL_CIPHER_get_id() and SSL_set_options() have always been available
in LibreSSL. The redefinition of SSL_CIPHER_get_id() will cause build
breakage once SSL_CIPHER will be made opaque in libssl.
ok sthen
add a patch from tb@ to fix upcoming build breakage with libressl;
we do have SSL_CIPHER_get_id() and SSL_set_session() so no need to #define
them; currently tis works but SSL_CIPHER is going to become opaque which
will break the redefinitions.
* AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite.
When Asterisk sends a reinvite negotiating T38 faxing, it's possible a
crash can occur if the response contains a m=image and zero port. The
reinvite callback code now checks session_media to see if it is null or
not before trying to access the udptl variable on it.
ASTERISK-29305
if a port needs 2.x then set MODPY_VERSION=${MODPY_DEFAULT_VERSION_2}.
This commit doesn't change any versions currently used; it may be that
some ports have MODPY_DEFAULT_VERSION_2 but don't require it, those
should be cleaned up in the course of updating ports where possible.
Python module ports providing py3-* packages should still use
FLAVOR=python3 so that we don't have a mixture of dependencies some
using ${MODPY_FLAVOR} and others not.
AST-2021-001: Remote crash in res_pjsip_diversion
AST-2021-002: Remote crash possible when negotiating T.38
AST-2021-003: Remote attacker could prematurely tear down SRTP calls
AST-2021-004: An unsuspecting user could crash Asterisk with multiple hold/unhold requests
AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver
there is an active fork of this code on github.com/davies147/astmanproxy,
but it uses various linuxisms (pthread_timedjoin_np, prctl) so sticking
with the old one.
AST-2020-003: Remote crash in res_pjsip_diversion -
A crash can occur in Asterisk when a SIP message is received that has a
History-Info header, which contains a tel-uri.
AST-2020-004: Remote crash in res_pjsip_diversion -
A crash can occur in Asterisk when a SIP 181 response is received that
has a Diversion header, which contains a tel-uri.
IMAP voicemail has moved from building all of Asterisk with a separate
build option (with imap files linked to the main binaries) to a separate
module which can be switched in config. (Only one voicemail module is
allowed at a time, if you have multiple of these installed you can
select between them with noload in modules.conf).
Quirks doesn't handle a flavour moving to unflavoured+subpackage; use
@ask-upgrade so that users of the imap flavour (and only them) are
warned about this at update time.
