6c8fc9e955
iax: downgrade false positive errors to debug messages
...
When dialing with an IAX account, we get a lot of "No SIP CALL" error
messages being logged. This is clearly wrong and until we avoid
looking for SIPCalls from an IAX account, we should present this
message as debug info.
2013-07-10 16:34:12 -04:00
9dc8964150
* #20661 : opus: ensure that HAVE_OPUS is defined
2013-07-08 14:58:34 -04:00
bc787648c4
Merge branch 'master' into multichannel
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Resolved Conflicts:
daemon/src/audio/audioloop.cpp
daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
daemon/src/audio/codecs/audiocodec.h
daemon/src/audio/pulseaudio/pulselayer.cpp
2013-07-08 14:06:50 -04:00
9894e13979
* #25537 : daemon/dbus renamed daemon/client
2013-07-05 16:04:37 -04:00
ac11026595
* #25537 : daemon: move dbusmanager.* to client.*
2013-07-05 16:04:37 -04:00
7360999f3a
* #25537 : daemon: rename DBusManager => Client
2013-07-05 16:04:36 -04:00
f3ef185c83
sipaccount/tls: cleanup
2013-07-03 14:02:13 -04:00
fe027880d3
* #26628 : TLS: limit number of ciphers to avoid abort in pjsip
2013-07-03 13:34:31 -04:00
9dc5ca34fd
* #7078 : audio: suspend audio processing if peer hungup and no calls remain
...
checkAudio() was being called before the call was removed from the
call list.
2013-06-27 16:23:55 -04:00
f6025f9d66
* #26453 : sip: use HAVE_TLS around all ssl dependent code
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This fixes building after running ./configure --without-tls
2013-06-26 13:29:56 -04:00
944d4eb65c
Merge branch 'master' into merge-upstream
...
Conflicts:
daemon/src/sip/sipvoiplink.cpp
2013-06-25 14:48:02 -04:00
c15fef2fef
* #26295 : sip: don't crash if transport could not be created
2013-06-21 17:16:57 -04:00
1156d05f13
* #26183 : sdp: add telephone-event payload to media description
2013-06-20 11:33:26 -04:00
4c425b63fb
Merge branch 'master' into merge-upstream
...
Conflicts:
daemon/src/audio/codecs/audiocodec.h
daemon/src/audio/codecs/audiocodecfactory.cpp
daemon/src/dbus/callmanager.cpp
daemon/src/dbus/callmanager.h
daemon/src/managerimpl.cpp
daemon/src/sip/sipvoiplink.cpp
2013-06-19 16:08:38 -04:00
aa76bfb7b0
Merge branch 'master' into merge-upstream
...
Conflicts:
daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
daemon/src/audio/audiortp/audio_zrtp_session.cpp
daemon/src/audio/codecs/audiocodec.h
daemon/src/audio/codecs/audiocodecfactory.cpp
daemon/src/fileutils.cpp
daemon/src/history/history.cpp
daemon/src/managerimpl.cpp
daemon/src/managerimpl.h
2013-06-19 15:41:35 -04:00
49bb07f15e
* #25946 : sip: fix calling IP2IP calls from history
2013-06-13 13:46:54 -04:00
ac555482bd
* #25916 : sip: restrict RTP port ranges
2013-06-12 15:16:03 -04:00
20c74f99df
* #25295 : sip: update published address for STUN
2013-06-12 15:16:03 -04:00
cd4d4488e4
* #25295 : audiortp: get ports from STUN
2013-06-12 15:16:02 -04:00
9bba948933
sipvoiplink: cleanup
2013-06-12 15:16:02 -04:00
7b7f1fbbcf
* #25295 : sipaccount: remove dead code
2013-06-12 15:16:02 -04:00
94ed1f03e1
* #25900 : sip: fixed broken stripPrefix code and added regression test
2013-06-12 11:21:44 -04:00
58a780d5e9
* 25787: sip: use STUN address in VIA sent-by
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This fixes registration when using STUN.
2013-06-10 14:41:09 -04:00
cbf0abc7c5
* #25472 : sip: fix port number calculation
2013-06-05 13:46:50 -04:00
ab6ad8932e
* #25393 : siptransport: fix Log crash
2013-06-03 18:08:44 -04:00
248ad66a92
* #23661 : daemon: restore getCallList
2013-06-03 16:21:50 -04:00
a8cfbf96ac
account: move split_string into Account
2013-06-03 15:21:17 -04:00
d3b7d16956
* #23661 : daemon: removed unneeded hasCalls
2013-06-03 14:45:46 -04:00
638727aced
* #23661 : daemon: removed callAccountMap
2013-06-03 14:36:58 -04:00
6857b265b1
* Extracted JNI callbacks in new file
...
* Added accounts state changed jni callbacks
2013-06-03 09:27:10 -04:00
fbd1b2eed7
* #25076 : sip: fix forward declarations of structs
2013-05-28 12:00:43 -04:00
d13a09c248
* #14077 : video: send and receive RTP on one socket
...
Thanks to the new custom_io flag in libavformat's SDP demuxer, we can manage
our own UDP transports for RTP and RTCP. This allows us to comply with
RFC 4961.
If an older version of libavformat is detected, we fallback to sending and
receiving on different sockets.
2013-05-23 11:10:12 -04:00
8b3f0b976b
* #24106 : sip: fix instant messaging regression
2013-05-09 16:23:23 -04:00
bb0ab251f9
* #24017 Modified src/Android.mk to include im module
2013-05-09 11:03:07 -04:00
0ee2b621ee
Merge branch 'android' of git+ssh://git.sflphone.org/var/repos/sflphone/git/sflphone into android
...
Conflicts:
daemon/src/history/history.cpp
2013-04-25 14:44:08 -04:00
ac5b63f883
* #23362 Modified swig interface for getHistory implementation
...
* #23361 Added log in siptransport.cpp
2013-04-25 14:37:39 -04:00
ff9274dec4
build fixes
2013-04-19 00:01:05 +10:00
5b639fdfad
Opus changes
2013-04-05 19:11:14 +11:00
d5f33df660
Use the first PA device if the prefered one is not available.
2013-04-05 16:01:23 +11:00
1401bc930b
SDP protocol support for Opus; using only one codec class for mono/stereo in Opus
2013-03-28 16:01:45 +11:00
8d13be0ef5
* #21631 : daemon: void NULL pointer dereference on unexpected case
2013-03-19 11:39:13 -04:00
274c3e0280
* #21631 : siptransport: guarantee that ifr_name is NULL-terminated
2013-03-15 12:23:41 -04:00
ed3d20a4fc
* 21631: sip: avoid buffer overrun
2013-03-14 18:15:06 -04:00
7c09d54701
* #21507 : sipcall: handle exception if audiortp is no longer in scope
2013-03-14 17:53:39 -04:00
1fabd14c99
* #21631 : sipvoiplink: fix broken NULL check
2013-03-14 17:43:43 -04:00
5b0453cc22
* 21631: sipvoiplink: NULL check account before using it
2013-03-14 17:37:21 -04:00
6570249de3
* #21631 : sipaccount: don't use account before NULL check
2013-03-14 17:28:38 -04:00
891bc0aad3
* #21631 : daemon: don't allow access to deleted call
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The function must return right after deleting the call, or we could
have an illegal memory access.
2013-03-14 17:05:49 -04:00
429d1f3603
* #21507 : daemon: get audio codec names from audiortpfactory, not SDP
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This fixes the race-condition + SEGFAULT
2013-03-14 14:55:03 -04:00
81b95dd7f2
* #21556 : sip: re-register if registration attempt failed with timeout
2013-03-13 17:21:43 -04:00
