63a406737a
* #27724 : sdp: use published IP address for STUN
2013-08-20 15:29:36 -04:00
7fb07ff36e
* sip: #28675 : add correct contact header before sending OK
...
This ensures that SFLphone gets an ACK when doing calls with
sip2sip.info.
2013-08-20 15:23:18 -04:00
0be5951ff1
* #27724 : sdp: remove old contact header (if present)
2013-08-16 14:08:38 -04:00
dd4a7ebaf2
* #28529 : sip: ensure that ports are unique
2013-08-15 16:38:15 -04:00
0bde5fdeae
* #28529 : sip: add members for RTP port ranges
2013-08-15 11:08:51 -04:00
c166017951
* #28351 : daemon: rename iter to item for range-based for loops
...
"item" correctly reflects that in these loops, the object in question
is not an iterator but a reference to the actual object in the
collection.
2013-08-14 11:46:52 -04:00
f5417983e5
Merge branch 'master' into cpp11
2013-08-14 10:54:01 -04:00
d4ddf0677a
* #28358 : sip: ensure correct Contact Header is present for RINGING and OK
2013-08-13 13:36:11 -04:00
c1baecf0c1
* #28358 : sip: move addContactHeader into separate function
2013-08-13 11:44:03 -04:00
ba0f2f46a4
* #28351 : sipvoiplink: use range-based for loops
2013-08-12 17:26:26 -04:00
14c3a2ed0d
* #28215 : sip: on incoming calls, go to TRYING before going to RINGING
2013-08-09 11:11:18 -04:00
4d437efb11
* #27558 : audiortp: don't fail silently if destination is not set
2013-07-24 16:10:33 -04:00
802eda4573
* #26839 : sip: Android can use new pjsip API
2013-07-19 16:22:58 -04:00
e2d84da777
* #27364 : client: merge APIs
2013-07-17 14:32:23 -04:00
70810d43c0
* #26839 : client: move video_controls.h into client
2013-07-16 16:01:52 -04:00
720ae97e73
* #26839 : client: move configurationmanager.h into client/
2013-07-16 15:47:49 -04:00
9d20b5812d
* #26839 : client: move callmanager.h into client
2013-07-16 15:30:07 -04:00
a962dc8dd4
* #26839 : dbus: move into client/dbus
2013-07-16 15:01:17 -04:00
90cf7ac96c
* #26839 : sipvoiplink: cleanup
2013-07-15 14:18:16 -04:00
a66e98ba74
* #26839 : sipvoiplink: cleanup
2013-07-15 14:06:37 -04:00
62fcac17d6
* #27232 Modified build process and organisation of JNI layer
2013-07-12 12:08:48 -04:00
cea07778db
Merge branch 'master' into merge-dbus
...
Conflicts:
daemon/src/account.cpp
daemon/src/audio/audioloop.cpp
daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
daemon/src/managerimpl.cpp
daemon/src/managerimpl.h
daemon/src/sip/sipaccount.cpp
daemon/src/sip/siptransport.cpp
daemon/src/sip/sipvoiplink.cpp
2013-07-11 17:21:33 -04:00
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
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
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
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
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
58a780d5e9
* 25787: sip: use STUN address in VIA sent-by
...
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
248ad66a92
* #23661 : daemon: restore getCallList
2013-06-03 16:21:50 -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
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
ff9274dec4
build fixes
2013-04-19 00:01:05 +10:00
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
891bc0aad3
* #21631 : daemon: don't allow access to deleted call
...
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
...
This fixes the race-condition + SEGFAULT
2013-03-14 14:55:03 -04:00
