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185 commits

Author SHA1 Message Date
Johan Pascal
2825585beb Enable AES256 for SRTP after ZRTP negociation but keep AES128 as default 
- setting in sip section srtp_crypto_suite in the configuration file
+ update ms
 2015-03-02 21:15:57 +01:00
Ben Sartor
e2dbfd5477 configfile now supports setting zrtps key agreements 
Signed-off-by: Johan Pascal <johan.pascal@belledonne-communications.com>
 2015-03-02 21:13:06 +01:00
Simon Morlat
49a91becb7 cleanups, add network simulation api to liblinphone 2015-02-24 18:11:12 +01:00
Johan Pascal
219451388d Merge remote-tracking branch 'origin/master' into dev_dtls 2015-01-27 10:42:06 +01:00
Simon Morlat
16180e2430 change the way payload type numbers are assigned, so that an application can support more payload type than the RTP profile table allows to contain. 
Compliance with RFC3264 (offer answer model) is improved, by reusing numbers in case of reINVITEs.
Fix memory leaks
Move offer/answer related tests into a new test suite.
 2015-01-21 22:38:46 +01:00
Johan Pascal
85ca8c3cac Merge remote-tracking branch 'origin/master' into dev_dtls 2015-01-14 00:16:11 +01:00
Gautier Pelloux-Prayer
9e6fa8ceb6 Doxygen: replace invalid @returns with @return 2015-01-09 11:42:05 +01:00
Guillaume BIENKOWSKI
9ff2841b18 Use debug trace instead of message 2015-01-07 10:55:24 +01:00
Jehan Monnier
25532e6285 ms2:stun api shall be public 2014-12-12 16:56:13 +01:00
Jehan Monnier
94b4002cbf fix ice parsing 2014-12-11 08:35:42 +01:00
Jehan Monnier
f0a3a75d99 fix ice issue when ufrag/pwd are present in sdp media description only 2014-12-10 14:20:48 +01:00
Simon Morlat
93493976b3 tester automatically creates unique accounts on flexisip server before running tests. This allows several developer to run the test suite simultaneously ! 2014-12-01 15:25:54 +01:00
Simon Morlat
2e515642f0 fix bad call state notification (Released) when receiving a call with incompatible codecs. 
Normally this should not trigger any notification.
Fix bug allowing two incoming calls to be notified if ICE is used.
 2014-11-18 16:01:51 +01:00
Simon Morlat
3d744d4070 * add test for ipv6 calls 
add linphone_call_media_in_progress() method for app to easily check that ice has finished or not its processing.
Update GTK app accordingly, so that adding video is no longer possible while ICE is in progress.
 2014-11-07 18:02:29 +01:00
Guillaume BIENKOWSKI
c63bf0ee9e Fix a verbose non-error 2014-11-04 17:28:27 +01:00
Simon Morlat
d91b0eaa28 fix video payload becoming unusable by mistake 2014-10-28 16:00:58 +01:00
Simon Morlat
cf3b09e35b forcely do not use some codecs under the following conditions: 
rate!=8000 and rate!=16000
no hardware AEC
AEC required (thus software)
webRTC AEC is used
not opus (because opus can accept 16khz in input)
 2014-10-27 15:53:55 +01:00
Simon Morlat
08539a0895 atomically sync the lpconfig on disk 2014-10-15 10:57:08 +02:00
Jehan Monnier
16b583b441 fix re-invite whiout sdp 2014-09-25 15:51:26 +02:00
Jehan Monnier
251ba960c5 add new functions linphone_core_add_listener to enable multiple vtable to be registered 2014-09-16 15:00:12 +02:00
Gautier Pelloux-Prayer
717db9fd8d Improved strict compilation flags 2014-09-11 15:46:05 +02:00
Ghislain MARY
eaa1d6bb1d Add reference count handling to the LinphoneCallParams object. 2014-09-01 14:58:32 +02:00
Ghislain MARY
3d512a019c Fix update of primary contact. 2014-08-22 14:47:45 +02:00
Ghislain MARY
b6a9bdeed5 Use belle-sip reference counting for LinphoneProxyConfig and LinphoneCall objects. 2014-08-21 16:20:12 +02:00
Ghislain MARY
31ab25d815 Handle PayloadType objects in the Python wrapper. 2014-08-13 16:10:48 +02:00
Simon Morlat
85e6548b59 add setting to disable symmetric rtp 2014-07-02 15:13:08 +02:00
François Grisez
960253d058 Rename the MKV_WRITER filter into MKV_RECORDER 2014-06-24 13:22:10 +02:00
Simon Morlat
ae4298faaf video recorder in place 2014-06-19 11:59:49 +02:00
Ghislain MARY
6f95bbc5d2 Fix bug 0001279: Wrong usage of n_active_streams in the media descriptions. 
Inactive streams are now allowed between active streams in the SDP.
 2014-06-10 13:26:41 +02:00
Simon Morlat
f9c01ebdb4 fix ICE status not updated at callee side in case of video mline rejected. 
add new tests.
 2014-06-05 16:00:41 +02:00
Ghislain MARY
dafdbb3444 Correctly handle negotiation of RTP profiles (APV/APVF/SAVP/SAVPF). 2014-06-04 11:59:50 +02:00
Ghislain MARY
2110281d2e Handle AVPF and SAVPF profiles. 2014-06-02 11:02:40 +02:00
Simon Morlat
fbc8f77e3a allow crypto lines to be configured from linphonerc, and improve code handling SRTP crypto lines 2014-05-21 13:11:13 +02:00
Simon Morlat
c8cdbdd543 allow setting explicit bitrate for video codecs 2014-05-07 12:04:53 +02:00
Simon Morlat
fb2b7ee40d ipv6 detection fix 2014-05-05 11:47:19 +02:00
Simon Morlat
eba52ef155 fix bad handling of link-local ipv6 addresses 2014-05-05 11:27:35 +02:00
Simon Morlat
4296c3945c update oRTP, fix bad error output, and restore UPDATE method in allow header (removed by mistake) 2014-05-02 23:22:36 +02:00
Simon Morlat
3a6aa9f08d deep modifications about audio & video codec bitrates are handled. 
- vbr codecs can automatically have different output bitrates depending on whether video is used and/or allowed total output bandwidth
- application can specify an output IP bitrate for a given codec, which allows to control the quality of vbr codecs.
Note: a belle-sip upgrade is required to fix a bug around channels parsing in rtpmap.
 2014-05-02 20:24:51 +02:00
Simon Morlat
f6d63524d3 fix declared number of channels in SDP for opus codec, to follow opus-rtp draft. 
add ugly hack to allow older versions of linphone to call new versions with opus.
 2014-05-01 12:14:05 +02:00
Gautier Pelloux-Prayer
4386f18b21 replace tabs with spaces and remove trailing spaces 2014-04-22 17:22:51 +02:00
Simon Morlat
7bd50e004f allow usage of system-choosen random ports. 
Implies a lot of refactoring in streams management.
 2014-04-07 17:37:50 +02:00
Simon Morlat
5cff5bebe5 fix bad enum cast and invalid enum translation 2014-03-28 17:12:49 +01:00
Simon Morlat
269f8d1c4e add new API to obtain full details about failures (calls, registration, events). 
Fix bug when receiving a 487 after cancelling call, resulting in a call waiting tone to be played.
 2014-03-21 18:15:28 +01:00
Simon Morlat
4d6894901e remove unused variable and update ms2 2014-03-20 16:25:15 +01:00
Simon Morlat
a7aab35b4f add function to override common telephony tones by wav files 2014-03-18 17:12:40 +01:00
Simon Morlat
e022c57627 fix few bugs around ipv6 detection 2014-03-09 22:18:37 +01:00
Guillaume Beraudo
92747748af Honor sip_randop_port during transport migration 2014-03-06 11:16:59 +01:00
Simon Morlat
95030951d1 add new function to play a file locally, in or out of calls. 
add new function to define a tone or wav file to be played automatically upon call errors
 2014-03-04 22:58:56 +01:00
Simon Morlat
9d5c1e7403 add possibility to set/get subject in SDP 2014-02-18 17:30:52 +01:00
Simon Morlat
9fc721e71c fixes and cleanup 2014-02-10 12:08:14 +01:00