Commit graph

502 commits

Author SHA1 Message Date
Jehan Monnier
fd0a7cfd73 add more DTLS tests 2015-02-13 18:01:56 +01:00
Simon Morlat
90ee807c7e event queue needs to be unregistered since RtpSession is kept 2015-02-11 09:31:09 +01:00
Simon Morlat
e9f89d162f do not unregister event queue while msticker is still running with the RtpSession.
The stream must be first stopped, then queue can be destroyed safely
2015-02-10 15:12:21 +01:00
Simon Morlat
5dbc66938c rename LinphoneCallParamsMediaDirection into LinphoneMediaDirection, as it has no reason to be attached to LinphoneCallParams and could be re-used in other contexts. 2015-02-07 13:30:30 +01:00
Simon Morlat
7b62f3313d repair linphone, broken by previous commit implementing stream directions. 2015-02-07 13:23:33 +01:00
Ghislain MARY
0d94ad277f Handle media direction when creating the local media description according to the call params. 2015-02-06 19:04:11 +01:00
Ghislain MARY
1d080cb1f5 Add default_max_bandwidth parameter and apply it to video when no bandwidth is specified in the signalling. 2015-02-05 17:54:03 +01:00
Simon Morlat
7798932b93 fix to previous commit: a real local interface must be decided and bound to to send multicast.
For unicast calls, continue to bind to 0.0.0.0 as we need it for multi-homed environments.
2015-02-05 01:16:05 +01:00
Simon Morlat
9b95f24fc3 avoid multiple warnings due to ms_is_multicast() not used correctly, make code stream type agnostic 2015-02-05 00:09:46 +01:00
François Grisez
2ef0e530b6 Fix compilation. Compatibility with C99 2015-02-04 16:19:33 +01:00
Jehan Monnier
a08aacea3f add android support for multicast rtp 2015-02-04 12:17:15 +01:00
Jehan Monnier
3e1a1430f4 add Android wifi lock management at LinphoneCall level 2015-02-04 12:17:15 +01:00
Jehan Monnier
06fc0526ec multicast impl 2015-02-04 12:17:14 +01:00
Margaux Clerc
8484642ce2 Add video ifdef 2015-02-04 10:57:38 +01:00
Ghislain MARY
ecf4ba1b5c Enable setting a different video window id for each call. 2015-02-02 14:25:57 +01:00
Johan Pascal
92c1c6d4ac code cleaning
remove useless DTLS debug traces
2015-01-31 22:40:56 +01:00
Simon Morlat
cced42ebc5 add test for generic CN, update oRTP and ms2 2015-01-30 20:00:35 +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
Johan Pascal
54e91b6394 Update srtp to rely on stream sessions structure and not complete stream structure 2015-01-12 15:35:28 +01:00
Gautier Pelloux-Prayer
9e6fa8ceb6 Doxygen: replace invalid @returns with @return 2015-01-09 11:42:05 +01:00
Johan Pascal
5fdf3b82ba Enable DTLS-SRTP protection on video stream 2015-01-08 12:56:10 +01:00
Johan Pascal
e93a80f322 Merge remote-tracking branch 'origin/master' into dev_dtls 2015-01-05 18:17:05 +01:00
Johan Pascal
88e2ba7625 move srtp from ortp to mediastreamer2 2015-01-05 15:02:52 +01:00
Jehan Monnier
7fe891b4ae add ice option to enable backward compatibility with previous version of ice 2014-12-27 19:45:19 +01:00
Jehan Monnier
d7437ef1f5 enable ice with tunnel 2014-12-22 21:47:35 +01:00
Simon Morlat
8f633b21fc fix stack overflow. 2014-12-18 16:17:26 +01:00
Ghislain MARY
33aaac313b Fix compilation with Visual Studio. 2014-12-16 11:32:51 +01:00
Jehan Monnier
551cb17583 fix crash in outgoing call case 2014-12-16 11:22:14 +01:00
Jehan Monnier
8d13609402 various ice fix for better interwork 2014-12-15 16:02:27 +01:00
Jehan Monnier
3939954500 add ssrc attribute in case of srtp dtls 2014-12-15 13:04:40 +01:00
Simon Morlat
1ac1cd5fe8 fix RtpTransport leak on video side as well 2014-12-12 18:56:14 +01:00
Simon Morlat
f48780782c fix compilation 2014-12-12 16:54:30 +01:00
Simon Morlat
1cf049cabd fix leak of RtpTransport when call is updated/paused. Fix reporting of bandwidth, which was displayed even if the stream was inactive. 2014-12-12 16:46:37 +01:00
Jehan Monnier
45e1da743c make sure rtp destination is change as soon as ice is terminated 2014-12-12 15:47:09 +01:00
François Grisez
db5fc6ea89 Fix the reading of .linpohne.ecstate 2014-12-12 12:12:44 +01:00
Johan Pascal
8637eacae5 Merge remote-tracking branch 'origin/master' into dev_dtls
Conflicts:
	mediastreamer2
	oRTP
2014-12-10 15:18:14 +01:00
Johan Pascal
b986af3733 Add dtls srtp 2014-12-10 15:11:36 +01:00
Guillaume BIENKOWSKI
dc2d250d99 Perform sound card usage check when a call is dismissed or when a call transitions to pausing state 2014-12-10 14:23:10 +01:00
Jehan Monnier
273207b25c differentiate rtp/rtcp bandwidth reporting 2014-12-08 18:07:06 +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
Johan Pascal
e3b3a5aa5b move ZRTP management from oRTP to Mediastreamer2 using transport modifier 2014-11-16 23:26:14 +01:00
Simon Morlat
d0095948b9 Allow the tunnel mode to work with any proxy config (except for sips destinations)
fix the tunnel tests, that were not working correctly.
2014-11-14 20:17:54 +01:00
Gautier Pelloux-Prayer
a76aa60ff5 Free dtmf timer on call destruction, if needed 2014-11-12 17:32:24 +01:00
Gautier Pelloux-Prayer
0aabc05145 Add linphone_call_send_dtmfs method to allow sending a DTMF sequence instead of a single one, and add a test suite 2014-11-12 14:58:19 +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
Gautier Pelloux-Prayer
abe5a19431 Do not crash if SDP could not be parsed and had some unit tests. - avoid crash if missing SDP in REINVITE ACK - resume previous media parametrs instead of aborting call in case of invalid SDP in REINVITE 2014-11-06 17:25:36 +01:00
Jehan Monnier
78c11c8f6e compute call log duration since connected state instead of from call creation 2014-10-20 22:49:40 +02:00
Simon Morlat
965add9d6e add new states LinphoneCallEarlyUpdating and LinphoneCallEarlyUpdatedByRemote to properly handle the early dialog UPDATE scenarios.
fix test suite.
2014-10-20 15:10:40 +02:00