Commit graph

116 commits

Author SHA1 Message Date
Simon Morlat
589d3cd540 appnaping improvements
- take a background task during the ice gatethering for incoming call
- add timestamps to gtk debug window, which was required to investigate the issue.

Requires up to date belle-sip
2015-02-06 19:31:42 +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
Jehan Monnier
06fc0526ec multicast impl 2015-02-04 12:17:14 +01:00
Simon Morlat
53bc2cd5a0 add tests to check sips and ipv6 support of flexisip 2015-02-02 18:13:55 +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
Guillaume BIENKOWSKI
838520350c Use const for sal_address_is_ipv6() 2015-01-15 17:19:58 +01:00
Guillaume BIENKOWSKI
0c4e7456d9 Prevent creating sip addresses which are not valid when using them 2015-01-15 17:19:45 +01:00
Johan Pascal
e93a80f322 Merge remote-tracking branch 'origin/master' into dev_dtls 2015-01-05 18:17:05 +01:00
Jehan Monnier
3939954500 add ssrc attribute in case of srtp dtls 2014-12-15 13:04:40 +01:00
Jehan Monnier
20fe706f7d check if number of ice candidate does not exceed stirage size 2014-12-11 14:56:37 +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
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
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
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
Simon Morlat
3e1e3c93f4 fix crash when declining an update
refine conditions for refusing an UPDATE
2014-09-25 14:15:06 +02:00
Simon Morlat
c6a3053756 fix incorrectly named functions and compilation errors due to merge 2014-09-08 19:05:43 +02:00
Simon Morlat
f4a4a6440b Support for incoming UPDATEs within dialog.
For tests, the possibility to send an UPDATE with linphone_core_update_call() has been added thanks to a $
Added possibility to configure Supported SIP header.
2014-09-08 19:02:29 +02:00
Gautier Pelloux-Prayer
aed5bd789a Improve tunnel test: check that SIP packet actually use the tunnel 2014-09-08 14:20:31 +02:00
Gautier Pelloux-Prayer
82ec76a4e1 Reuse previous nonce if outbound proxy realm is set to avoid reauthentication 2014-07-25 14:37:01 +02:00
Ghislain MARY
e41203a4c5 Fix compilation for Windows Phone 8. 2014-06-25 17:46:44 +02:00
Gautier Pelloux-Prayer
5cf381b667 add linphone_core_get_user_agent to retrieve local user agent 2014-06-24 14:17:11 +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
Johan Pascal
3c918dfd8b Merge branch 'master' of git.linphone.org:linphone into dev_filetransfer 2014-06-09 13:40:06 +02:00
Jehan Monnier
ba5c902bba add option sip_update to linphonerc to disable SIP UPDATE 2014-06-03 10:24:44 +02:00
Ghislain MARY
0cd71d6548 Fix issue with rtcp-fb attributes in SDP of response. 2014-06-02 11:02:41 +02:00
Ghislain MARY
1a5f37eaba Allow activation of AVPF for a call based on the proxy configuration. 2014-06-02 11:02:40 +02:00
Ghislain MARY
2110281d2e Handle AVPF and SAVPF profiles. 2014-06-02 11:02:40 +02:00
Johan Pascal
c10b5f652b File transfer implemented following RCS5.1 recommendation
- memory leaks to be fixed
2014-05-29 00:10:49 +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
Jehan Monnier
bb6d660594 rework proxy config management edit()/done() method to only send unregister message when really needed 2014-05-20 18:38:56 +02:00
Simon Morlat
7553aa6492 - linphone now puts Route headers in requests (except register) for outbound proxy configurations, according to RFC3261
This behavior can be reverted by putting [sip]->use_no_initial_route=1 in the configuration file.
- accept presence NOTIFY without bodies, instead of replying 415
- remove belle-sip warning at start due to stack not created early enough.
2014-04-25 23:13:26 +02:00
Gautier Pelloux-Prayer
ceb8533cf9 Merge branch 'quality_reporting'
Conflicts:
	coreapi/linphonecall.c
	mediastreamer2
	tester/call_tester.c
2014-04-22 14:42:20 +02:00
Gautier Pelloux-Prayer
f3efcb1286 Quality reporting: fix unit tests, and remove some trailing spaces 2014-04-17 16:58:50 +02:00
Gautier Pelloux-Prayer
1cba3da32d Quality reporting: fill 'from-tag' and 'to-tag' fields 2014-04-17 16:38:39 +02:00
Jehan Monnier
43aa6ef34f change encryption state management 2014-04-17 16:22:49 +02:00
Simon Morlat
a8176a398d rework SRTP support so that recv and send key can be set and updated independently. 2014-04-08 23:41:14 +02:00
Simon Morlat
266207c5f0 change behavior of linphone_core_get_sip_transports() if random port selection was specified.
Only linphone_core_get_sip_transports_used() will return the real port if random port selection was specified.
2014-03-26 11:30:07 +01:00
Simon Morlat
c25273e9ca add C function to disable chat 2014-03-25 12:48:50 +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
Ghislain MARY
4c3baa0528 Define RTCP XR structure in oRTP instead of sal. 2014-03-19 16:07:04 +01:00
Jehan Monnier
260e7e1d5c minimal SIP UPDATE support 2014-03-18 09:08:25 +01:00
Simon Morlat
a0b7b1f3b0 rework implementation of SUBSCRIBEs delayed to successful registration.
In case of network errors, SUBSCRIBE could be sent before registration refresh, this should be fixed.
2014-03-11 17:25:07 +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
Ghislain MARY
8bc0b2e8a4 Handle RTCP XR SDP attribute parsing and creation. 2014-03-04 16:28:23 +01:00
Simon Morlat
1af4a7c091 improve SDP<->SalMediaDescription conversion and offer answer algorithm 2014-02-28 16:31:05 +01:00
Ghislain MARY
ca5f624bc6 Divide huge function in smaller blocks. 2014-02-24 14:48:47 +01:00
Guillaume BIENKOWSKI
b57f8b1526 Added 302 redirection support.
+ removed macros for sal_op_get_contact compatibility
2014-02-19 10:50:36 +01:00
Simon Morlat
9d5c1e7403 add possibility to set/get subject in SDP 2014-02-18 17:30:52 +01:00