Damian Minkov
4b124d7b8a
Updates the license headers.
11 years ago
Boris Grozev
ce54491504
Adapts to the changes to libjitsi moving the Recorder interface to the "recording" package.
12 years ago
hristoterezov
db6a1b3fa0
Implements publishing a conference in chat room (Merged from publish-conference branch).
13 years ago
Emil Ivov
85511815b3
Allows use of distinct port ranges for audio and video streams. Distinct ranges can be defined through: net.java.sip.communicator.service.protocol.[MAX|MIN]_[AUDIO|VIDEO]_PORT_NUMBER. Use of a single range from 5000 6000 is still the default and is still configurable via the previously existing properties
13 years ago
Damian Minkov
069ffa44a8
Implements call waiting disabled property and reject incoming calls property on busy provider.
13 years ago
Lyubomir Marinov
5f8c5a9063
Refactors the cross-protocol conference support.
14 years ago
Lyubomir Marinov
89d68994ef
Switches Jitsi trunk to libjitsi.
14 years ago
Damian Minkov
8b81846a4b
Adds option to hangup peer with error code and reason in OperationSetBasicTelephony.
14 years ago
Sebastien Vincent
db293d098b
Ongoing work on cross protocol conference calls. Adds missing portaudio hotplug patch.
15 years ago
Emil Ivov
9feab56577
Changes project name in source license headers from SIP Communicator to Jitsi
15 years ago
Lyubomir Marinov
ef36f9a3c6
Fixes issue #899 : Incorrect Javadoc for function createCall.
16 years ago
Lyubomir Marinov
a951b90be6
Lists the format supported by the call Recorder into it in order to have them defined in one place, introduced MediaException and throws exceptions when the call recording cannot be started successfully.
16 years ago
Lyubomir Marinov
fe81fd4212
Tries to fix a "javax.media.NoDataSinkException: Cannot find a DataSink for: null" reported by Emil Ivov on the dev mailing list in the thread "Call Recording".
16 years ago
Lyubomir Marinov
347408cc5b
Commits callRecording.patch and recordButton.png provided by Dmitri Melnikov on the dev mailing list in the thread "Call Recording".
16 years ago
Emil Ivov
9bb144b9fa
Moves code from call peer's sip implementation to a super class that would be shared with the jabber implementation
16 years ago
Damian Minkov
bf4d3f5ef6
Remove unused code, muting individual call peers.
16 years ago
Damian Minkov
097db0acd7
Fix mute button behaviour in conference calls.
16 years ago
Damian Minkov
82f658f998
Adds local user sound level indicators and their implementation. Change Players to be Processors in neomedia service impl, in order to have control on codec chain and to add there the sound level indicators.
17 years ago
Emil Ivov
13479291fd
Migrates SIP over to neomedia. (Work in progress). Media port initialization.
17 years ago
Yana Stamcheva
8a0d4781ec
Local user sound level indicator listeners and implementation in the UI + some fixes in the call conference ui
17 years ago
Lyubomir Marinov
e4eeb97c85
- Removes a SuppressWarnings because its warning can be resolved gracefully.
...
- Fixes a few javadocs.
17 years ago
Emil Ivov
b6ae58d807
Renames occurrences of callParticipant to callPeer so that it would better reflect our new Call architecture that also includes conferencing and ConferenceMembers
17 years ago
Emil Ivov
3f7d9ddd26
Renames CallParticipant to CallPeer so that it would better reflect our new Call architecture that also includes conferencing and ConferenceMembers
17 years ago
Emil Ivov
4884f5a92b
Renames CallParticipant to CallPeer so that it would better reflect our new Call architecture that also includes conferencing and ConferenceMembers
17 years ago
Romain Kuntz
e7e71f8c1b
Committed the rest of the ZRTP implementation from Emanuel Onica:
...
- added the secure button on the call panel
- added a new operation set for secure call
- removed implementation of the secure mode from jabber/mock
18 years ago
Emil Ivov
02baebb228
Asserts use of TCP when it has been set as the default transport during protocol initialization.
...
Implements Alan Kelly's fix that optimizes SRV resolution code
Removes redundant naming in SipRegistrarConnection
Removes use of resource strings for keep alive method selection as they should not be localized
Renames OperationSetBasicTelephony.getSecure() to isSecure()
18 years ago
Romain Kuntz
0fb8343cf5
Merged ZRTP-related code
18 years ago
Lyubomir Marinov
8589c49d20
Introduces support for muting a call.
18 years ago
Emil Ivov
a2491f9b12
Committing the on hold patch from Lubomir Marinov
18 years ago
Emil Ivov
a68fed43a2
Support for SIP (Work in Progress)
...
Added throws clauses to most methods.
20 years ago
Emil Ivov
476d7118ba
Woking on implementing support for SIP
...
Added a createCall method with a contact param
20 years ago
Emil Ivov
1a04e68bb4
Code format
20 years ago
Emil Ivov
c82e7e1327
Initial sip-communicator-1.0 commit
21 years ago