narayana/conversations-classic
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111 commits

Author SHA1 Message Date
Daniel Gultsch 6a6c9fb3bf ignore race condition when toggling fixes #3822 2020-07-09 19:14:28 +02:00
Daniel Gultsch fada3a63c9 store entire transport info for after session was accepted. fixes #3790 2020-06-22 18:07:27 +02:00
Daniel Gultsch 169ee99afa do not attempt to reject call if session had already ended. fixes #3798 2020-06-18 20:32:58 +02:00
Daniel Gultsch 0ba4892d3e RTP: write log message on background thread 2020-06-12 09:08:09 +02:00
Daniel Gultsch 644ad99520 create rtp end user state for connection lost. fixes #3769 2020-06-12 07:57:11 +02:00
Daniel Gultsch 552e17e39a remember terminal RTP session state 
if the activity is not connected during finish it won’t receive the last end user state.

this code remembers it even if the actual session is already gone. so when activity reconnects and
we can’t find the real rtp session we can look up the last state instead.
 2020-06-11 21:17:15 +02:00
Daniel Gultsch 1853242c66 do not throw when finishing jingle ft twice. fixes #3765 
the state machine in jingle file transfer does not prevent that the connection
is being finished twice
 2020-06-07 15:00:00 +02:00
Daniel Gultsch 637c208f55 ask for resource and use jingle direct init when JMI is not available. fixes #3751 2020-05-30 14:56:12 +02:00
Daniel Gultsch 59d1a2982e RtpSessionActivity: throw instead of finish when session wasn’t found 2020-05-28 09:22:58 +02:00
Daniel Gultsch a0920b83e2 use Account.getDomain() for direct access to domain jid 2020-05-17 10:24:46 +02:00
Daniel Gultsch b6703dbe38 switch xmpp-addr to jxmpp-jid 2020-05-15 17:06:16 +02:00
Daniel Gultsch 2c4788b7c7 send retract when unable to setup webrtc as initiator. fixes #3717 2020-05-11 12:20:32 +02:00
Daniel Gultsch 4d3d3a7038 tie breaking racing jingle message proposals. fixes #3698 2020-05-10 14:09:16 +02:00
Daniel Gultsch 2c5bed61a1 introduce extra RTP state to handle going from sending proceed to receiving retract 2020-05-09 21:35:21 +02:00
Daniel Gultsch f4247379bd catch UnsatisfiedLinkError when trying to init libwebrtc. fixes #3707 2020-05-09 19:48:54 +02:00
Daniel Gultsch 92fc22b313 show call duration in audio calls. fixes #3708 2020-05-09 11:14:39 +02:00
Daniel Gultsch 285c750e69 throw IllegalStateException when trying to finish from a non terminal state 2020-05-08 18:36:52 +02:00
Daniel Gultsch 350fc57d87 properly wrap IPv6 addresses 2020-05-08 17:52:41 +02:00
Daniel Gultsch 5af4c865a7 make sure we finsh() the connection after transitioning into terminal state 2020-05-08 17:22:27 +02:00
Daniel Gultsch 3c3f5d8e6f mark missed calls as unread (bold) in overview. fixes #3687 2020-05-03 18:07:00 +02:00
Daniel Gultsch e70b6eec98 do not mirror back camera. fixes #3693 2020-05-03 11:54:58 +02:00
Daniel Gultsch 63ddd97b6b add button to switch cameras during video call 
RIP symmetry :-(

fixes #3683
 2020-05-02 17:15:50 +02:00
Daniel Gultsch f7a0d2031a disable TLS cert validation for stun/turn server 
turns out libwebrtc doesn’t use the system root CA store but comes with only a few default CAs.

in anyway we will probably only use tcp/443 to bypass firewalls and not to actually secure anything.
 2020-05-01 20:17:23 +02:00
Daniel Gultsch 7ac5e8e828 properly close WebRTCWrapper even when init failed 2020-05-01 13:56:24 +02:00
Daniel Gultsch deae2b109f do not crash UI after ignoring improperly formatted jingle init 2020-04-29 15:54:02 +02:00
Daniel Gultsch 8a586527c4 check if setting local description was succesful 2020-04-29 15:32:27 +02:00
Daniel Gultsch a49d69c878 parse candidates from session-init and session-accept 2020-04-29 10:36:54 +02:00
Daniel Gultsch 0d4b175760 better failure behaviour after direct init from jitsi 2020-04-29 08:51:38 +02:00
Daniel Gultsch fc4397e5b9 play busy and dial tones 2020-04-27 17:51:38 +02:00
Daniel Gultsch 9fbf73d1ea do not log failed calls that never rang 2020-04-26 10:38:19 +02:00
Daniel Gultsch 4f5415ecba terminated rtp connection do not count as busy 2020-04-24 09:41:54 +02:00
Daniel Gultsch c0036b4ca6 increase busy timeout to 30s 2020-04-24 09:16:59 +02:00
Daniel Gultsch a008993d06 add 20s busy timeout to incoming calls 2020-04-22 21:59:20 +02:00
Daniel Gultsch 892d913e2c parsing iq erros also need to finish the connection 2020-04-22 18:42:07 +02:00
Daniel Gultsch 9fa9ca9cbc catch securityException when parsing rtp description 2020-04-22 16:35:08 +02:00
Daniel Gultsch 9afac21b0b don’t throw when user double taps accept button 2020-04-22 14:49:48 +02:00
Daniel Gultsch 876b1149d5 avoid double termination after failed connection 2020-04-21 22:59:54 +02:00
Daniel Gultsch eb911b8196 show 215 status in server info 2020-04-21 11:40:05 +02:00
Daniel Gultsch 1cc0dfad84 move sdp logging to different tag 2020-04-20 15:57:31 +02:00
Daniel Gultsch 5a0979b41e store 'ended call' when ended from proceed 2020-04-20 15:57:31 +02:00
Daniel Gultsch a12760300c ensure that rtp connection is registered with connection manager 2020-04-20 15:57:30 +02:00
Daniel Gultsch c20c40a807 ensure webrtc connection gets closed after connection failure 2020-04-20 15:57:30 +02:00
Daniel Gultsch 7dfd47a5c4 better crash than leave WebRTCWrapper unclosed 2020-04-20 15:57:30 +02:00
Daniel Gultsch 934b98d199 add microphone availability check 2020-04-20 15:57:30 +02:00
Daniel Gultsch 16d34c2ba0 parse turns and stuns (regression from earlier commit) 2020-04-20 15:57:30 +02:00
Daniel Gultsch 2f437ea845 ignore iq errors if session has already been terminated 2020-04-20 15:57:30 +02:00
Daniel Gultsch fa3ef07580 be more strict with ice candidate parsing 2020-04-20 15:57:30 +02:00
Daniel Gultsch 0a18ab35c0 fixed 215 credential detection 2020-04-20 15:57:30 +02:00
Daniel Gultsch e545e95d39 getMedia() would throw null pointer when called after going from proposed to some error state 2020-04-20 15:57:30 +02:00
Daniel Gultsch b95d406e61 use more approriate reason when failing because of parse errors 2020-04-20 15:57:30 +02:00