Commit graph

400 commits

Author SHA1 Message Date
Daniel Gultsch 9187739450
play dial tones on STREAM_MUSIC when phone is silent
when the phone is silent only the first ~three tones are played when
attempting to play out the tone over STREAM_VOICE_CALL

it’s unclear exactly why this is the case (in the past we went back and forth
between STREAM_VOICE_CALL and STREAM_MUSIC) exactly to fix issues around silent
mode.
Apparently we failed to test this past three sounds.

This commit changes the stream back to music - but not generally as this was in
the past - but only for when the phone is on silent
2023-04-03 16:00:06 +02:00
Daniel Gultsch 0cec499565
make sure we don’t dispose video source twice 2022-12-30 12:16:19 +01:00
Daniel Gultsch 13606aae60
add todo item in turn server code 2022-12-29 14:53:05 +01:00
Daniel Gultsch ce0992036a
disable proximity sensor after switching from audio to video 2022-12-29 12:53:59 +01:00
Daniel Gultsch 909aa72b25 catch exception in getSignalingState() 2022-12-24 10:55:16 +01:00
Daniel Gultsch 36efd51a7f fix transports/descriptions not upgraded to jingle ft
fixes #4429
2022-12-20 19:28:47 +01:00
Daniel Gultsch 4ef4207593 show switch to video only if other party has caps
fixes #4421
2022-12-12 10:15:13 +01:00
Daniel Gultsch bb52962f0d delay candidates until after session-init/accept 2022-12-05 15:40:07 +01:00
Daniel Gultsch 2c7c44e957 null PeerConnection reference before disposing; otherwise getState() might be issued against disposed object 2022-12-01 20:46:18 +01:00
Daniel Gultsch 80d195d35e avoid race condition when restarting ICE 2022-11-30 17:32:46 +01:00
Daniel Gultsch 4e8ceadfbf prepare JingleRtpConnection for content-adds 2022-11-28 08:59:23 +01:00
Daniel Gultsch f4be142e4d add helper methods for content modification to RtpContentMap 2022-11-22 10:13:48 +01:00
Daniel Gultsch e2f98f6bbc ensure cc-ed proceed is equivalent to accept 2022-11-22 10:13:07 +01:00
Daniel Gultsch 9897fa3a45 rename initiateIceRestart to renegotiate to handle content adds 2022-11-21 09:10:01 +01:00
Daniel Gultsch 304205b2e3 take senders attr into account when converting to and from sdp 2022-11-20 17:00:40 +01:00
Daniel Gultsch 59ea66ca78 make sure VideoSourceWrapper is stored in property 2022-11-19 14:19:07 +01:00
Daniel Gultsch 27d8da2ab4 refactor WebRTCWrapper to allow for track adds 2022-11-19 13:03:34 +01:00
Daniel Gultsch 8fb2c11771 use plurals for missed call strings 2022-11-19 08:14:50 +01:00
Daniel Gultsch 6ececb4d2b refactor webrtc video source + capture code 2022-11-12 13:37:56 +01:00
Dmitry Markin a6b88ba9e9
Add missed call notifications
Co-authored-by: Daniel Gultsch <daniel@gultsch.de>
2022-08-29 12:41:35 +02:00
Daniel Gultsch e8736d5f1b bump guava library 2022-08-22 11:29:04 +02:00
Daniel Gultsch fe3433e427 do not accept empty credentials as ice-restart 2022-08-10 09:11:09 +02:00
Daniel Gultsch d41020ccf3 ignore race condition after reject from notification
fixes #4351
fixes #4261
2022-08-05 10:46:15 +02:00
Daniel Gultsch 62a379862e jingle rtp: improve logging and error reporting 2022-08-01 10:14:49 +02:00
Daniel Gultsch 42bd8e6d61 minor code clean up 2022-06-22 07:56:44 +02:00
Stephen Paul Weber 78048bbd3d Enable WebRTC-BindUsingInterfaceName/Enabled/
This makes 464XLAT networks (such as T-Mobile LTE) work.

https://bugs.chromium.org/p/webrtc/issues/detail?id=10707
2022-03-10 16:29:00 +01:00
Daniel Gultsch 372078629b fix ice candidate sending when different credentials are used 2022-02-25 17:26:36 +01:00
Daniel Gultsch 1f772df74f remove security check that ensures rtp connection was properly finished
this only causes race conditions
2022-02-25 16:24:16 +01:00
Daniel Gultsch d6be6ddd18 use full file name for all new files 2022-02-22 16:05:02 +01:00
Daniel Gultsch d7f38a3e5a fix precondition for timeout handling 2022-02-12 10:19:54 +01:00
Daniel Gultsch b6442c0bd4 add Samsung S4 to hardware aec blacklist
fixes #4267
2022-01-18 11:30:23 +01:00
Daniel Gultsch db834a1f07 indicate call reconnect in notification 2021-11-19 12:26:11 +01:00
Daniel Gultsch f8a94161db don't play tone going from connect->reconnect->connect 2021-11-19 12:25:27 +01:00
Daniel Gultsch a508a81553 externalize rtc config generation into seperate method 2021-11-17 11:33:15 +01:00
Daniel Gultsch 61fb38cd84 clean up some error handling error ICE restarts 2021-11-17 10:49:16 +01:00
Daniel Gultsch 0a18c8613f assume credentials are the same for all contents when restarting ICE 2021-11-16 17:08:34 +01:00
Daniel Gultsch abb671616c synchronize setDescription calls 2021-11-16 15:17:12 +01:00
Daniel Gultsch 297a843b9c use implicit rollback (needed to be enabled on libwebrtc) 2021-11-16 13:17:10 +01:00
Daniel Gultsch 0698fa0d8c store peer dtls setup for later use in ice restart 2021-11-16 11:21:11 +01:00
Daniel Gultsch 70b5d8d81a set proper peer dtls setup on ice restart received 2021-11-15 21:49:31 +01:00
Daniel Gultsch 0a3947b8e3 terminate with application failure when failing to apply ICE restart
This is fairly unlikely to happen in practice
2021-11-15 17:18:45 +01:00
Daniel Gultsch 3f402b132b respond with tie-break to prevent ICE restart race 2021-11-15 13:03:04 +01:00
Daniel Gultsch 5b80c62a63 treat transport-info w/o candidates and changed credentials as offer 2021-11-14 18:22:18 +01:00
Daniel Gultsch 717c83753f delay discovered ice candidates until JingleRtpConnection is ready to receive
otherwise setLocalDescription and the arrival of first candidates might race
the rtpContentDescription being set
2021-11-11 21:02:17 +01:00
Daniel Gultsch b6dee6da6a reverse: webrtc: include oldState in onConnectionChange
turns out we don’t need it and a better way is for RtpConnection to keep track of *all*
states in the current generation
2021-11-11 17:05:36 +01:00
Daniel Gultsch 9c3f55bef2 use stopwatch to keep track of jingle rtp session duration 2021-11-11 16:52:18 +01:00
Daniel Gultsch 9843b72f6f always use Camera2Enumerator 2021-11-11 15:23:45 +01:00
Daniel Gultsch 61851e5f84 do not automacially hang up failed webrtc sessions 2021-11-11 14:40:15 +01:00
Daniel Gultsch 4ec0996dff webrtc: include oldState in onConnectionChange 2021-11-11 11:19:37 +01:00
Daniel Gultsch fda45a7c86 use implicit descriptions for WebRTC
using the parameter-free form of setLocalDescription() solves some potential race conditions
that can occur - especially once we introduce restartIce()
2021-11-10 16:40:24 +01:00