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
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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
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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
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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
Daniel Gultsch
bae9fc45c2
make rtcpMux optional
2021-10-31 10:20:58 +01:00
Daniel Gultsch
20e4d108d4
fixed regression of not handling jingle content map parsing failures
2021-10-05 15:43:05 +02:00
Daniel Gultsch
3ede2d00bd
remove logging
2021-10-02 16:54:19 +02:00
Daniel Gultsch
d2a387e82f
correctly calculate socks destination
2021-10-02 16:44:36 +02:00
Daniel Gultsch
da14f83a42
ensure all bytes are read in socks handshake. fixes #4188
2021-10-02 14:24:36 +02:00
Daniel Gultsch
63f5f8c89d
modify TODOs in JingleRtpConnection upon better understanding of the WebRTC stack
2021-09-08 10:47:34 +02:00
Daniel Gultsch
7466d12505
ring during device discovery
2021-05-22 19:37:20 +02:00
Daniel Gultsch
87f99d3570
Transferables interface needs to differentiate between 0 and null file size
2021-05-17 15:51:21 +02:00
Daniel Gultsch
67e5f839f1
ignore crypto callbacks when rtp session has already been terminated
2021-05-08 11:50:18 +02:00
Daniel Gultsch
9182a300c5
report fingerprint missmatch as securiy exception
2021-05-08 10:35:07 +02:00