31dfb0c
cache useTor information in activity
Daniel Gultsch created
Commit log
72c551d
bump to 2.8.0-beta.2
Daniel Gultsch created
a127603
ensure that rtp connection is registered with connection manager
Daniel Gultsch created
c20c40a
ensure webrtc connection gets closed after connection failure
Daniel Gultsch created
7dfd47a
better crash than leave WebRTCWrapper unclosed
Daniel Gultsch created
934b98d
add microphone availability check
Daniel Gultsch created
ebda472
version bump
Daniel Gultsch created
48f7523
paint local mic off button in pip
Daniel Gultsch created
644e5aa
remove video sinks when calling onStop. otherwise going in and out foreground will give us endless sinks
Daniel Gultsch created
16d34c2
parse turns and stuns (regression from earlier commit)
Daniel Gultsch created
ab26816
allow pip during connecting
Daniel Gultsch created
2f437ea
ignore iq errors if session has already been terminated
Daniel Gultsch created
fa3ef07
be more strict with ice candidate parsing
Daniel Gultsch created
0a18ab3
fixed 215 credential detection
Daniel Gultsch created
8472712
play notification sound pre notification categories
Daniel Gultsch created
e545e95
getMedia() would throw null pointer when called after going from proposed to some error state
Daniel Gultsch created
ea2ed85
support picture in picture for video calls
Daniel Gultsch created
21e412e
only show remote video when connected
Daniel Gultsch created
0c4f0c0
improve busy behaviour with multiple devices
Daniel Gultsch created
4558b9a
select proper media for retry
Daniel Gultsch created
45d5d1f
capture in ~1920 resolution when available
Daniel Gultsch created
b95d406
use more approriate reason when failing because of parse errors
Daniel Gultsch created
ec6bcec
use different aspect ratio for landscape
Daniel Gultsch created
36e1179
put 'video' in ongoing video call notification
Daniel Gultsch created
d7e93e1
add a couple of todos to RtpSessionActivity
Daniel Gultsch created
f995965
parse 0339 source groups from sdp
Daniel Gultsch created
01a9a52
show enable/disable video in video calls
Daniel Gultsch created
445009c
request camera permissions
Daniel Gultsch created
5a20faa
show 'incoming video cal' notification
Daniel Gultsch created
d4788fc
display video call based on availability
Daniel Gultsch created
b4df191
make seperate menu items for audio and video calls
Daniel Gultsch created
17d9b02
properly paint local video over remote
Daniel Gultsch created
d057ae3
transmit media from proposal to actual session
Daniel Gultsch created
8c273e7
parse media from session proposal
Daniel Gultsch created
1489dba
release resource. stop caputuring when webrtc ends
Daniel Gultsch created
b20b00e
use toolbar to display status text in RtpSessionActivity
Daniel Gultsch created
339bdae
rudimentary video caputuring
Daniel Gultsch created
bfb9a62
complete list of reasons
Daniel Gultsch created
dd42a6b
don’t transition when calling endCall and session was already terminated
Daniel Gultsch created
65b4366
RtpConnection: synchronize all externally call methods to guard state transitions
Daniel Gultsch created
172d2c6
depulicate 'propose's when doing mam catchup
Daniel Gultsch created
e16e0d8
cancle ongoing jingle sessions on xmpp rebind
Daniel Gultsch created
493ca68
add <rtcp-mux/> in description
Daniel Gultsch created
ef22071
turn proximity wake lock and/off depending on speaker configuration
Daniel Gultsch created
9bc264b
do not use proximity wake lock on speaker phone
Daniel Gultsch created
981aeaf
make mute and speaker button work
Daniel Gultsch created
b924a63
copy audio manager from AppRTCDemo
Daniel Gultsch created
5b98107
put jingle messages in MAM and parse call log during catchup
Daniel Gultsch created
9a41d11
do not show context menu for call logs
Daniel Gultsch created
4be2309
more conditions under which to print call log
Daniel Gultsch created