fa3ef07
be more strict with ice candidate parsing
Daniel Gultsch created
Commit log
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
3439f40
show call log messages in conversation stream
Daniel Gultsch created
1dc88f3
avoid terminating twice
Daniel Gultsch created
82f9a77
be more conservative when parsing rtp content
Daniel Gultsch created
deaa76b
when using onNewIntent make sure to store intent otherwise onBackground might just overwrite it again
Daniel Gultsch created
609120c
only ever create one wake lock in rtpsessionactivity
Daniel Gultsch created
c9f7e17
use foreground service for ongoing call notification
Daniel Gultsch created
c6db651
allow all jingle states to transition into terminated
Daniel Gultsch created
5eea961
improved strategy for ignoring self addressed jingle messages
Daniel Gultsch created
7b382d2
include more human readable text in application errors
Daniel Gultsch created
07e671d
do not offer jingle calls when using Tor
Daniel Gultsch created
9d83981
respond with busy if there is anthor rtp session
Daniel Gultsch created
d19b5e0
show notification during ongoing call
Daniel Gultsch created