Issues
- 1
OTObject class
#52 opened by GoogleCodeExporter - 4
Error compiling telepresence
#50 opened by GoogleCodeExporter - 1
using vs2012 building telepresence
#51 opened by GoogleCodeExporter - 1
- 0
- 0
no video when linphone calls telepresence
#45 opened by GoogleCodeExporter - 0
previous client's image is still shown on video. how to fix the bug?
#46 opened by GoogleCodeExporter - 2
- 2
fail to compile
#40 opened by GoogleCodeExporter - 0
Fails to build with ffmpeg 2.x
#41 opened by GoogleCodeExporter - 0
- 0
telepresence behind NAT
#43 opened by GoogleCodeExporter - 0
no audio after a while
#44 opened by GoogleCodeExporter - 0
ssl key and too much latency
#36 opened by GoogleCodeExporter - 0
Placing Call on Hold, and then Resuming causes Telepresence Server to crash.
#37 opened by GoogleCodeExporter - 0
telepresence with Asterisk
#38 opened by GoogleCodeExporter - 0
DTLS/SRTP Intermittent Failure
#39 opened by GoogleCodeExporter - 1
- 0
fail to test telepresence
#34 opened by GoogleCodeExporter - 0
- 1
fail to test telepresence
#30 opened by GoogleCodeExporter - 1
UnRegister codec
#31 opened by GoogleCodeExporter - 1
ffmpeg used in telepresence
#32 opened by GoogleCodeExporter - 4
join the default bridge by webrtc client, but report 'transport error'
#29 opened by GoogleCodeExporter - 2
- 1
/usr/bin/ld: cannot find -lopenal
#28 opened by GoogleCodeExporter - 1
Failed to find libtinySAK
#23 opened by GoogleCodeExporter - 0
Technical Guide incomplete
#24 opened by GoogleCodeExporter - 0
- 0
Black Screen for Video
#26 opened by GoogleCodeExporter - 9
- 2
Crash
#21 opened by GoogleCodeExporter - 2
- 1
Crash on video call with conf-call.org
#20 opened by GoogleCodeExporter - 1
Crash on group video call with sipml5
#17 opened by GoogleCodeExporter - 3
- 1
Help regard AS mode of Presence Server
#15 opened by GoogleCodeExporter - 5
Crash on group audio call with sipml5
#16 opened by GoogleCodeExporter - 4
Conference call not work
#12 opened by GoogleCodeExporter - 0
[deleted issue]
#13 opened by GoogleCodeExporter - 0
[deleted issue]
#14 opened by GoogleCodeExporter - 1
INVITE not getting forwarded to the other SIP client even after both the clients get registered succesfully
#10 opened by GoogleCodeExporter - 2
Error when use make command
#11 opened by GoogleCodeExporter - 2
telepresence crash on INFO message
#7 opened by GoogleCodeExporter - 1
avfilter_unref_bufferp deprecated
#8 opened by GoogleCodeExporter - 1
segmentation fault and core dump during testing using http://conf-call.org/ WebRTC client
#9 opened by GoogleCodeExporter - 10
Problem with (at least) h263
#6 opened by GoogleCodeExporter - 1
Mute button not working properly
#4 opened by GoogleCodeExporter - 1
problem with VidyoGateway
#5 opened by GoogleCodeExporter - 1