I am using linphone opensource code for my code, I get a few questions
on my transplating.
1. When the three-way handshake has done,the RTP packets is
transported normally, if one of the participants is exit accidently(no sip
message sending) ,the other is still running, on the Debug windows,it keeps
showing the the RTCP packet loss on the linphone 3.5.2. However it does not show
on the linphone 3.4.3 ,I searched the code, the function audio_stream_alive is
called after the audio_stream_iterate ,so the thread keeps handling the rtcp
packets lossing event. It supposed to take the audio_stream_alive ahead.
2. On the Windows Os ,the linphone 3.5.2.exe gets those problems, when
the Debug window is cleared ,It does not work sometime (no debug info shows)
until resetting the programme. and the URL editbox does capture the input focus
on uncertain conditions.
You guys do a great job on the linphone-developer, I am excited to see
each time the release becomes more perfect .
please forgive my syntax errors.