Seems it is CPU resource limitation. Is it possible to synchronize audio and
video tracks after such CPU resource limitations, during record process. I
mean, how to determine that desynchronisation happened in playing process?
Maybe some RTCP packet data information can help or something else?
Using LIVE555 library as streamer in UNICAST UDP mode found if i'm using video
and audio tracks, and during playing mode somebody else make a new connection,
the client with previous connection receive video frame with about 500ms delay.
Trying to open new connections several times, the first c
I'm analyzing RTSP stream data and need to have the time difference in
milliseconds between each video frame. Stream server send timestamp values like
below:
1271120994
1271124594
1271128194
...
...
...
but it does not look like difference between frames is coming in milliseconds
or microseco
How Live555 client side library makes UDP ports reachable from the global
network for incoming UDP traffic when both client and server are under NAT and
all UDP ports are not opened for the network? Does it use STUN or something
else?
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Воскресенье, 22 июня 2014, 6:06 +04:00 от minus :
>
>
>OpenRTSP test sample always stops after 2 minutes. Why it happens and ho
OpenRTSP test sample always stops after 2 minutes. Why it happens and how to
prevent it?
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