To get proper audio/video synchronization, you must create a
"RTCPInstance" for each "RTPSink". However, the
"OneDemandServerMediaSubsession" class does this automatically, so
because you're subclassing this, you don't need to do anything
special to implement RTCP - you already have it.
Howe
Hi all,
I'm new to the live555 library.
I have a MediaServerSession with two SubSessions (one for H263 video and the
other for G.711 A-law audio), both SubSessions extend from
OnDemandServerMediaSubsession, the one I use for the video is called
CH263plusVideoDXServerMediaSubsession and the
On Tue, Sep 02, 2008 at 01:43:54PM -0700, Ross Finlayson wrote:
We don't make available old versions of the code, and offer absolutely no
support for old versions. Everyone should work with the newest version
of the code.
Ok Ross, one more question... Can openRTSP cope with a variable
frame
On Tue, Sep 02, 2008 at 01:43:54PM -0700, Ross Finlayson wrote:
> We don't make available old versions of the code, and offer absolutely no
> support for old versions. Everyone should work with the newest version
> of the code.
Ok Ross, one more question... Can openRTSP cope with a variable
fra
The used sink has a big enough buffer, so if there is some data loss,
that appear before the sink. I saw the following lines in
MultiFramedRTPSource:
// Try to use a big receive buffer for RTP:
increaseReceiveBufferTo(env, RTPgs->socketNum(), 50*1024);
is it possible that my problem caused by t
The used sink has a big enough buffer, so if there is some data loss,
that appear before the sink. I saw the following lines in
MultiFramedRTPSource:
// Try to use a big receive buffer for RTP:
increaseReceiveBufferTo(env, RTPgs->socketNum(), 50*1024);
is it possible that my problem caused by th
