In my case, the web camera is always generating a single file
(test.m4e), which continuously grows in size (w.r.t time).
I have tried to use testMPEG4VideoStreamer application on a fixed
size test.m4e file. And it works fine. But when this single file
(test.m4e) continuously grows in size, the t
Hi,
In my case, the web camera is always generating a single file (test.m4e), which
continuously grows in size (w.r.t time).
I have tried to use testMPEG4VideoStreamer application on a fixed size test.m4e
file. And it works fine. But when this single file (test.m4e) continuously
grows in size, t
My encoder produces 1 encoded frame every 1/15 of a second,
RTSPServer is reading faster and OS blocks when there is no data
available. How the increase of pipe buffer will help if encoder does
not produce more data?
To be honest, I'm not certain that increasing the pipe's buffering
will fix
Thanks for the note. Yes, the implementation of RTP-over-TCP (for
both clients and servers, and, in particular, its interaction with
RTSP) has been difficult, and in fact there were major changes
(improvements and bug fixes) made to the code back in July and August
of last year to support this
Ross,
Thank you for the quick response.
My encoder produces 1 encoded frame every 1/15 of a second, RTSPServer is
reading faster and OS blocks when there is no data available. How the
increase of pipe buffer will help if encoder does not produce more data?
Will implementing my own "FramedSource"
Does this this mean the live555 client is only meant to broadcast
the streaming media and not to send the source to another media
server to be streamed?
No, it means that we no longer support clients 'pushing' data to
servers at all. This is generally a bad idea, and there's no
standard prot
I am trying to make Live555 work with live video source. My H263
encoder writes encoded frames to Unix named pipe (FIFO) with a rate
of 15 fps. I created a RTSPServer instance, ServerMediaSession and
added a H263plusVideoFileServerMediaSubsession
(reuseFirstSource=True), passing the FIFO file n
Does this this mean the live555 client is only meant to broadcast the
streaming media and not to send the source to another media server to be
streamed? Is there a simple way to do that with the current provided
live555 library to mimic the darwin injection but to other media servers?
On Sun, Mar
I'm able to use the darwin test program to stream to a darwin media
server, but when trying to stream to a wowza server i with some of
the test programs it doesn't see to work.
Sorry, no idea. You're going to have to ask 'wowza'.
Which test program can i use to get a working audio and vide
Moderator
I'm able to use the darwin test program to stream to a darwin media server,
but when trying to stream to a wowza server i with some of the test programs
it doesn't see to work. Which test program can i use to get a working audio
and video stream. Also how would i change the test sampl
Hello,
I am trying to make Live555 work with live video source. My H263 encoder
writes encoded frames to Unix named pipe (FIFO) with a rate of 15 fps*. *I
created a RTSPServer instance, ServerMediaSession and added a
H263plusVideoFileServerMediaSubsession
(reuseFirstSource=True), passing the FIFO
Am I understood correctly: client session object deleted in
multicast mode, even one client from multi has a timeout
Yes, but note that the "RTSPClientSession" object is specific to that
particular client (and it's TCP RTSP connection to the server). If
that times out, then only that one
Mr. Ross,
Thank you very much for quick answer.
Am I understood correctly: client session object deleted in multicast mode,
even one client from multi has a timeout
ThanX again.
On Sun, Mar 6, 2011 at 12:33 PM, Ross Finlayson wrote:
> why multicast checking removed from livenessTimeoutTask A
why multicast checking removed from livenessTimeoutTask API???
Because there was no longer any need to exclude multicast sessions
from the normal client liveness checking.
I.e., as with all RTSP/RTP streams, the server will timeout the
stream if it doesn't receive any indication of 'liveness
Dear Sir!
I upgrade my old version of live555 on one latest (from 2011/Jan/24) and pay
attention on small difference in functionality in RTSP server that I don't
clearly understood. I mean this API:
void RTSPServer::RTSPClientSession
::livenessTimeoutTask(RTSPClientSession* clientSession);
As wri
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