I'm seeing the same problem -- and struggling to find an explanation. My
set up is:
1. live555MediaServer serving an mkv file in Docker container
2. live555-based RTSP client reading the stream in another Docker container
on the same device
3. TCP is being used for RTP transport. Because of
this,
mail/live-devel/2005-November/003543.html
Thanks for your help!
On 5/17/22 8:14 PM, Ross Finlayson wrote:
On May 17, 2022, at 3:33 PM, Alex Agranovsky wrote:
Server-side: using stock live555MediaServer running on macOS.
Client-side:
- The problem occurs regardless of transport (both TCP
sure what could cause it -- still looking.
On 5/17/22 1:54 PM, Ross Finlayson wrote:
On May 17, 2022, at 11:36 AM, Alex Agranovsky wrote:
Hi,
I'm debugging a situation, where the first frame on the connection is always
dropped by live555 RTSP client.
Can you say more about your con
on how to set this up?
Thanks,
Alex Anderson
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Hello,
Trying to run live555MediaServer with a 1080p mkv input, and seeing this sort
of errors:
MultiFramedRTPSink::afterGettingFrame1(): The input frame data was too large
for our buffer size (301380). 439528 bytes of trailing data was dropped!
Correct this by increasing "OutPacketBuffer::m
I'm sorry. I don't think I was very clear about how I'm sharing the
Groupsocks, so I think there may be some confusion.
I'm not sharing a Groupsock between RTP and RTCP. Instead, I have one
Groupsock shared between the RTPSink of the multicast out and the RTPSource
of the multicast in, and that Gr
Hello,
I have implemented a multicast out component using SimpleRTPSink and
RTCPInstance, and a multicast in component using SimpleRTPSource and
RTCPInstance. When the multicast in component is receiving from a multicast
out component on the same device, I have them share Groupsocks for RTP and
RT
Hello,
In each of the the test*Receiver apps, they utilize RTCPInstance and pass
it an estimatedSessionBandwidth of 160. I realize this parameter doesn't
have to be accurate, but it should be roughly the bitrate of the stream,
right? The test*Streamer apps use an estimatedSessionBandwidth of 5000,
n it arrives. I could not figure out how to start the Framer idle and
provide one frame on request...
Thanks,
Alex
From: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson
Sent: Wednesday, August 21, 2013 3:28 AM
To: LIVE555 Streaming Medi
ce,
addNewDataSource)
5. In my DataStreamFramer, I implemented a parser which extracts
a timing info from the data and aligns it with the "wall clock".
What am I doing wrong?
Thanks,
Alex
*
Sorry about that J
When I run the testOnDemandRTSPServer and run the play RTSP url of the H.264
video in vlc 2.0.5 it works just fine (unsurprisingly). But for my
application I need to use RTP, not RTSP.
What do I need to do in order for the live to stream RTP?
From: live-devel-boun...@ns.li
Thanks for the quick answer.
The problem with this construction is that even if I run the
testH264RTSPVideoAudioStreamer, I can play the stream (using the URL
"rtsp://.") in VLC 1.11 (and 0.8.6 for that matter) but not in VLC 2.0.5.
What could be the problem, and how can I fix that?
Thanks,
Hi,
I'm streaming an H264 movie using live555. I figured out you need an SDP
file in order to play the stream in VLC, so I built one and was able to play
my stream. The problem is, after I upgraded my VLC to 2.0.5, the VLC won't
play my stream.
The SDP file I use is:
v=0
o=- 127764715
ed as RECORD_PIC_IFRAME
from... to... for normal
working trick-mode.
Thanks,
Alex.
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Thanks a lot.
I've changed source with next:
if (username != NULL && password != NULL) {
result = rtspClient->describeWithPassword(url, username, password,
False, 10);
} else {
result = rtspClient->describeURL(url, NULL, False, 10);
}
And now I have 1:24 before exiting instead of 2
Hello.
Having a look to the software openRTSP it's amazing.
But I have one question - is it possible to decrease timeout before exiting
if host specified in url is unreachable ?
E.g. I use openRTSP in my script and have to rotate the output files that's
why I start it again and again and when
Hi.
Does the library support rtp streaming for jpeg JFIF images?
When I stream normal jpeg images it's all ok (tried with different header
lengths, different qfactor and different types) , but if I stream JFIF ones my
player can't decode the received images.
th
Hi.
My program has a main loop in which other libraries call their own select().
In order to make this call non-blocking, for each of these selects I use a
timeout of 0 seconds, and I make a very short pause (about 3 milliseconds)
beetween each loop cycle so to avoid a CPU heavy load.
In additi
Hi
1) Suppose that I set OutputPacketBuffer::maxSize to a large value.
Now, I wonder:
is fDurationInMicroseconds, inside doGetNextFrame() of a MySource, ignored
until the buffer is filled (so it has no effect if I set it to a value
different than 0) ?
-
thanks.
I have just tried your solution, calling
videoSink->transmissionStatsDB().numReceivers() but:
1) the count is incremented only when the stream is viewed from a receiver on a
different host than the streamer's one.
2) the count is not decremented when the stream is not viewed anymore fro
Hi,
is there a way to count how many clients are currently viewing a multicast
video stream ?
If so, which classes/functions should I consider to add on a test program like
testXYZVideoStreamer.cpp ?
Thanks
Alex
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Hi,
Is there anyone who managed to run it on a non daVinci processor?
I'm having many problems compiling the package and use it with the NDK.
I'll Be glad to get some help/info.
Thanks,
Alex
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i get some tips for getting stared
Thanks,
Alex
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, see here:
http://msdn.microsoft.com/en-us/library/ms740141(VS.85).aspx
"Any two of the parameters, readfds, writefds, or exceptfds, can be
given as null. At least one must be non-null, and any non-null
descriptor set must contain at least one handle to a socket."
Alex
On Tue, Nov
patch for MPEG4VideoStreamDiscreteFramer
which allows the user to retrieve the keyframe status of the last
processed frame.
These patches are against live.2008.11.13.tar.gz.
Regards,
Alex
PollingBasedTaskScheduler.patch
Description: Binary data
MPEG4VideoStreamDiscreteFramer_keyframe_flags.patch
Description: B
ot; which can be used to disable
"Medium::close(fInputSource);" in the above before the cleanup
sequence. Is this an appropriate solution?
Thanks,
Alex
FramedFilter_detachInputSource.patch
Description: Binary data
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l
tag is equally
useful for compiling as a DLL under GCC/Cygwin on Win32.
This patch is against live.2008.11.13.tar.gz.
Regards,
Alex
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hi,
I stream a mp4(h264+aac) file from a Darwin Streaming Server v5.5.3
and write a livemedia test client to test it.
When I do seeking operation, I find that when the client send pause,
seek and play command and start playing again, there will be a gap in
the DSS's RTCP SR ntpTimestamp(I print out
es per second say) rather than continuously? Should the delay
function go in the startCapture() function or the doGetNextFrame() or is
there a variable I need to set when adding the source to the environment
taskscheduler?
Thanks,
Alex
Alex Tarter
Ultra Electronics
Sonar & Co
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