Webrtc is typically used in p2p applications. RTSP strikes me more as a
server/client relationship which would be better suited for websockets.
On Thu, Aug 29, 2019 at 9:06 AM Vishnu Mohan
wrote:
> Hi Ross,
>
>I was using HLS based transport streams for stream over HTTP, Some
> times with so
I've noticed that if I made specific modification to my NAL units before
passing them over to live555 (including totally randomizing the bits) the
NAL unit will not be sent at all to the client. Does live555 parse NAL
units prior to sending them out to clients? And if so how can I get live555
to st
I promise I won't feel entitled to any sort of encryption with RTSP. Like I
said I'll implement it manually with libsodium. That being said, isn't the
extra copy unavoidable? It seems that before any h264 NAL unit is passed to
RTP, it first needs to be encrypted and have some sort of authentication
My plan was to just add a layer ontop of live555 like libsodium to handle
my encryption. They also have options for handling dropped packets,
something UDP is very susceptible to.
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I am trying to encrypt my h264 packets sent with live555 using a pretty
basic stream cipher. One of the main requirements of a stream cipher is
that after a portion of a key is consumed to encrypt a bit of data that
portion can never be used again. So because of this the amount of data
being encryp
And I am also noticing now that netstat says that the rtsp port I am using
is in the following state "FIN_WAIT2". After I kill my client then it moves
to "TIME_WAIT". Which leads me to believe that the clean up code
listed in testProgs/testRTSPClient.cpp
does not sufficiently close the rtsp connect
I wanted my RTSP stream to be UDP based not TCP based to take advantage of
the latency advantage UDP has over TCP. However I noticed the following
when I took a look at netstat while my program was running:
~# netstat | grep 8554
tcp0 0 x.x.x.x:8554wsip-x-x-x-x:39224 ESTABLISHED
My particular use case has my program started and stopped rapidly in short
periods of time. But I can't start and stop it like this if I can't quickly
re-bind to the rtsp port. Maybe this doesn't have anything to do with my
clean up code? Is it related to this:
https://live-devel.live555.narkive.co
Hi all, in the code for testProgs/testOnDemandRTSPServer.cpp there is no
code specified that should be used to clean up the RTSP server
cleanly. testProgs/testRTSPClient.cpp actually does specify some code to
clean up the program RTSP client.
I have tried adding "Medium::close" and an "eventLoopWa
I don't call "getNextFrame()" or "doGetNextFrame()" anywhere in my code...
where is it being called from?
On Thu, Feb 7, 2019 at 2:45 PM Ross Finlayson wrote:
> In LIVE555 applications, a “FramedSource” (subclass) object can be read
> from (using “getNextFrame()”) only once at a time. In other
Okay, now is there any correlation between "doGetNextFrame()" being called
by whatever process calls it and "isCurrentlyAwaitingData()" being set to
true?
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What does this function do? I am not seeing any documentation on this.
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Hi! I have set up an RTSP server based on this code:
https://github.com/RafaelPalomar/H264LiveStreamer. The program however
jumps up to 100% CPU usage every time any client connects. Why might this
be? I have even tried commenting out the x264 encode part (and I instead
pass in empty nal units) to
I also have discovered that this delay occurs when I stream from VLC. Which
means this is likely not an issue my client is causing but rather something
on the RTSP server end what could it be?
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h
Okay I have just incorporated a jitter buffer and It is working but there
is still a very strange delay occasionally in the stream. It will buffer
for a whole second before displaying any frame on the screen. I'm a bit
lost now as to what could be causing this issue. Any advice?
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Firstly, I apologize for not cutting off previous replies in this email
chain before replying.
Also with the packets that come in with the identical presentation time
(down to the millisecond) should I wait and collect them all together
before passing them into the decoder or should I pass them in
time that the frame should be rendered. Seems impossible
On Thu, Jan 3, 2019 at 3:39 PM Ross Finlayson wrote:
>
>
> > On Jan 4, 2019, at 12:29 PM, Kevin Bailey
> wrote:
> >
> > What exactly does the presentation time entail? Is it actually the time
> in which a fra
disappear after any period of still frames.
On Thu, Jan 3, 2019 at 3:22 PM Ross Finlayson wrote:
>
>
> > On Jan 4, 2019, at 12:07 PM, Kevin Bailey
> wrote:
> >
> > Hi all! Can anyone point me to an example of how to implement a jitter
> buffer into an rtsp clien
Hi all! Can anyone point me to an example of how to implement a jitter
buffer into an rtsp client using live555? The example that is included with
live555 doesn't have any sort of jitter buffer and the stream frequently
freezes. Thank you!
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I am fairly confident that I am slicing my NAL units, but is there any
chance you might be able to link me to an example of how to appropriately
slice NAL units with x264? Figured you might know where an example might be
considering you've dealt with that kind of issue on many occasions.
I will lo
Thank you again for the support. I have split up my h264 key frames into
smaller NAL slices and I am still seeing artifacts on the screen. It is
also important to note that my h264 key frame sizes were already smaller
than 100,000 bytes each (closer to 20,000 bytes each).
Could this be the cause o
> wrote:
> >
> > 1/ In your H.264 encoder - i.e., between the creation of a raw video
> frame, and the time that it is encoded into (one or more) H.264 NAL units.
>
> Ross didn’t go into any detail on this point, Kevin, so I thought I’d
> weigh in on it.
>
> I thin
Hi all! I am not quite understanding why my code has so much latency... My
main use case is not for gaming, but I did try to play a game over the
stream and the latency is HORRIBLE. I am not quite understanding what the
cause could be...
Here are some links to my code on pastebin:
https://pastebi
blem when the server
started again?
Thank you so much.
Kevin
--
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.
.
.
.
.
.
.
--
On Thu, May 7, 2015 at 8:30 AM, Kevin Chen wrote:
> Hi Ross,
>
> After checking al
Hi Ross,
After checking all my code carefully, it turned out to be my fault, I
create rtcpGroupAudiosock using the rtp port for audio instead of rtcp
port, which fix the problem, and both audio and video can stream together,
and sync well, but one problem still there when using VLC, two time stamp
video_ts.tv_sec += uSeconds/100;
encoder_video_ts.tv_usec = uSeconds%100;
}
fPresentationTime = encoder_video_ts.tv_sec;
--
And for the AAC audio, for each frame data, I set "fPresetationTime" from
"gettimeofday".
since audio and video are feeding using two sepa
Ross,
Thank you very much for this. Exactly the information I required. I have
written a SAP implementation and this is now working well for my needs.
Thank you for your help
Kevin
On 7 February 2014 02:52, Ross Finlayson wrote:
> I am writing an application which I would like to strea
tp without the control protocol?
Thank you in advance for any help
Kevin
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I was wondering what was the status of SRTP support?
And what, if anything, I could do to help?
Kevin Anthony
Software Engineer
Activu Corporation
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Hi,
First of all, thanks for the excellent work on this project. I found it has so
much for building multimedia streaming solutions.
I developed some code with Live555 to receive mpeg ts stream over raw UDP and
send it out wrapped in RTP. It works fine for me. Now, I need to receive UDP
packet
How can I feed my "H264VideoStreamFramer" to a "H264VideoRTPSink", to stream
the H.264 stream via RTP ?Can anybody give me a example? thanks.
--
Kevin
2009-06-19
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.
--
Kevin
2009-06-18
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"generic" payload ?
Am I missing something anywhere ?
Thanks a lot,
Kevin.
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Ross Finlayson a écrit :
At 07:52 AM 6/5/2008, you wrote:
MP3 File -> ByteStreamFilesource -> SimpleRTPSink -> rtp
stream->SimpleRTPSource->BufferSink-> myRingBuffer
But now, I wonder if there is a simple way to get RTP packets in this
buffer.
Yes - At the receiving end, use
achieve this.
Regards,
Kevin.
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trying to do this, by modifying the FileSink class (to create an
BuffSink) and the DeviceSource class (to create my BuffSource). I'll let
you know if I encounter some problems :)
Thank you for your help and your time,
Kevin.
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quot;BufferSink" and "BufferSource" that I haven't figured out yet ?
Any Idea ?
Regards,
Kevin.
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