On Aug 28, 2018, at 4:25 PM, Shyam Kaundinya
wrote:
>
> At times I see artifacts on the Rx side. Any thoughts on what may cause this
> behavior and how to fix it ?
Live555 is set up as a real-time streaming library primarily, not a YouTube
style buffer-and-play pseudo-streaming system. That
> On Aug 28, 2018, at 4:18 PM, jared-...@jaredrobinson.com wrote:
>
> Hello,
>
> I'm brand new to live555, so maybe this is a dumb question, but how can I
> gracefully disconnect all connected clients, without shutting down the server?
You call “GenericMediaServer::closeAllClientSessionsForSe
Hello,
I'm brand new to live555, so maybe this is a dumb question, but how can
I gracefully disconnect all connected clients, without shutting down the
server?
I want to do this when I change the password so that clients will get
disconnected and then they'll reconnect using the new password
> In the offical version of live555 this field is missing ( RequestRecord class
> ) probably because it is a specific onvif field.
>
> i think that a standard way to have this ( and other ) field could be the
> adding of a generic extra header in RequestRecord class.
>
> is it possible ?
No.
> 2/ Since I am sinking with UDP multicast, there is no concept of "a client
> establishing a connection. As such would I need to send out the prop-sets
> (VPS+SPS+PPS) before sending out every incoming frame or every i-frame or at
> some kind of interval as a background handler ?
I suggest se
A few more questions (not a repetition) on high bit rate streaming over a
wireless link ...
4/ I have replaced all occurrences of OutPacketBuffer::maxSize with a
large number as per the FAQ
OutPacketBuffer::maxSize = (4096 * 2160 * 3);
5/ Are there any other changes I should be making
Thank you for your response.
1/ Re: your response
Re: a
I just want to clear the air on the misunderstanding/miscommunication reg.
modification of supplied code. I did not modify the supplied source code. I
took testRTSPClient.cpp and made my own version of it, which follows the
example of func
> Oops - my mistake. It should have been instantiated in the
> “RawVideoRTPSource” constructor.
>
> I’ve just installed a new version (2018.08.28) of the code that fixes this.
And sure enough - I made a mistake there, so I’ve just installed a new new
version (2018.08.28a) that should fix it fo
> On Aug 28, 2018, at 8:44 AM, GENESTIER Denis
> wrote:
>
> OK, many thanks Ross.
> Currently, I am still testing your version and it seems to give the same
> results that mine.
>
>> Yes, because it is possible (in RFC 4175) for data from more than one line
>> to be contained within a singl
> On Aug 28, 2018, at 11:54 AM, Shyam Kaundinya
> wrote:
>
> The class diagrams in the live Documentation link on the live website
> http://www.live555.com/liveMedia/doxygen/html/ are missing
>
> I tried this link, not there either – the images don’t load.
> http://www.live555.com/liveMedi
> Re#2.
> a> In trying to implement the FAQ recommendation of using
> fmtp_spropvps(),sps,pps and then passing the values to
> parseSPropParameterSets, I tried to follow the code in
> H265VideoRTPSink::auxSDPLine and createNew functions and removed the parts of
> code that look for a fragmente
The class diagrams in the live Documentation link on the live website
http://www.live555.com/liveMedia/doxygen/html/ are missing
I tried this link, not there either - the images don't load.
http://www.live555.com/liveMedia/doxygen/html/classDelayQueueEntry.html
This used to work in the past if
OK, many thanks Ross.
Currently, I am still testing your version and it seems to give the same
results that mine.
> Yes, because it is possible (in RFC 4175) for data from more than one line to
> be contained within a single RTP payload, you need to subclass
> “BufferedPacket” to allow for this
Oops! Typo here…
#2 should read …
Do I need to also modify the
increaseSendBufferTo
and
sysctl net.core.wmem_max = the_maximum_buffer_size_that_you_need
as well to match the receivebuffer ?
From: Shyam Kaundinya
Date: Tuesday, August 28, 2018 at 8:07 AM
To: "live-devel@lists.live555.com
Hi
My product involves streaming high bit rate (>15M) 4K video over a high-bit
rate wireless link. I use LIVE555 proxy server on both sides. Based on
reviewing the forum articles, I have made the following changes, to accommodate
the high bit rate and an unreliable link.
1. I replaced all
Re#1. Yes. I use proxy to support additional clients which are RTSP.
Re#2.
a> In trying to implement the FAQ recommendation of using
fmtp_spropvps(),sps,pps and then passing the values to parseSPropParameterSets,
I tried to follow the code in H265VideoRTPSink::auxSDPLine and createNew
functio
> Can you please suggest how can we make the RTSP server to describe the audio
> stream information in the SDP.
Are you creating a “ServerMediaSubsession” object that describes the audio
(sub)stream, and then adding it to your “ServerMediaSession” object using
“ServerMediaSession::addSubsessi
Dear Ross,
How are you?
Its been long time I have posted any question on this forum.
We come across with a problem on our rtsp server when we start the RTSP server
to stream the video and audio from camera the audio stream information is
missing in the SDP
Somehow VLC player can able to stream
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