Hi Mr. Finlayson,
I forgot to update the bug-report.txt and did not test the container
fully That is a mistake on my side. Sending a fixed zip archive.
For the bug, it is a bit difficult to reproduce - that is true.
Best regards,
Martin Mirchev
Hello Mr. Finlayson,
We still observe the stack-use-after-return bug in live.2023.06.14 while
running in Ubuntu:20.04.
You can reproduce this bug as the README in the attachment as follows:
1. build the docker image:
docker build . -t suaf
2. create the docker container:
docker run -it
ile in the experiments folder. The first instance should display the
error that has been discussed.
The ASAN options are a little less verbose but you can modify it to make
it fully verbose.
Yours sincerely,
Martin Mirchev
___
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Hello Mr. Flayson,
Yeah... After reading it a little more, I saw that everything is okay
with the code... Sorry for the false alarm...
Best regards,
Martin Mirchev
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http://lists.live555.com
Thanks for your help. It seems in fact related to our network setup.
#
" This e-mail and any attached documents may contain confidential or
proprietary information. If you are not the intended recipient, you are
notified that any dissemination, copying of this e-mail and any attachments
thereto
VLC client, the first VLC client showed a fluent
picture again
4. stopped the first VLC client
Attached are the logs generated by live555ProxyServer (IPs and credentials were
replaced).
Does somebody have an idea, how I can resolve this problem or what I am doing
wrong here?
Thanks for
something back.
Regards,
Martin.
From: live-devel [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Jeff
Shanab
Sent: 01 December 2015 01:46
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] RTSP Server Packet Loss
I concour.
Here are some data points.
I ha
ikely
that if you need to increase that value to deal with larger frames you will
also need to increase the socket send buffer size.
Regards,
Martin.
-Original Message-
Date: Sat, 28 Nov 2015 05:23:32 +1300
From: Ross Finlayson
To: LIVE555 Streaming Media - development
any reason it's not a good idea to make the sending socket blocking. The
alternative would be to keep the socket in non-blocking mode but make sure that
the packet is re-queued for sending upon receiving the WSAEWOULDBLOCK error,
that seems like a more involved change though.
Regards,
M
s and can ignore presence of additional attributes would seem to be
preferable.
Let me know if such a modification would be acceptable.
Thanks, Martin
[http://transfer.icomedias.com/icomedias-email-banner/icomedias-sharepoint2013.png]<http://go.icomedias.com/1000.0260/bild>icomedias
ist Mic
Hi,
I’m running into issues trying to receive streams from IP Cameras requiring
digest authentication. It looks like the current implementation of digest
authentication is based on RFC 2069 which was obsoleted 2617 in June 1999.
Here’s an example authentication header returned by one of my camer
n the Media::close() method. Is there a simple
solution?
Martin
From: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson
Sent: 04 May 2012 21:19
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] raw PCM from TCP sock
are being lost in a sequence of tests.
I haven't done anything with the TaskScheduler code. I have changed
MediaSink.cpp to set OutPacketBuffer::maxSize appropriately for the amount of
data I am sending.
Thanks
Martin
From: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.
le of how DeviceSource needs to be modified to
achieve this.
Thanks (I realise this is a big ask)
Martin
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g the file with http via the web server is fine
but using rtsp via the 8554 port causes it to appear to play but be completely
silent.
Has anyone else had this problem or could hint at where to look in this
particular source code?
Thanks, Martin
___
Hi,
I have a problem with the recorded video file duration.
I use openRTSP with the "-d" parameter.
For exemple when I set "-d 30" my video file duration is around 53 seconds.
Does the stream quality can influence my record duration ? Beacause when I look
for the video recorded it seems to lag.
Hi,
I would like to use OpenRTSP to record an IP camera stream on my PC.
I arrived to connect it by using this kind of command : openrtsp.exe -4
rtsp://192.168.0.200/rtsph264
but I don't know how convert the stream received in a video file ?
Someone to help me ?
Cheers
Boris Saint-Martin___
Hello,
Beginner in openRTSP usage, I have many questions :D
I'm trying to capture a RTSP stream from my internet modem.
The URL is
"rtsp://mafreebox.freebox.fr/fbxtv_pub/stream?namespace=1&service=202&flavour=hd".
I can view the stream using VLC but not with openRTSP.
Here is the log :
Microsof
Hello,
I would like to record IP camera streams by using openRTSP.
My problem is to compile the source with Visual Studio 2008.
I have built the makefile but I can't load it in VS. It trying to convert the
project but it failed...
Is someone can help me to build it or simply give me the windows
patch allocates one byte more, like fResponseBuffer does, so
that it is safe to call getResponse1().
Best Regards,
--
Martin
--- live_old/liveMedia/RTSPClient.cpp 2010-02-09 10:35:28.0 +0100
+++ live_new/liveMedia/RTSPClient.cpp 2010-02-09 16:48:52.0 +0100
@@ -982,7 +982,9 @@
e historic versions mentioned above? I would appreciate any help.
Sincerely
Martin Stellmacher
Elektronik/Infotainment/Produkte, EI-P 2
Electronics/Infotainment/Products
IAV GmbH
Carnotstrasse 1
10587 BERLIN
GERMANY
Phone: +49 30 39978-9472
Internet: http://www.iav.de
IAV GmbH, Sitz/Registe
, see the attached patch. This seems to work
fine for me.
Regards,
// Martin Storsjödiff -ur live-orig/liveMedia/RTSPClient.cpp live/liveMedia/RTSPClient.cpp
--- live-orig/liveMedia/RTSPClient.cpp 2009-02-23 12:03:54.0 +0200
+++ live/liveMedia/RTSPClient.cpp 2009-03-11 21:29:31.0
. And also
that RTSP is a symmetric protocol (compared to asymmetric HTTP).
So I guess we have to deal with these situations. Attached is a
patch that tries to fix this.
Best Regards,
---
Martin
diff -u -r ../live.old/liveMedia/RTSPClient.cpp ./liveMedia/RTSPClient.cpp
--- ../live.old
play the video when streaming, but not when play locally?
2) The received "video-MP4V-ES-1" is different from the original "para.m4e", at
least at the beginning section of the stream. Why? How do I know if the
received stream is intact?
Thanks,
-martin
__
Hi all,
I am trying to send a H.264 HD vídeo in two diferent networks using the same
computer with two network cards. It works perfectly when I am streaming it at
6 Mbps using one network card. But, when I replicate the same or other similar
video into the other network, the bitrate slow down to 4
Hi all,
I stream a mp3 file using MP3FileSource and MPEG1or2AudioRTPSink? It works
fine but is it correct? Should I use another sink class? What type of session
class should I use for an on demand service?
Thanks in advance,
Ramon
___
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integrate
stream duplication functionality with software already producing
H263-1996 stream, so I just need liveMedia to be able to receive such
stream, not to send it out.
Sorry if i didn't make that clear in my previous post.
Best regards
Martin L
, or if
anyone have some starter points?
Best regards
Martin Lindhe
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Hi all,
I am using testMP3Streamer to send "test.mp3", and I can listen it opening SDP
file with VLC, but with QuickTime it doesn't work. Although time starts to
advance, I can't listen anything. Is it normal? Has QT some other parameter
requirements in SDP file?
Thanks in advance,
Ramon
___
r more
information about that issue.
http://www.faqs.org/rfcs/rfc2190.html
http://www.faqs.org/rfcs/rfc4629.html
http://www.faqs.org/rfcs/rfc4628.html
Best regards
Martin Lindhe
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Hi all,
I am trying to send a HD H.264 video and I found a problem when the frame size
is bigger than unsigned range (0-65535) and I try to copy it to fTo pointer.
First I thougt to change frameSize from unsigned to long, but error still
apeared. I think it's not only a frameSize problem, so maxim
Hi,
I saw that RTSPServer implement digest authentication control on RTSP
Protocol, but does it work with basic authentication ("username:password@")?
How does boolean PasswordIsMd5 work? Does that flag permit RTSPServer to
choose digest or basic authentication?
If it's not this case, what could
se, without
> the others noticing, hence, without stopping the event loop.
>
> Thanks for your suggestion though,
>
> Bob
>
>
> On Mon, 2007-09-24 at 12:47 +0200, Ramon Martin de Pozuelo Genis wrote:
>> Hi Bob,
>> you may use a watchVariable in the doEven
Hi Bob,
you may use a watchVariable in the doEventLoop like this
...
watchVariable=0;
env->taskScheduler().doEventLoop(&watchVariable);
...
and add a socket or a new thread that permits you to change this variable
externally. When the Scheduler watch this variable is modified
Hi all,
in some moment of my application I want to close a RTSPServer, but I note that
when I do a Medium::close(rtspServer) I don't close the RTSPClientSessions
that handle the incoming commands. Then, when Server is stopped and some
incoming command arrived it broke down my application. Is there
Well, I saw that closing completely QuickTime (not only video window) and
trying to reconnect it reclaims again user and password so I supose that it
remains in Quicktime memory and it's not possible to control on server side,
because it is working properly, receiving user and password again. Could
Hi,
I use UserAuthentication option in RTSP making use of
UserAuthenticationDatabase and QuickTime as client. First time client wants to
connect to a streaming session QuickTime asks for user and password to
complete authentication with server. But if client closes it and then tries to
reconnect, i
> Again, it sounds like you're trying to reinvent the wheel. The
> "OnDemandServerMediaSubsession" class works just fine - you should
> just use it (by defining your own subclass). Note the several
> examples in the code. You should be using "testOnDemandRTSPServer" -
> not "testMPEG4VideoStream
Sorry, I forgot ask you about SDP file. Now address is correct (0.0.0.0) but
what about port? "m" line of SDP is still using port I specified when I create
groupsock, could I change it? Should I change it? If the first description is
sending broadcast in te same port it will cause a problem.
Thank
> You probably shouldn't be using "addDestination()" - that is a
> specialized function used only to implement on-demend unicast
> streaming to multiple clients from a single source. (Note that, for
> unicast on-demand streams, the SDP description should contain the
> special address 0.0.0.0, not
Hi all,
I created a session with its respective rtpGroupsock & rtcpGroupsock and then
I added another destination using:
rtpPort= new Port(newPort);
rtcpPort= new Port(newPort+1);
destinationAddress.s_addr = inet_addr((char*)newIP);
rtpGroupsock->addDestination(destinationAddress,*rtpPort);
rtcpGr
Hi,
I saw there are implemented classes to stream MPEG4 Audio based on RFC3016
(MPEG4LATMAudioRTPSink, MPEG4LATMAudioRTPSource). What kind of session may I
use to create an on-demand service of this audio? Using this classes could I
stream a MPEG4 elementary audio stream based on RFC3640?
Thanks i
>However, if your intention is to stop streaming to dead clients, then
>remember the "reclamationTestSeconds" parameter to
>"RTSPServer::createNew()" - see "liveMedia/include/RTSPServer.hh".
Thankyou very much Ross, but it is not exactly what I need. I am offering some
services from my Web Service
Hi all,
Could I close an on demand session from server side? How will it be the best
way to do that?
I tried "rtspServer->removeServerMediaSession(sms)" and then, this subsession
is not accessible for future RTSP requests, but a client that is playing this
session continues playing the video. I wo
Hi!!
What would I do to add a extension RTP header?? I saw there are some functions
in MultiFramedRTPSink that seems to do this (setSpecialHeaderBytes /
setSpecialHeaderBytes) ? Is it OK? In that case, exactly where and when have I
to add a call to this functions for a correct implementation?
Tha
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