Hello,
This may just be me missing something small / being stupid, but..:
when I connect to an RTSP camera stream like so:
openRTSP rtsp://USER:PASSWORD@192.168.1.20:5554/live/ch1
part of the output I see is:
--
Received 170 new bytes of response data.
Received a comp
Hi,
I modified testRTSPClient to only process ONVIF metadata from a "smart" camera.
In DummySink::afterGettingFrame there's a line:
if (strcmp(fSubsession.mediumName(), "application") == 0) {
...
after which it processes the metadata.
This runs for a short while, after which recvfrom in
Gro
Hi !I'm just starting to understand the library, help with advice.I get data from the IP CCTV camera on a private video protocol. Data from the camera comes in fragments in real time.In the SDK camera, I register a callback function in which I process the data. After processing this data in the buf
> You could check this for sure by running ?testRTSPClient? (rather than
> ?openRTSP?), because ?testRTSPClient? outputs a line for every RTP packet
> that it receives.
Thanks, that's useful: indeed only seeing video/H264 packets arrive.
Do you know of a public test camera that produces a met
)
thanks again,
Patrick
--
Message: 1
Date: Fri, 1 Nov 2019 10:32:18 +0100
From: "P. Min"
To: live-de...@us.live555.com
Subject: Re: [Live-devel] openRTSP, how to capture ONVIF metada stream
co
> > This runs for a short while, after which recvfrom in
> GroupsockHelper::readSocket returns error 104.
>
> What does that error number mean on your system? (In other words, what
> does it show for error number 104 when you run ?man errno??)
"Connection reset by peer"
(perror() sho
Another follow-up: it looks like the source stream is flaky.
Other clients (like ffmpeg, and yellow.js running under node) also stop after a
while.
It would be great if there was a way to somehow reconnect to the stream when
this happens
(I suppose I could just restart the program)
_
Hi,
In the downloaded code the doEventLoop() was not exiting and it was a
infinite loop. Now i changed the code
so that it will exit on keyboard hit. Then also it is not exiting properly.I
think it is
because of some memory leaks. Can anyone give an idea about the resources to
be freed while exitin
Hi Ross,
In the BasicTaskScheduler::doEventLoop I have added code for exiting on
keyboard hit. Now if I am hitting keyboard, it will exit from the loop ,but
after that it hangs.I think it may be due to some memory leaks. So if you
can help me on the resources to be deleted before forcefully exi
Hello Ross,
I and Saurabh are working on same project.
** **
We tried analyzing into what is happening by using the testRTSPClient as
suggested by you and here is the verbose dump of the testRTSPClient:
** **
Opening connection to 10.17.1.111, port 8554...
...remote connection op
Dear Ross,
Thanks for your response. I have dump debug info which you have suggested.
Please find log below.
LOG
INFO-START**
*LIVE555 Proxy Server*
Dear Ross,
Thanks for your quick support. It is working now.
It also comes to us notice that same type,same setting different camera
has different metadata.
Regards
Kiran
On Fri, Jul 6, 2012 at 6:48 PM, Ross Finlayson wrote:
> OK, I figured out the problem - it was caused by the nonstandard
Hello,
Currently I'm working on improving an RTSP/RTCP/RTP server based on Live
library which transmits the live H264 stream in HD resolution.
Trying to lower the CPU consumption by the server (the hot-spot was
MultiFramedRTPSink::sendNext callback), I increased the maxPacketSize
value (by callin
Thanks for your quick and informative reply!
Yes, the LAN - it's exactly my case.
I see that the new version of the library is delivered.
I'm going to use it and to send back my results of a performance testing.
Regards,
Max
On 07/23/2015 09:39 PM, Ross Finlayson wrote:
> Increasing the maximum
I've added ability to optionally change the packet size with the help of
an environment variable, the executable is always the same.
(The technical details comes after the stat.)
Unfortunately, my application is quite complex, multi-threaded, contains
many layers (video4linux, DSP h264 encoder, ..
Hello,
We need to broadcast the PCM (S16LE/16000) audio as a separate channel
with our RTSP-server.
I've found this letter for the AAC format:
http://lists.live555.com/pipermail/live-devel/2014-January/017980.html
What is a preferred RTPSink class that I should use for this audio
format in
Thank you very much, it has worked out.
[rtsp @ 0x7f35f8c0] SDP:aq=0KB vq=0KB sq=0B f=0/0
v=0
o=- 137163047 1 IN IP4 0.0.0.0
s=Live session, streamed by LIVE555
i=live?token=1080p_noaudio
t=0 0
a=tool:LIVE555 Streaming Media v2015.07.23
a=type:broadcast
a=control:*
a=range:npt=0-
Hi,
Sometimes, under some condition I need to reply to the client of my RTSP
server - "403 Forbidden".
As a temporary solution, I can override (and I do) "Boolean
RTSPServer::specialClientAccessCheck(...)" but it gives me "401
Unauthorized" which is not exactly the case, see this link for details
Thank you very much, Ross!
I think it is possible to do a patch which will not break the old
behaviour and let other replies.
(And I personally have no doubt that you know better than me how to do
it; just as a proposal...)
Say, in the header we add another virtual method with the string as a
ret
ks, Inc.
> http://www.live555.com/
>
>
> ___
> live-devel mailing list
> live-devel@lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
http://lists.live555.com/pipermail/live-devel/2015-December/019794.html
[Live-devel] RTSP response status code
Hi,
A cycle with ServerMediaSubsessionIterator is used in
RTSPServer::RTSPClientSession::handleCmd_SETUP() to calculate the value
of fNumStreamStates.
I wonder why ServerMediaSession::numSubsessions() isn't used for it?
Could they have different results?
Regards,
Max
___
On 06/29/2016 11:29 PM, Ross Finlayson wrote:
> yes - if a new “ServerMediaSubsession” object happened to be added to the
> “ServerMediaSession”
So, if a new subsession is added to the session, it should be done with
the help of ServerMediaSession::addSubsession(subsession) method which
increases
on windows implementation ofgettimeofday()
Quoting Ross Finlayson :
> Unfortunately I'm not an expert on Windoze-specific API stuff.
You may have a look at vlc times functions :
http://git.videolan.org/gitweb.cgi?p=vlc.git;a=blob;f=src/misc/mtime.c;h=0dbb4df578308b38e6e3ff9487b0e9143f11853b;
Example with a x64 OS:
VS2010:
c:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\
VS2008:
c:\Program Files (x86)\Microsoft Visual Studio 9.0\VC\
I btw use the short path (that you can see with dir /X) to avoid issue with
space:
c:\PROGRA~2\MICROS~2.0\VC
/C
From: live-devel-boun...@ns.li
After doing some testing with openRTSP and looking through the code it appears
like "Absolute Time" is currently not supported for RTSP streams. I.e. I can't
specify a specific time that the stream should seek to, e.g. Range:
clock=20120629T07.00Z, all according to paragraph 3.7 at
http://w
Is there a way to get the recording timestamp for stored content?
When looking at the presentation time that is received with each frame I see
that it is matching the current time of the server, but when playing back an
archived stream I'm interested in when it was recorded. Can that be found
some
I find your question confusing, because it wasn't really clear how it
relates to our software. Could you please clarify your question, explaining in
particular how you are using our software? (Remember that our software can be
used to build RTSP clients, RTSP servers, RTSP proxy server
When implementing liveMedia using multiple streams in one process I see two
choices:
1. Each stream is kept totally separate. I.e. each stream have their own
TaskScheduler, UsageEnvironment, eventLoopWatchVariable and each
doEventLoop() is running in a separate thread.
2. The rtspClient's share
I'm writing a playback client based on the testRTSPClient example. For each
new stream to connect to, I setup a new environment, i.e. create new
UsageEnvironment, StreamClientState etc.
Everything works great and I can start and stop streams multiple times
without seeing any issues. There is one e
FYI, the latest version (2012.08.20) of the "LIVE555 Streaming Media" code
adds optional RTSP server and RTSP client support for streams that are
indexed by 'absolute' time - i.e., using strings of the form
"MMDDTHHMMSSZ" or "MMDDTHHMMSS.Z".
Ross Finlayson
Ross, Thank you very muc
- The absolute timestamp I ask for doesn't match what I receive, Many hours
off, but at least it's consistent.
For what it's worth, this sounds like it could potentially be a time zone
issue. I've run into issues in the past where some of the routines I used
interpreted my UTC timestamp as
Best possible uptime is essential for the RTSP client I'm implementing. I've
therefore looked into how to best handle reconnection if the stream for any
reason is disconnected. I noticed fairly quickly that liveMedia is taking
care of that really good and there is no reason for me to try to impleme
There is one case where it doesn't work though, and I'm not sure how to
handle it. This is if I do a seek while the stream is disconnected, then it
never reconnects. In some cases I play a 10s loop where a timer do a seek
every 10s and jumps back (using absolute seeking). Those streams never
reconn
Thanks for the report. I have now installed a new version (2012.09.07) that
should fix the problem (with receiving RTP-over-TCP on Windows).
Thanks a lot!
(FYI: The bug got accidentally introduced back in version 2012.07.24, when we
added a fix to better handle the disconnection of the remote
Right now Live555 loop run in separate thread. The clean up function can be
called from the another thread
No! You cannot safely do this! Read the FAQ (that you were asked to do
before posting to this mailing list)!
Apart from "triggerEvent()" (see below), all calls to LIVE555 code *must*
The code example below is called from an external thread. Is that ok
No - absolutely not!!! What you're trying to do - call
"TaskScheduler::scheduleDelayedTask()" from an external thread - is
extremely wrong!
Look folks, how many times do I have to say this: A LIVE555 application runs
as
One last question on this, as I'm in the middle of reorganizing the code.
Shouldn't it be safe to create the rtspClient object in another thread?
Once again, NO! "RTSPClient" objects - like all subclasses of class
"Medium" - update shared data structures (stored within the
"UsageEnvironment"
There is one case where it doesn't work though, and I'm not sure how to
handle it. This is if I do a seek while the stream is disconnected, then it
never reconnects. In some cases I play a 10s loop where a timer do a seek
every 10s and jumps back (using absolute seeking). Those streams never
reconn
This is the first time you've mentioned an exception. (Previously, you said
just that the stream failed to reconnect afterwards.)
I've attached a modified version of our "testRTSPClient.cpp" demo
application. It starts playing the specified stream - as usual - but then,
after 60 seconds, send
When connecting to a server that doesn't respond, or is just slow, the
timeout appears to be around 25s, i.e. the time before the response handler
is called.
Is there a way to decrease this time?
Why? Well, in the project I'm working on there are lots of changing cameras
that are on a 10s c
Lately I've seen some random exceptions when streams are being shut down. I
believe these were introduced when I changed the code so it waits for a
TEARDOWN response. This was btw done as Cisco mentioned their server
requires that.
I'm probably doing something wrong, as it appears the error happen
Cisco has confirmed a bug in their VSM-server (v6.3.2) when using RTSP. To
get around the bug we've been told to "disabling the use of RTCP when they
setup the playback sessions. This is done by only specifying one client port
(the RTP port) on the SETUP request. As the sessions are only kept up fo
I'm using the liveMedia (2012.11.05) connecting to a Cisco VSM (version
6.3.2-47d). The server, as I've mentioned in a previous post, requires the
client to wait for the TEARDOWN-response, otherwise the server logs will be
filled with errors and things will eventually go bad. This can arguably be
c
a
method from the wrong thread, usually inadvertently.
Matt S.
On Thursday, November 08, 2012 10:26:16 AM, Erlandsson, Claes P
(CERLANDS) wrote:
> I'm using the liveMedia (2012.11.05) connecting to a Cisco VSM
> (version 6.3.2-47d). The server, as I've mentioned in a previous pos
When downloading and building the latest version (2012-01-04) on Windows I
ran into problems.
I get a fatal error for all projects. Example for liveMedia:
liveMedia.mak(54) : fatal error U1036: syntax error : too many names to left
of '='
Stop.
I noticed Makefile.tail has changed re
I experience an issue with TEARDOWN that I can't resolve.
I've attached an example (testRTSPClientCycleStreams.cpp) that for me
results in an Unhandled exception after a while; it usually happens after 10
minutes to an hour. The example is the testRTSPClient example with some
modifications to cycl
I ran your modified application for several hours (on FreeBSD, under GDB),
and also on Linux using "valgrind", but unfortunately was unable to find any
problem.
I suspect that whatever bug is causing this is something that (for some
reason) is causing an exception only in your environment. I'm
Included the attachment I was mentioning below... It shows the last minute
or so of the output from the modified testRTSPClient.
/Claes
Thanks a lot for taking the time to test it.
I'm not really sure how to proceed, but I guess I can try to locate a Linux
client to compile and ru
fyi: While downloading and compiling the latest (live.2013.01.15.tar) code I
noticed testProgs, mediaServer and proxyServer still fails on Windows, i.e.
Makefile.tail contains ?= which nmake doesn't like:
PREFIX ?= /usr/local
/Claes
smime.p7s
Description: S/MIME cryptographic signature
hat I've seen recommended are
- "Dr. Memory": http://code.google.com/p/drmemory/
- "OllyDbg":http://ollydbg.de/
Thanks. Never heard of those two, but will look into.
I would also suspect threads going havoc, but as liveMedia is
singl
Once in a while, about 1 out of maybe 3000 connections, I notice a
connection failure. The client runs on Windows.
The DESCRIBE response handler receives resultCode -10057 and resultString
"Unknown error". I first suspected the server was to blame by not
responding, but when I look at socket er
In the following examples I'm writing about streams that are (in the code)
referred to as indexed by 'absolute' time. For video storage and security I
believe they're often referred to as archive streams.
Seeking while already streaming
When doing a "seek" while the stream is playing, i.e.
Is this intended, or is it somehow server dependent?
I don't know. Please show the complete RTSP protocol exchange for each
case, so I can try to figure out what's happening.
I've attached the protocol exchange for the following example:
1. Start up an archive stream without any ti
Can someone confirm if reverse play works fine?
Whenever I supply a negative scale value to the Play-function it always
plays forward. The speed is correct, but I've never been able to play in
reverse.
It might be some RTSP fluke with the Cisco VSM server we're using, but
wanted to verify i
Whenever I supply a negative scale value to the Play-function it always
plays forward. The speed is correct, but I've never been able to play in
reverse.
Don't forget to also specify a non-zero 'start time', so that the server
knows where to start reverse-play from.
Thanks, I do that. I
I unfortunately don't have much more info, but I believe I also experience
this issue.
Last week I upgraded from 2013.03.07 to 2013.04.23 and we suddenly started
to experience program crashes (liveMedia used within a DLL in Windows client
streaming mjpeg). I didn't suspect the liveMedia update
I receive a mjpeg stream from a Cisco server. In the SDP I see this line:
a=fmtp:26 width=704;height=480;4CIF=1
When looking at fVideoWidth & fVideoHeight in MediaSubsession they are
always 0 and it looks like they're only populated if "a=x-dimensions:%d,%d"
is found in the SDP.
I
We stream lots of live MJPEG streams using a pretty simple client that is
based on the testRTSPClient example. In very rare cases we see a delay/lag
that I can't explain.
When it happens it is like the live video is delayed 2s. If I at the time
start the same stream in VLC and specify a cache o
Thanks for the confirmation Ross.
We do use UDP, i.e. the default, which makes this lag so very strange. I can
clearly see all frames being received on the first UDP-port (when viewing
ports in use with e.g. TCPView).
This 2s lag has been observed three times in total over the last four
mon
When I start and stop the liveMedia event loop I see a small handle leak in
my Windows test client. When looking in Process Explorer I can see the
amount of file handles with the name "\Device\Afd" growing.
I don't have to stream anything for it to occur, i.e. I just start the event
loop and th
"\Device\Afd" handles
that are increasing. I believe "Afd" stands for Auxiliary Function Driver
and that it's related to sockets, but "Afd" is new to me as I've have never
come across it before.
Any idea where that handle leak originates? Could it be Windows
>I try to increase "OutPacketBuffer::maxSize" to 10 or 20 but the
>problem it's the same.
How big are your frames? I typically see larger frames than that from high
resolution cameras and have the buffer set to 50 or 100.
/Claes
smime.p7s
Description: S/MIME cryptographic sign
her
'#include's:
#ifdef __ANDROID_NDK__
#include
#define ANDROID_OLD_NDK __NDK_MAJOR__ < 17
#endif
I hope these changes will be useful. I have attached a patch file and resulting
source file for clarity. Have a blessed day.
Regards,
-Brian P. Chase
GroupsockHelper_cpp.patch
Description: Grou
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