> I am sending a RTP + MPEG-TS live stream from a windows machine over the WAN
> to a server machine which is running "testMPEG2TransportReceiver" and piping
> its output to "testOnDemandRTSPServer". I then connect a client to receive
> the RTSP stream. I noticed that the "testOnDeamandRTSPServe
--- Begin Message ---
I am sending a RTP + MPEG-TS live stream from a windows machine over the WAN to
a server machine which is running "testMPEG2TransportReceiver" and piping its
output to "testOnDemandRTSPServer". I then connect a client to receive the RTSP
stream. I noticed that the "testOnDe
> I am using testOnDemandRTSPServer. I can stream using TCP or UDP from QT
> player(with client side settings changed). What changes do I need to make so
> that only UDP streaming is allowed.
Venkat,
I'm a bit surprised by your question, because most people consider
RTP/RTCP-over-TCP stre
Hi,
I am using testOnDemandRTSPServer. I can stream using TCP or UDP from
QT player(with client side settings changed). What changes do I need to
make so that only UDP streaming is allowed.
--
*With Warm Regards,*
Venkat Karthik
VVDN Technologies Pvt Ltd
*Cell : *+91 9884292064 | *Skype
> Hi, I'm having a problem getting the testprog RTSP Server to actually show a
> video file. I encoded the video file myself so it should be the correct
> format (MPEG4 H.264 Baseline AVC).
Is this a H.264 Video *Elementary Stream* file - i.e., with video only, and
*not* in a MPEG-4 file format
On 18/12/12 13:52, Ross Finlayson wrote:
No. As you can see from the "MPEG2TransportUDPServerMediaSubsession"
implementation, a new "MPEG2TransportStreamFramer" is created (and
used as the data source) each time "createNewStreamSource()" is called
- i.e., each time we start reading from the in
> I have a hunch at this point that the TransportStreamFramer is not being
> reinitialized when a new client arrives
No. As you can see from the "MPEG2TransportUDPServerMediaSubsession"
implementation, a new "MPEG2TransportStreamFramer" is created (and used as the
data source) each time "creat
On 18/12/12 11:20, Ross Finlayson wrote:
Unfortunately I wasn't able to reproduce your problem at all. I ran
"testMPEG2TransportStreamer" to continuously stream a Transport Stream
file via multicast, and also ran "testOnDemandRTSPServer" to receive
this multicast stream, and use it as input fo
Unfortunately I wasn't able to reproduce your problem at all. I ran
"testMPEG2TransportStreamer" to continuously stream a Transport Stream file via
multicast, and also ran "testOnDemandRTSPServer" to receive this multicast
stream, and use it as input for a unicast RTSP server. I then kept runn
On 17/12/12 13:53, Ross Finlayson wrote:
First, I'll try to look into exactly what's happening. If there's a
bug in the supplied LIVE555 code, then I'll try to fix it.
Thanks Ross, I appreciate this greatly.
I have done a little further investigation myself and thought I'd share
it with yo
> I have just noticed something - when the first client connects, there is an
> IGMP membership report issued to join the group I have specified for any
> sources.
>
> However, when that first (and only) client tears down the session, there is
> no IGMP membership report for the 'Leave'.
That'
On 17/12/12 12:18, Ross Finlayson wrote:
If you do this, then, yes, the input source object will get closed
(and its destructor called) whenever the last RTSP client leaves.
This is the proper behavior, because we want the input source to be
closed when noone is requesting its data. (Similarl
On 17/12/12 12:18, Ross Finlayson wrote:
Since the "testOnDemandRTSPServer" demonstrates how to stream from
*files* to (unicast) clients, it does not 'demonstrate' multicast to
unicast RTSP relaying at all. Therefore, you must have modified the
supplied application's code in some (unspecified)
> I have been experimenting with a multicast to unicast RTSP relay, as
> demonstrated in testOnDemandRTSPServer.
Since the "testOnDemandRTSPServer" demonstrates how to stream from *files* to
(unicast) clients, it does not 'demonstrate' multicast to unicast RTSP relaying
at all. Therefore, you
Hi Ross,
I have been experimenting with a multicast to unicast RTSP relay, as
demonstrated in testOnDemandRTSPServer.
A behaviour I have noticed, is that the first client connects and
receives the stream correctly, and all subsequent connections do too -
they can SETUP, PLAY and TEARDOWN to
hi,everyone
i replace video file with pipeline for testOnDemandRTSPServer.cpp,but i find i
could not play in rtsp client.However i could play in testH264VideoStreamer.cpp
by the same way.
i research code in two cpp file.i find startplaying code in two cpp file are
similar,that is
H264VideoRTPSi
Unfortunately I can't explain your problem.
However, if your intention is to use our RTSP server implementation to stream a
live Transport Stream, then I suggest that you have the server access the
Transport Stream data directly (as an input source), rather than having the
Transport Stream data
Yes, this is Apple's problem, not ours; See:
http://www.live555.com/liveMedia/faq.html#quicktime-player-mp3-bug
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
___
live-devel mailing list
live-devel@lists.live555.com
http://lists.li
Hi all
It seems using the testOnDemandRTSPServer example program somehow causes the
QuickTime app to play an mp3 silently.
I have run live/testProgs/testOnDemandRTSPServer on a machine that is also
running Apache from a directory that contains various wav and mp3 files. I have
an index.html fi
> i run the server with the command ./testOnDemandRTSPServer, but by using the
> request rtsp://192.168.1.3:8554/mpeg2TransportStreamTest i can read only one
> stream "test.ts".
> the problem that I just want to read multiple streams "*.ts" with the same
> runtime.How can i distinguish them?
V
t.
Thanks for your help Ross!
- Mensaje original -
De: Ross Finlayson
Fecha: Jueves, Febrero 3, 2011 3:56 am
Asunto: Re: [Live-devel] testOnDemandRTSPServer always stream audio/MPA and
video/MPV?
> >When progress >= duration I do a 'loop' on the stream, something
&
When progress >= duration I do a 'loop' on the stream, something
like (sorry i dont have the code here):
pauseStream
seekStream (to the abs byte 0)
startPlaying
That work fine when I stream a MPEG1or2* file, but when I stream
using the way I just mentioned (MPEG4 + ADTS), it does not work.
D
Ross,
I checked what you mentioned about the duration...
I see that fDurationInMicroseconds is filled where you said, but method
duration() of OnDemandServerMediaSubsession is always 0 for both
streams, while if I stream a mpeg file with MPEG1or2* objects,
duration() method returns the durati
I learned that I should not modify the source code Ross ;)
I did not modify any part of the code, anything change I need, I make
subclasses.
I want to finish this to make a test and send it here, maybe you can add
it to the lib or maybe it will be useful for somebody.
I need to check about
sms->addSubsession(MPEG4VideoFileServerMediaSubsession::createNew(*env,
v_inputFileName, reuseFirstSource));
sms->addSubsession(ADTSAudioFileServerMediaSubsession::createNew(*env,
a_inputFileName, reuseFirstSource));
[...]
And I have a stream that have MPEG4-GENERIC and MP4V-ES.
The prob
to retrieve duration of MPEG4Video
and ADTSAudio.
It isnt implemented yet? or am I doing something wrong?
Thanks!
- Mensaje original -
De:
Fecha: Lunes, Enero 31, 2011 12:53 pm
Asunto: Re: [Live-devel] testOnDemandRTSPServer always stream audio/MPA and
video/MPV ?
> I see, so I'm
o: Re: [Live-devel] testOnDemandRTSPServer always stream audio/MPA and
video/MPV ?
> >So, is there any way to stream a MPEG4+AAC file with live555?
>
> No, we currently do not have code for demultiplexing a ".mp4" or
> ".mov"-format file.
> --
>
So, is there any way to stream a MPEG4+AAC file with live555?
No, we currently do not have code for demultiplexing a ".mp4" or
".mov"-format file.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
___
live-devel mailing list
live-devel@l
Ross,
Thanks for your answer.
I understand what you say, after sending that mail I start checking the 'stream
process' and I found the codecs problems and also why it's using always MPA/MPV.
My question now changes, I need to stream a video+audio file, I have a .mpg
file but I can convert it
I'm seeing something that I consider weird.
There's nothing "weird" about this. The diagnostic output is quite
clear about what the software is doing (or more precisely, is not
doing) in this case.
Warning: We don't implement a QuickTime Video Media Data Type for
the "MPV" track, so we'll
Hello,
I'm seeing something that I consider weird.
I downloaded a lot of .mpg files which I'm streaming using
testOnDemandRTSPServer. I run it and then launch openRTSP with '-q' arg to
obtain a mov file.
When openRTSP open the stream I see:
Created receiver for "video/MPV" subsession (client po
Hi,
I am new to live 555. Could someone to me how to use the test
program TestOnDemandRTSPServer. I compiled and executed this test
program. All it gave was a list of urls. But I am not able to play
any video from that url. Help required immediately please.
I very much doubt that you re
Hi,
I am new to live 555. Could someone to me how to use the test program
TestOnDemandRTSPServer. I compiled and executed this test program. All it
gave was a list of urls. But I am not able to play any video from that url.
Help required immediately please. Thanks a lot.
_
Hi,
I'm developing a custom subclass of OnDemandServerMediaSubsession for a
video encoder, and I have integrated it into the testOnDemandRTSPServer
application. I noticed that when I try to open multiple VLC windows for
viewing the stream (I have reuseFirstSource set to True), I can open up to
I have a problem with testOnDemandRTSPServer with mpg file!
In fact, I receive the mpeg ok but audio is broken (audio is ok
after pause or stop and re-play).
Why? Video Subsession is ok, Audio Subsession don't work very well.
I use VLC to play the stream!
Please put the (".mpg") file on a pu
Hi,
I have a problem with testOnDemandRTSPServer with mpg file!
In fact, I receive the mpeg ok but audio is broken (audio is ok after pause or
stop and re-play).
Why? Video Subsession is ok, Audio Subsession don't work very well.
I use VLC to play the stream!
Thanks in advance,
Bobo
__
>Hi ,
> I am trying to test testOnDemandRTSPServer with a
>test.mpg file to play it as an mpeg1or2AudioVideoTest
>but getting the following errors when i request a
>play request from VLC player
>
>MPEG1or2VideoStreamParser::parseSlice(): Saw
>unexpected code 0x1b5
>MPEG1or2VideoStreamParser::par
Hi ,
I am trying to test testOnDemandRTSPServer with a
test.mpg file to play it as an mpeg1or2AudioVideoTest
but getting the following errors when i request a
play request from VLC player
MPEG1or2VideoStreamParser::parseSlice(): Saw
unexpected code 0x1b5
MPEG1or2VideoStreamParser::parseSlice(
38 matches
Mail list logo
