In any case, to use your example, it's bad design for the RTSP
*clients* to be deciding which multicast address each server should
be using, not merely because it relies upon the server having this
ability enabled. It makes far more sense for each *server* to choose
(or be assigned) its multic
If is there other way to force RTSP streaming server to add the
frame rate and video frame size attributes into SDP description text?
I don't know; you're going to have to ask the authors of that server.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/__
I used debugger to trace the openRTSP, and I found it never called
"nv->taskScheduler().doEventLoop();" in the main function of openRTSP. I am
wondering why you put the code here and when it will be called by the program.
Thank you!
Hong Liu
--
>I found there was no the line of "a=x-dimensions:,"
>to update the default width and height by the real dimension. I just
>found the lines of "a=cliprect:0,0,288,352" and "a=framesize:96
>352-288".
>
>First, I didn't find any explanation about the sematics for these
>two attributes on Intern
>I'm working on an application that streams live generated content (audio
>and video) using the Darwin Streamnig Server. I decided to use ffmpeg
>for the encoding part and live555 for the streaming part.
We're no longer providing support for the "DarwinInjector" code.
Instead, we recommend that y
>./openRTSP -o -V rtsp://10.1.247.112:8554/testStream
>Sending request: OPTIONS rtsp://10.1.247.112:8554/testStream RTSP/1.0
>CSeq: 1
>User-Agent: ./openRTSP (LIVE555 Streaming Media v2007.05.22)
>
>
>Received OPTIONS response: RTSP/1.0 200 OK
>CSeq: 1
>Date: Thu, Jan 01 1970 02:23:39 GMT
>Public:
>I used openRTSP to get a H264 file in mp4 format from Darwin
>Streaming Server. I encoded into the H264 with image size 352x288 by
>x264 and encapsulated into the mp4 file using MP4Box. I tried the
>command of "openRTP -v -4 rtsp://192.168.0.1/a01.mp4" and however
>the received mp4 file is in
Hi Jay Furmanek,
I am about to implement the exact same functionality as you (playing an H264
Annex B encoded bytestream through live555) and was wondering if you'd be
willing to share your code :) to save me a couple of days work?
Thanks,
Abe Friesen
Hello list,
I'm working on an application that streams live generated content (audio
and video) using the Darwin Streamnig Server. I decided to use ffmpeg
for the encoding part and live555 for the streaming part. The basic
architecture is as follows: In a first thread I generate the content,
a
Hi,
I'm doing some tests with the test programs between two PCs
in my LAN. I'm not able to make the RTSP server transfer a file to
openRTSP.
What I did is:
start testMPEG4VideoStreamer on the first PC, and I get:
/home/live555-arm-20070524/testProgs/testMPEG4VideoStreamer
Play this stream using
Dear Ross,
I used openRTSP to get a H264 file in mp4 format from Darwin Streaming Server.
I encoded into the H264 with image size 352x288 by x264 and encapsulated into
the mp4 file using MP4Box. I tried the command of "openRTP -v -4
rtsp://192.168.0.1/a01.mp4" and however the received mp4 fi
Did you try your GNU C++ compiler? (probably arm-926ejs-linux-c++ in your
case).
Thanks for the hint Leon!
I modified the line in
config.armlinux
that define the "cross-linker" so that arm-926ejs-linux-c++
is used and now I have all compiled for my arm target!
Thanks a lot.
I.
--
"Much les
>>in my "IPTV QoS Monitoring" project i've tried to use openRTSP to get
QoS information for a particular stream. The output of jitter as a QoS
parameter is also interesting for me, so I'v extended the openRTSP code
to output this information. My question is, if I did it the right way,
because th
Hello
On 5/25/07, Igor Trevisan <[EMAIL PROTECTED]> wrote:
> Hi, I'm trying to cross compile for my arm9 target
> the "Live555 Streaming Media" packet and I got a long list
> of errors starting with the following:
>
> make[1]: Entering directory
> `/home/igotre/bin/live555-arm-20070524/testProgs'
>>in my "IPTV QoS Monitoring" project i've tried to use openRTSP to get
>>QoS information for a particular stream. The output of jitter as a QoS
>>parameter is also interesting for me, so I'v extended the openRTSP code
>>to output this information. My question is, if I did it the right way,
>>becau
Hi, I'm trying to cross compile for my arm9 target
the "Live555 Streaming Media" packet and I got a long list
of errors starting with the following:
make[1]: Entering directory
`/home/igotre/bin/live555-arm-20070524/testProgs'
arm-926ejs-linux-gcc -otestMP3Streamer -L.
testMP3Streamer.o../liveMed
>Please correct me if I'm mistaken but as fas as I can tell live555 does
>not fully implement RFC 3984 as it is now. Are there any plans to
>implement STAP and MTAP packets for interleaved streaming?
Our implementation supports the *reception* of STAP and MTAP packets
(in "H264VideoRTPSource"), b
>Hi, everyone
>I used the testOnDemandRTSPServer to stream video und openRTSP to
>receive the streamed data. My question is: where I should change the
>received packet data before the client stores the received packet on
>the disk, e.g. adding some data or a special header to the received
>pac
Dear Ross,
Please correct me if I'm mistaken but as fas as I can tell live555 does
not fully implement RFC 3984 as it is now. Are there any plans to
implement STAP and MTAP packets for interleaved streaming? Although
these packets are not very important for conventional video streaming,
they a
Hi, everyone
I used the testOnDemandRTSPServer to stream video und openRTSP to receive
the streamed data. My question is: where I should change the received packet
data before the client stores the received packet on the disk, e.g. adding
some data or a special header to the received packets. Any
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