> Is there any test code to send a TS file over a socket to the Live555 server
> and then provide that up as a stream?
Yes, look at the following two demo apps in the "testProgs" directory:
1/ testMPEG2TransportStreamer
2/ testOnDemandRTSPServer
Note the message:
"mpeg2Tra
> I create the replicator with the false option to avoid it deletes the input
> Source when removing all replicas.
> StreamReplicator::createNew(*env, Source, false)
>
> then create the source to feed it into the sink
> FramedSource* source = replicator->createStreamReplica();
>
> What would
Is there any test code to send a TS file over a socket to the Live555
server and then provide that up as a stream?
Thank you in advance.
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Hi Ross
I create the replicator with the false option to avoid it deletes the input
Source when removing all replicas.
StreamReplicator::createNew(*env, Source, false)
then create the source to feed it into the sink
FramedSource* source = replicator->createStreamReplica();
What would be the best
Thanks for the detailed Information.
I am using live RTP source as stream source to a subclass of
OnDemandServerMediaSubsession.
In this case RTP when source stops sending data , the application gets a
message upon which i want to close the session.
On Mar 16, 2012, at 1:53 AM, Ross Finlays
> How to invoke a RTCP BYE message to the client from the RTSP Server
> application (e.g testOnDemandRTSPServer) ?
This will happen automatically when the stream ends - i.e., when the server
reaches the end of the file that's being streamed. There is nothing that you
need to do to get this; th
Hi,
How to invoke a RTCP BYE message to the client from the RTSP Server application
(e.g testOnDemandRTSPServer) ?
Regards,
Sambhav
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Hi,
I have gone to the link you have mentioned
Yes. The easiest way to do this is to change the appropriate
"test*Streamer.cpp" file to read from "stdin" (instead of "test.*"), and
then pipe the output of your encoder to (your modified) "test*Streamer"
application.
I have changed char const
>>> When I am sending :
>>>
>>> SETUP rtsp://127.0.0.1/testStream/track1 RTSP/1.0
>>> CSeq: 4
>>> User-Agent: ./openRTSP (LIVE555 Streaming Media v2012.02.29)
>>> Transport: RTP/AVP;multicast;destination=230.1.1.1;client_port=7000-7001
>>>
>>>
>>> the testMPEG2TransportStreamer responds with:
>>
W dniu 15.03.2012 15:05, Grzegorz Gaj napisał(a):
>> You can't
do this with the "LIVE555 Media Server" (or "testOnDemandRTSPServer"),
because that is a unicast server (i.e., delivering streams via unicast
only). If you want to stream a Transport Stream file via multicast, then
I suggest using
> I have gone to the link you have mentioned
> Yes. The easiest way to do this is to change the appropriate
> "test*Streamer.cpp" file to read from "stdin" (instead of "test.*"), and then
> pipe the output of your encoder to (your modified) "test*Streamer"
> application.
>
> I have changed c
Hi All,
I have gone to the link you have mentioned
Yes. The easiest way to do this is to change the appropriate "test*Streamer.
cpp" file to read from "stdin" (instead of "test.*"), and then pipe the
output of your encoder to (your modified) "test*Streamer" application.
I have changed c
Hi All,
I have gone to the link you have mentioned
Yes. The easiest way to do this is to change the appropriate "test*Streamer.
cpp" file to read from "stdin" (instead of "test.*"), and then pipe the
output of your encoder to (your modified) "test*Streamer" application.
I have changed c
> I am trying to make a test using live555 rtsp server and openRTSP client to :
>
> Keep RTSP session between client and server using TCP (server : 192.168.0.1
> <-> 192.168.0.10 : client).
>
> Send from server raw udp transport stream to mutlicast address
>
You can't do this with the "LIVE555
Hi all !
I am trying to make a test using live555 rtsp server and
openRTSP client to :
Keep RTSP session between client and server using
TCP (server : 192.168.0.1 192.168.0.10 : client).
Send from server
raw udp transport stream to mutlicast address on desired port
(192.168.0.1 -> 230.0.
> I have problem with misordered frames from when
> fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP() is true. How is that
> possible?
It's not. The LIVE555 code delivers the contents of RTP packets in the correct
order (based on the sequence numbers in the RTP packets). (Note that the
Hi,
I have problem with misordered frames from when
fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP() is true. How is that
possible? Is it bug or should I reorder frames myself again? My console output
looks like this:
Stream "rtsp://192.168.0.25"; video/H264: Received 15227 byt
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