packets?
No, it produces output for each frame, not NAL unit, so it is not a problem. I
have increased buffer size on network card to 2048, now it works much better.
I will try "openRTSP" also to see will it have some difference.
Thank you for your help
Best regards,
--
sing to 2048 helps a lot, but anyway if there are
some tricks / tunes in code that I can use in Live555 RTSP client - could you
please share this info?
Thank you in advance,
Best regards,
-----
Victor Vitkovskiy
Senior software developer
mailto: victor.vi
r answer,
Best regards,
-----
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
www.mirasys.com
-Original Message-
From: live-devel On Behalf Of Ross Finlayson
Sent: Monday, 7 February 2022 12:20
To: LIVE555 Streami
server and client to
use TLS?
Best regards,
-----
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
www.mirasys.com
-Original Message-
From: live-devel On Behalf Of Ross Finlayson
Sent: Monday, 7 February 2022 11:48
To: LIVE555 St
,
-
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
www.mirasys.com
-Original Message-
From: live-devel On Behalf Of Ross Finlayson
Sent: Monday, 7 February 2022 10:07
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-d
ion for non-HTTP/HTTPS server and we can't host only
HTTP / HTTPS, is that true?
Best regards,
-----
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
www.mirasys.com
-Original Message-
From: live-devel On Behalf
Hello Ross,
I have found an issue, one of subsessions was not added to RTSP server so it
was not freed on RTSP server close and that's why this UsageEnvironment reclaim
failed.
Best regards,
-
Victor Vitkovskiy
Senior software developer
m
delete this;
return True;
}
return False;
}
liveMediaPriv is not null, so it is not removed.
Could you please advise me how to make this liveMediaPriv null? What should be
called to do this?
Best regards,
-
Victor Vitkovskiy
Senior softwa
,
-
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
www.mirasys.com
-Original Message-
From: live-devel On Behalf Of Ross Finlayson
Sent: Wednesday, 26 January 2022 14:48
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] [Mirasys] Liv
roach fails with error described earlier just after the first part
of the frame is copied to output buffer.
Best regards,
-----
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
www.mirasys.com
-Original Message-
From
eems that we need to add also RTP header data to this 3 buffer (480 bytes)
and that’s why I have an error, is this correct?
Best regards,
-----
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
www.mirasys.com
-Origin
dynamically?
E.g. if I receive a frame that is bigger then OutPacketBuffer::maxSize and I
will increase this value, what should be done to update fTo buffer and it’s
fMaxSize for next doGetNextFrame call?
Best regards,
-
Victor Vitkovskiy
Senior software
Hello Ross,
I have found a mistake, sorry to disturb you
Best regards,
-
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
www.mirasys.com
-Original Message-
From: live-devel On Behalf Of Victor
Vitkovskiy
Sent
e what is wrong with RTSP client?
Best regards,
-
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
www.mirasys.com
-Original Message-
From: live-devel On Behalf Of Ross Finlayson
Sent: Friday, 14 January 2022 17:12
To: LIVE555 Streamin
gards,
-
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
www.mirasys.com
-Original Message-
From: live-devel On Behalf Of Victor
Vitkovskiy
Sent: Wednesday, 12 January 2022 13:20
To: LIVE555 Streaming Media - development & use
Su
Hello Ross,
Yes, that was the case, now it works, thank you for your help.
I will continue with this, still we have opened question with metadata
streaming.
Best regards,
-
Victor Vitkovskiy
Senior software developer
mailto: victor.vitkovs...@mirasys.com
teNew(*env, streamName,
streamName, descriptionString);
sms->addSubsession(H264ServerMediaSubsession::createNew(*env,
reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceURL(rtspServer, sms);
Could you please tell me what I am doing wrong?
Best regards,
---
e of no RTSP clients wants to get
stream.
So that's why I need to detect when new RTSP client want to receive stream from
RTSP server and also when client is done with stream receiving.
Best regards,
-----
Victor Vitkovskiy
Senior software developer
mailt
eNewStreamSource and createNewRTPSink, is
this correct?
Or I need to subclass from OnDemandServerMediaSubsession and do the same thing
(like reading SDP information from H.264 stream)?
Best regards,
-
Victor Vitkovskiy
Senior software developer
mailto:
Dear Support,
My name is Victor, I am investigating possibility to use Live555 in Mirasys VMS
system and I have several questions, could you please help me?
1. We need to create RTSP server that will stream data from our system if
client connects to it.
Unfortunately, I have not found any
Hello there. I got a couple of questions regarding the webrtc demo:
1. Does it support audio? I have an IP-Camera which emits two streams
over rtp, one is video the other audio.
2. Since the streams will be transcoded, how expensiv is the procedure
(in terms of CPU power and bandwith)?
Is th
and test other players rather then VLC. But
I hope that my patch will be useful for other people maybe with some
additional changes to support non-MJPEG frames.
--
Best Regards!
---
Victor V. Vinokurov
LAIN LLC Programmer
http://dialog-nibelung.ru
--- AVIFileSink.cpp.orig2017-07-18 09
Cam we get an answer please? Your webrtc server could solve many problems for
us.
From: Victor Rotenberg [mailto:vic...@servision.net]
Sent: Sunday, February 19, 2017 11:51 AM
To: 'LIVE555 Streaming Media - development & use' ;
'Ross Finlayson'
Subject: RE: [L
To: LIVE555 Streaming Media - development & use ;
Ross Finlayson
Subject: Re: [Live-devel] Webrtc server
never mind, I found the demo - http://webrtc.live555.com/
-Eric
On February 17, 2017 at 10:16 AM Ross Finlayson mailto:finlay...@live555.com> > wrote:
Victor,
I haven’t d
Hello!
We were impressed to see the demo page of the webrtc server converting rtsp
stream to web page. Can we download the webrtc server please?
Thanks.
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http://lists.live555.com/mailman/listinfo/liv
TNESS FOR A PARTICULAR PURPOSE.
regards,
victor--- End Message ---
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Thank you Ross for the very quick fix. It works fine with the new version.
Victor
___
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live-devel
r. The web server drops the request with 400 (Bad-Request)
because of the missing Host field. This happens before the request can reach
the mod.
Regards,Victor
___
live-devel ma
Dear experts,
I have read some of the FAQs and mail list.
Thank you for your effort.
Now the live555 server can stream out from encoder with unicast
udp/tcp or multicast respectively.
But my problem is that how to build rtsp server supports TCP and
Mulitcast at the same.
Good day!
For some reasons i want to have class MPEG1or2FileServerDemux seekable
to have the possibility to seek on the input file. My small patch do
this work. I hope that this possibility will be usefull not only for me
and will be included in future releases of live555.
--
See you!
---
Vityus
This is what happens :
[0x810aeb8] live555 demux debug: StreamClose
[0x81068b0] main input debug: EOF reached
[0x81068b0] main input debug: waiting decoder fifos to empty
vbv buffer overflow
[0x814de38] faad decoder debug: AAC SBR (channels: 2, samplerate: 16000)
[0x810aaf0] stream_out_transcode s
Good morning, and thank you for your great software.
Im victor from Brazil, and I am having a issue using vlc with rtsp
support (livemedia).
Im getting a stream from an axis camera (rtsp/mpeg4) on linux, and
everytime i get around 7 minutes of streaming, vlc says i have reached
EOF. and it stops
Ok, yes it does... How can I start reading from it?
Yes it does - just call "H264VideoRTPSink::createNew()"
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Any idea if i can find any specific example for H.264 ?
As i couldn't find base class for H264MediaSubsession, I am trying to
create H264 subsession with
PassiveServerMediaSubsession::createNew(RTPSink& rtpSink,
RTCPInstance* rtcpInstance)
The question now
I need to use certain predefined port numbers for RTP streaming, for
example 60004 for audio substream, 60006 for audio...
I tried to set response header for "DESCRIBE" to use those numbers, like:
m=video 60004 RTP/AVP 32\r\n vs. m=video 0 RTP/AVP 32\r\n but it
doesn't seem to make any difference
Hello,
I have 2 UDP sockets receiving synchronized RTP audio ( AAC) and RTP
video (H.264) streams and i have to re- send that stream to RTSP
client.
Any idea how to connect RTSPServer to this source? It looks like I
have to override "lookupServerMediaSession" ..
Do i need to create new type of Med
I am trying to follow basic flow of events in Dynamic MediaServer sample...
Can someone point me how the "play" event is signaled to RTP sink to
start actual streaming?
Thanks,
-V
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http://lists.live5
What is the timestamp unit in openRTSP? second or microsecond?
Thank you.
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If I do need to get the time, how can I?
2009/12/7 Ross Finlayson
> I want to add pause and replay function in openRTSP. I think I can user
>>
>> RTSPClient::pauseMediaSession(MediaSession& session)
>> &
>> Boolean RTSPClient::playMediaSession(MediaSession& session, double start,
>> double end,
I want to add pause and replay function in openRTSP. I think I can user
RTSPClient::pauseMediaSession(MediaSession& session)
&
Boolean RTSPClient::playMediaSession(MediaSession& session, double start,
double end, float scale)
to realize this function. But I have to get the time when it pause ,so
t?
Many thanks in advance
Victor
-Mensaje original-
De: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.live555.com] En nombre de Ross Finlayson
Enviado el: martes, 06 de octubre de 2009 4:04
Para: LIVE555 Streaming Media - development & use
Asunto: Re: [Live-devel] Li
/www.live555.com/liveMedia/doxygen/html/classServerMediaSession.html>
ServerMediaSubsession" ?
Many Thanks in advance
Victor
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I use live555 lib under windows.
I write code to get MPEG file duration:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
BasicUsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
MPEG1or2FileServerDemux* demux =
MPEG1or2FileServerDemux::createNew(*env, inputFileN
i've try to get file duration of mpeg1 video file like this:
int VIDEOMC_API GetFileDuration(const char* inputFileName)
{
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
BasicUsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
MPEG1or2FileServerDemux* demux =
MP
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