I implemented H264Payload, but I found my first RTCP SR was
sent very lately. It was sent after the 150th RTP package.
This is probably because your "ServerMediaSubsession" subclass's
"createNewStreamSource()" function is not setting the result
parameter "estBitrate" correctly. That
Hi, Guys!
I implemented H264Payload, but I found my first RTCP SR was sent
very lately. It was sent after the 150th RTP package.
And the timestamps in it were not as same as RTP packages, NTP time was
error.
How shall I fix those problems?
Best wishes,
Aiden
_
In other words, once I do the above, what else
do I have to do to clean up in rebuild again in preparation of
streaming the next file?
If you want to keep the existingoutput network connections, and
continue streaming the next file, then you don't need to make *any*
changes to the existing
Thus the code can be changed to
double timeDiff
= timestampDiff/(double)(timestampFrequency);
I believe this change make keeping a reference to fOurRTPSource
In class RTPReceptionStats and RTPReceptionStatsDB useless.
Yes, you're right. This will get cleaned up in the next release of
the
1. When I try to use the command as follow:openRTSP -s 0 -v
rtsp://admin:[EMAIL PROTECTED]:6000/udpstream | testOnDemandRTSPServer
I can saw the video using VLC at other computer. But the video
quality is so bad enough(lots of mosaic, lose lots of frame, delay etc.).
How should I i
Dear Ross, all,
I'm new to live555. I have a video server that it can stream out video
data and the stream's format is use mpeg4. I can receive stream from that
device using openRTSP. And I also can see mpeg4 video in VLC using
testOnDemandRTSPServer. Now, I want to receive stream from the video
Minor comment about RTPSource.cpp.
While reading the code for some understanding of the internal mecanism I
came
across the code.
double timeDiff
= timestampDiff/(double)(fOurRTPSource.timestampFrequency());
in method RTPReceptionStats::noteIncomingPacket.
But the method already receive th
On Thu, Oct 16, 2008 at 6:17 AM, Ross Finlayson <[EMAIL PROTECTED]> wrote:
>
>> The behavior I expect is that at the end of the stream, when
>> stopPlaying() is called on both sinks, live555 should send and RTCP
>> Bye packet, causing MPlayer to exit.
>
> No, that's not what the code does. When th
I have cross compiled openRTSP for mips platform and I am using its for
recording mjpeg streams(30fps frame rate 32kb frame size). I need to run
multiple instances of this client ( say 16). But the problem is one
openRTSP client consuming around 12% of CPU time.
That's strange, considering th
Hi All,
I have cross compiled openRTSP for mips platform and I am using its for
recording mjpeg streams(30fps frame rate 32kb frame size). I need to run
multiple instances of this client ( say 16). But the problem is one
openRTSP client consuming around 12% of CPU time. So it is difficult to
run
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