> I am using live555 library to stream to proxyserver on rtp over tcp. I want
> to add variable bitrate mechanism for my device for that i suppose i need to
> verify some parameters on RTCP RR report. I was trying to look at packet loss
> status but everytime it was a constant value coming 0x00f
Hi Ross,
I am using live555 library to stream to proxyserver on rtp over tcp. I want
to add variable bitrate mechanism for my device for that i suppose i need
to verify some parameters on RTCP RR report. I was trying to look at packet
loss status but everytime it was a constant value coming 0x00ff
I have another small question.
To synchronize my clients, I do some calculations based on the
presentation time.
Is it possible that the presentation time contained in the frame
send by the server can be modified by live555 library on the client
side?
Yes, see
http://www.live555.com/l
?
Thanks in advance.
Regards,
Jacques Tebcherani
_
From: Ross Finlayson [mailto:finlay...@live555.com]
Sent: mardi 18 janvier 2011 12:27
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] Packet loss when receiving streamed data via TCP
I have developed a
I have developed a PCM audio Server/Client based on "LIVE555
Streaming Media" library (v2010.04.09).
Sorry, but absolutely no support is given for old versions of the code.
Does live555 can discards some frames?
No. Our code simply runs in a loop, sending frames (if it's a
server), or rec
Hi,
I have developed a PCM audio Server/Client based on "LIVE555 Streaming Media"
library (v2010.04.09).
I have configured my server to stream in unicast mode.
I have configured my client to receive the stream via TCP
(setupMediaSubsession() is called with streamUsingTCP parameter set to T
Hi All,
I have attempted to implement a packet loss mechanism; its description is given
below. I am facing some problems with it, both in terms of testing and the
logic behind the implementation. Please give me your opinions and inputs on how
to make this more efficient and robust.
Implementat
I am trying to simulate various levels of packet loss in the live555
streaming server ranging from 5-50% in steps of 5. The change has to
be done in MultiFramedRTPSource.cpp/MultiFramedRTPSink.cpp but it
uses (our_random() % 10) to generate 10% packet loss.
This method does not work for values
Hi All,
I am trying to simulate various levels of packet loss in the live555 streaming
server ranging from 5-50% in steps of 5. The change has to be done in
MultiFramedRTPSource.cpp/MultiFramedRTPSink.cpp but it uses (our_random() % 10)
to generate 10% packet loss.
This method does not work f
I'm using the simpleRTPsource in order to receive rtp/udp packets
which encapsulates TS (encapsulating an h.264 video stream). only
made a simple adjustment to testMPEG1or2VideoReceiver.
When my source is Live555 and the testMPEG2TransportStreamer it's fine.
However, I'm experiencing heavy pack
Hi,
I'm using the simpleRTPsource in order to receive rtp/udp packets
which encapsulates TS (encapsulating an h.264 video stream). only made
a simple adjustment to testMPEG1or2VideoReceiver.
When my source is Live555 and the testMPEG2TransportStreamer it's fine.
However, I'm experiencing hea
>I used the ¡°iperf¡± to check the Bandwidth
>between the PCs which have 3.8 Mbits/sec
Your video codec is "MPV", which is MPEG-1 or 2
video. Many MPEG-1 or 2 video streams are close
to, or greater than, 3.8 Mbits/second.
Perhaps your network just doesn't have enough bandwidth for your strea
Dear All,
Thank you very much for your quick response!
>First, you should make sure that your network has sufficient >bandwidth for
>your data stream.
I used the “iperf” to check the Bandwidth between the PCs which have 3.8
Mbits/sec >However, packet loss can be caused by insufficiently la
My earlier answer (to another questioner) also applies here:
--
First, you should make sure that your network has sufficient
bandwidth for your data stream.
However, packet loss can be caused by insufficiently large socket
reception buffers in the rec
Dear all,
Thank you very much for your help first.
I have a packet loss problem when using the “live555MediaServer.exe” and
“openRTSP.exe” for streaming a mpg video file.
I am using the following setup.
Server PC: P4 1.8G, 1.5G Ram, WinXP.
Client 1: IBM T40, 1.25G Ram, WinXP
Client 2: Ace
I suggest first running "openRTSP", with the "-Q" option, to see if
you really are seeing network packet loss.
If your packet loss is really due to lost packets on the network,
then there isn't much you can do about this, except get a better
network.
--
Ross Finlayson
Live Networks, Inc.
http
Hi Ross,
I am a novice programmer.I am doing video streaming over
WAN(using TCP).I am testing mpeg4 video with 25fps and 100kb bitrate.I was
getting very jerky stream on client side.When I tested I found error 10035 on
server.So I increment the os buffer size from 50kb to 128 kb.N
Finlayson
Sent: Thursday, October 11, 2007 8:04 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] Packet loss in case of high bit rate mpeg4 streams
As you've noted/realized, any packet loss - other than network packet loss -
that occurs in a system that uses
As you've noted/realized, any packet loss - other than network packet
loss - that occurs in a system that uses our code *must* be the fault
of insufficient buffering inside the OS. I.e., this is an OS
problem, and any fix/workaround needs to address this.
Unfortunately there is no good way fo
Hi Ross, hi Nitin,
I am also experimenting packet lost or malformed packets when I use high
bitrate clips and rtp over tcp streaming transport.
When I turn on the DEBUG mode, I noticed that sendRTPoverTCP generally returns
"failed" status. The errno is set to EAGAIN (or EWOULDBLOCK)
and is not
for
LIVE555MediaServer hence this explain the earlier mail.Sorry for that.
Regards
Nitin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ross Finlayson
Sent: Monday, October 08, 2007 7:03 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Liv
>We further observed that there is no packet loss on the server side
>if we use testOnDemandRTSPServer.cpp demo program to stream high bit
>rate mpeg4 elementary stream as compared to RTSP LIVE555MediaServer
>program for the same. So what could be the possible reason for this
>behavior when bot
Hi Ross,
We further observed that there is no packet loss on the server side if we use
testOnDemandRTSPServer.cpp demo program to stream high bit rate mpeg4
elementary stream as compared to RTSP LIVE555MediaServer program for the same.
So what could be the possible reason for this behavior when
.From wireshark trace on server side we found out that server is
dropping rtp packets.
Do you really mean to say that the *server* is dropping packets?
I.e., are you seeing packet loss in the over-the-network packets
traces? If so, then I don't know what might be causing this, and
unfortunat
Hi all,
We are using Live555 media server and openRTSP client to stream and receive
mpeg4 elementry stream.We observerd that when we stream high bitrate MPEG4 ES
stream (3 or 6 or 15 Mbps )there is a huge packet loss(almost 20% ) .From
wireshark trace on server side we found out that server i
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