in for the clarification.
I'll look into the peculiarities of the way windows reports these socket
timeouts.
Regards,
Mark.
[cid:image001.png@01D7ECF7.742618D0]<http://www.scientificgames.com/>
Mark Hinchcliffe
Solutions Architect
Scientific Games
O: N/A
M: N/A
Visit SGGaming.com<http
e willing and able to add the fix in at
your end.
Thanks.
[cid:image001.png@01D7EC23.8F97F4F0]<http://www.scientificgames.com/>
Mark Hinchcliffe
Solutions Architect
Scientific Games
O: N/A
M: N/A
Visit SGGaming.com<http://www.sggaming.com/>
[A picture containing drawing, table De
Hi Ross,
Thanks for the quick response.
I followed your instructions but sadly no joy, but I think I know why.
It seems the vpn has been interfering.
When I turn off the vpn running just the *unmodified* testH264VideoStreamer
with a file as input, it works, I can connect a client to it and pla
k_1_converter_sink passed in to the PassiveServerMediaSubsession.
Am I on the right train of thought?
Any help is greatly appreciated.
[cid:image005.png@01D70BB2.CDAAB540]<http://www.scientificgames.com/>
Mark Hinchcliffe
Solutions Architect
Scientific Games
O: N/A
M: N/A
Visit SGGamin
o
with "B" frames), then you should insert the appropriate "**Discrete*Framer"
filter between your source object and your "*RTPSink" object.
Mark Woodard
___
Mark Woodard
Mutualink, Inc.
313 S Jupiter
Allen, TX 75002
T
MING_OVER_TCP set to False. Will it, in fact, use UDP?
___
Mark Woodard
Mutualink, Inc.
313 S Jupiter
Allen, TX 75002
Toll Free: (866) 957-5465 Ext: 508
Direct: (972) 200-5020
E-Mail: mwood...@mutualink.net
Web: www.mutualink.net
*Certified
OK. So testRelay is a good starting basis. The only difference in my case
is that the source stream is unicast, not multicast.
Do I need to modify anything about the BasicUDPSink (or anything else) to
accommodate a unicast source?
___
Mark Woodard
OK, Ross, let me rephrase the question.
Does the testRelay example accept RTP packets from a source and pass them
on unchanged to the destination? If the packets are not unchanged, what is
changed about them?
___
Mark Woodard
Mutualink, Inc.
313 S
My source is the Milestone ONVIF Bridge, which I believe was built using
Live555 libraries. It deals with RTP over UDP.
My destination is a Mutualink Edge Server. It also expects RTSP packets.
___
Mark Woodard
Mutualink, Inc.
313 S Jupiter
Allen
I was under the impression that the testRelay sample
*was* relaying the data as RTP/UDP. I'm pretty sure that's what the
destination expects.
So should I be using the Proxy Server instead?
___
Mark Woodard
Mutualink, Inc.
313 S Jupiter
Allen,
55;
Groupsock outputGroupsock(env, outputAddress, outputPort,
outputTTL);
return (OurRelaySink*)BasicUDPSink::createNew(env,
&outputGroupsock, packetSize);
}
return NULL;
}
___
Mark Woodard
Mutualink, Inc.
313 S Jupiter
Allen
el [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Mark
Bondurant
Sent: 01 December 2014 18:11
To: 'LIVE555 Streaming Media - development & use'
Subject: Re: [Live-devel] Playback Speed
Oh. I didn't know that. I assumed it had to be part of the h264 headers (one or
the o
stered Office : 71 The Hundred, Romsey, SO51 8BZ. Company Number : 03428325
From: live-devel [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Mark
Bondurant
Sent: 25 November 2014 17:23
To: 'LIVE555 Streaming Media - development & use'
Subject: [Live-devel] Playback Speed
I'm
n charge of maintaining the timing
information and the player you use or write uses the stored timestamps to gate
out the frames.
Double speed is ominous though. Could it be that you have interleaved frames?
If it is AVI, then the creation code was passed the wrong numerator? or
denominator?
On T
I'm encountering a curious problem with payback speed. I'm pulling an RTSP play
stream from my cameras, copying the NAL's, prepending them with the start code,
as contiguous GOP units straight to disk, headers and all. And they play fine.
Just at double speed. The camera is set to NTSC/H264/CIF/
f a "Frame-able Source" and an H264
RSTP stream isn't? But if it isn't, why does everyone say it's simple to do?
From: live-devel [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Mark
Bondurant
Sent: Tuesday, November 04, 2014 11:16 AM
To: 'LIVE555 Streaming Media
I get this part now, but what I still don't get is how to pass incoming data
from RSTPClient to H264VideoStreamFramer. The constructor for it is:
H264VideoStreamDiscreteFramer::createNew(UsageEnvironment& env, FramedSource*
inputSource) {
A FramedSource input, but RSTPClient is not a FramedSour
Mon, Nov 3, 2014 at 6:52 PM, Mark Bondurant
mailto:ma...@virtualguard.com>> wrote:
As you surly must know, I'm a noob thrust unwillingly by circumstances into
this.
This is helpful. I don't need frames, just ten seconds of stream. But, doesn't
H264 have a definite beginnin
As you surly must know, I'm a noob thrust unwillingly by circumstances into
this.
This is helpful. I don't need frames, just ten seconds of stream. But, doesn't
H264 have a definite beginning with the following NAL packets updating the
initial packet? That's what all that predictive slices and
.
Now, here you've lost me. I don't know what a GOP is. I don't need forward or
backward. I just need to spit out a discrete autonomous ten second clip.
Someone else will play it.
On Mon, Nov 3, 2014 at 4:02 PM, Mark Bondurant
mailto:ma...@virtualguard.com>> wrote:
Hello
Sorry. That wasn't clear. Yes, FramedSource derives from MediaSource, which
derives from Medium. RTSPClient derives from Medium.
In the example program you have the ourRTSPClient with a StreamClientState
object attached. You "strobe" the session object to cause the client to pump
frames through
MediaSource, not Medium,
which RTSPClient derives from. RTSPClient doesn't fit together with
H264DiscreteFramer! (clunk, clunk, me trying to squish them together).
When things don't fit, I think that I'm missing something important. So what
I'm asking for is a clue. What am I missin
ntainer for streams sharing the same clock base.
>
> Is this to sync multiple cameras in close proximity? or distant?
>
>
> On Mon, Apr 28, 2014 at 7:23 AM, Mark Theunissen <
> mark.theunis...@gmail.com> wrote:
>
>> > What is the resolution / profile?
>>
>> I&
r, no buffer so that video is sent ASAP?
Gstreamer manages extremely low latency, but uses too much CPU for my
application.
I'm just using the test app modified to read from stdin.
Thanks
Mark
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live-devel@lists.live555.com
I've now implemented the suggestion you have above where I base the rendering
on the 'presentation time' from live555 - all works smoothly, both the
troublesome feeds and the ones that previously worked without this.
Thanks!
Mark.
From: Ross
large being for full hd?
I'm happy enough leaving the collection method in my RTSP client, maybe perhaps
just for H264, but I was wondering if this is masking a problem with something
else, or maybe collecting by timestamp isn't a good idea for some reason?
Thanks in advance for any s
ing.
Please assist with solving the playback of a speex rtp stream in
real-time.
Thanks,
Mark
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live-devel@lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
Was wondering if anyone here had an example or knew if it would be
relatively painless to integrate the Live555 RTSP server with an
application that ends up with video information in either an openGL
PBO or via directShow. Any pointers? Thanks guys.
Mark
Hi Ross,
I just read the rtcp.cpp and find a problem in multicast mode.
In RTCPInstance::incomingReportHandler1(), from line 342 :
// Ignore the packet if it was looped-back from ourself:
if (RTCPgs()->wasLoopedBackFromUs(envir(), fromAddress)) {
// However, we still want to handl
Hello there.
Since I could not google my way to config file suitable for
cross-compiling for iPhone OS (on a Mac), I wrote my own and decided
to share it with the community.
---
~ http://animuchan.net/ ~
config.iphoneos
Description: Binary data
___
li
ing a script. If no port is specified I've
set the server to start on port 8554. This is perhaps not everybody's default
but suits our environment.
Kind regards,
Mark Liggett
BTI Systems
24d23
< #include
40,78d38
< /*
< * Added command line option -p to specify a tcp por
point
using random access. I was wondering if anyone could tell me if this is
possible and if so, what files in the live555.com libraries I would need to
change to make it happen. Thanks in advance.
Best regards,
Mark
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live
E and continues to use the original URL that I gave
it.
SO, when i send SETUP and PLAY to:
rtsp://wmlive.bbc.co.uk/wms%5Cbbc7%5Chi_s1/
I can connect to the BBC and download their stream without a problem (using
a SimpleRTPSource). I'm not sure whether the server or client is at fault
here,
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