Thanks. I do understand how the event model works (Quite eloquent BTW).
The fact that it throws away complete NAL units if a piece of a fragment has
loss explains why it appears as dropping NALs.
Here is the only part I cannot figure out. Why does it not lose any packets if
I use OpenRtsp on th
> Could it be that the packets are incorrect and live555 cannot find the
> beginning and end, so it doesn’t clear out the buffer? Then it hits the max,
> dumps it and starts over.
No, but note that if - as with all payload formats - if one frame (in this
case, one H.264 NAL unit) is fragmented
> How can I fix this?
For now, only by (somehow) reconfiguring your Polycom server so that it puts
"0" first in the list of RTP payload format types in the SDP "m=" line.
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
___
live-devel maili
Thankyou.
I think I was able to confirm this with a printing breakpoint in
doGetNextFrame1() inside the if (fPacketLossInFragmentedFrame).
I also get a TCP_Zero window warning if tcp and socket unreachable if Udp
Could it be that the packets are incorrect and live555 cannot find the
beginni
How can I fix this?
Thanks
Markus.
From: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson
Sent: Thursday, May 02, 2013 2:49 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] playSIP - creates empty file
I am tryi
> If I run OpenRTSP on the command line and dump the frames into a file, all
> the frames are there and I can feed it back into VLC and it plays.
> In my code I am missing a lot of the P_FRAMES frame right after the I_FRAME.
> I then am randomly missing a few P_FRAMES randomly.
To me, this look
I have used live555 for years.
I am updated to 2013.04.30.
I tested with OpenRTSP on the command line with old and new versions of live555
and VLC on Linux and Windows, all work.
I am trying to debug an issue with H264 from security cameras and my
recorder/restreamer. I am hoping someone can giv
> I am trying to record the audio from a Polycom Telepresence m100 SIP software
> client.
>
> On 10.0.71.24 I run a software VTC client configure to use SIP (Polycom
> Telepresence m100).
> The URI is sip:10.0.71.24@10.0.71.24
>
> On 10.0.71.109 I run the live555 command line tool playSIP cal
I am trying to record the audio from a Polycom Telepresence m100 SIP software
client.
On 10.0.71.24 I run a software VTC client configure to use SIP (Polycom
Telepresence m100).
The URI is sip:10.0.71.24@10.0.71.24
On 10.0.71.109 I run the live555 command line tool playSIP calling 10.0.71.24.
