I'm working on rtsp performance test tool and I
will produce some code which I'd like to contribute. I know your strict
opinion on patches, so I'd like to know in advance if you would accept:
I don't have any definitive rules about what sort of patches I
accept. However, the patches that I'm m
We have implmented a receiver and it seems to be working fine for
most of the oulic content, however there are audio issues with seen
with this specific stream:
rtsp://mtvnmobile.qtod.llnwd.net/a4449/d1/n/124/17934/v001/mobilestor.download.akamai.com/17934/gametrailers.com/_tr_/83/8e/a4/t_reddea
I can provide you with a patch if necessary.
Yes, please do, and I'll take a look at it.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
___
live-devel mailing list
live-devel@lists.live555.com
http://lists.live555.com/mailman/listinfo/l
"openRTSP" application is client side at VOD structure. I really
want to ask is can I modify client_port at server side?
No. For unicast streams, the client's port is always choen by the client.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/__
Is Live555 compliant with ONVIF and PSIA?
To be honest, I hadn't heard of either of these things until you
mentioned them.
The standards body that defines Internet Standard protocols for video
streaming is the IETF, and it is those standards that we (at least
attempt to) follow.
I don't k
Many thanks to Ross Finlayson,
"openRTSP" application is client side at VOD structure. I really want to
ask is can I modify client_port at server side? Thanks in advance~
From: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.live555.com] On Behalf O
At 'SETUP' section in RTSP, client side send the request
includes client_port that it will used to receive rtp, these
client_port were determined by client side player, now due to the
firewall blocks most of ports, so I want to specify these
client_port, what should I do?
Note how the
Hi all,
Using the current release of OpenRTSP (2010.07.13), I noticed a similar error
in the num_packets_lost statistics line as Norbert Donath did in a old version
(look at
http://lists.live555.com/pipermail/live-devel/2004-December/001764.html).
The format of the packet lost number is curren
>Intended behavior. The standard mechanism by which servers detect
>the continued liveness of clients is via RTCP "RR"s. Note that the
>intention of the "timeout" parameter in the RTSP "SETUP" response is
>to indicate how long the server can wait after the last detection of
>client liveness b
how is the presentationtime of two streams synchronised?
Please read the FAQ!
I have to synchronise the mpeg-4 es and a wave file. I am able to
send the two streams together by creating single servermediasession
and adding two separate servermediasubsession, but they are not
synchronised.
I
Hi,
I am graduate student doing my master in Electrical Engineering at SDSU.
I am doing my thesis on efficient video streaming over multihop wireless
network using in band bandwidth estimation scheme ...
In addition, I am planing to use H.264 video codec and RTP as the transport
protocol ...
11 matches
Mail list logo
