> On Aug 30, 2019, at 6:44 AM, Son Nguyen wrote:
>
> I’m using Live555 to make VOD service support TrickPlay mode. As I saw that
> trick play mode only support .ts containers with indexed. But seems your RTSP
> client could not record RTSP live stream to TS file or i dont know how to do
> it
Hello
I’m using Live555 to make VOD service support TrickPlay mode. As I saw that
trick play mode only support .ts containers with indexed. But seems your
RTSP client could not record RTSP live stream to TS file or i dont know how
to do it.
Other question: Could we provide one RTSP server (one po
On Aug 29, 2019, at 10:39 AM, Vishnu Mohan wrote:
>
> I meant to say is there a reason/s why Webrtc is not a good method for
> webserver based streaming implementations ?
Ross said in his first reply that he had a bunch of compatibility problems. I
haven’t tried to implement anything involv
HI Ross,r
I meant to say is there a reason/s why Webrtc is not a good method for
webserver based streaming implementations ?
With Warm Regards,
Vishnu Mohan
VVDN Technologies Pvt Ltd
*Cell* : +91 9916316090 |* Skype* : vishnumohan1991
On Thu, Aug 29, 2019 at 9:52 PM Ross Finlayson
wrote:
>
Webrtc is typically used in p2p applications. RTSP strikes me more as a
server/client relationship which would be better suited for websockets.
On Thu, Aug 29, 2019 at 9:06 AM Vishnu Mohan
wrote:
> Hi Ross,
>
>I was using HLS based transport streams for stream over HTTP, Some
> times with so
> On Aug 29, 2019, at 12:00 PM, Vishnu Mohan wrote:
>
> Hi Ross,
>
>I was using HLS based transport streams for stream over HTTP, Some times
> with some engineers input I used to get confused with the webrtc technique.
> As per your comments, "I no longer believe that it’s the best way t
Hi Ross,
I was using HLS based transport streams for stream over HTTP, Some times
with some engineers input I used to get confused with the webrtc technique.
As per your comments, "*I no longer believe that it’s the best way to get
video from RTSP sources to render in web browser*", Can you poi
> On Aug 29, 2019, at 2:34 PM, Massimo Perrone via live-devel
> wrote:
>
> Ok, thank you. And when a paused session is resumed the frames presentation
> time remain "in the past", but RTP timestamps - due to the invocation of
> RTPSink::presetNextTimestamp() - are increased according to the
--- Begin Message ---
On 29/08/2019 11:23, Ross Finlayson wrote:
Yes. More precisely, the presentation times need to be aligned with
the server’s *wall clock* time - i.e., the time that you’d get by
calling “gettimeofday()”. This ensures that clients will be able to
properly use the RTCP “SR”
> This way things seems to work well, but I would like to be sure, so my
> question is: the fact that the frame presentation times are
> aligned/normalized with respect to server clock is an essential prerequisite
> in order to grant the correct timing for the sending of RTP packets?
Yes. More
I got it!
thank you for your reply
--
袁旺柳
上海游民网络科技有限公司/全体3K/技术中心
-- Original --
From: "Ross Finlayson";
Date: Thu, Aug 29, 2019 03:28 PM
To: "LIVE555 Streaming Media - development & use";
Subject: Re: [Live-devel] i want W
> On Aug 29, 2019, at 8:39 AM, 袁旺柳 wrote:
>
> I am interested in WEBRTC and need to using this technology in my product
> in live555
> can you send me the code ?
Sorry, but the WebRTC gateway was an experiment that is no longer being
developed. I found that it was too difficult to get it
I am interested in WEBRTC and need to using this technology in my product in
live555
can you send me the code ?
thank you
--
袁旺柳
上海游民网络科技有限公司/全体3K/技术中心___
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