<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;">Darwin is a streaming server. It is fed with RTP Audio/Video streams and is able to dispatch them to rtsp clients.<br>So yes, you can stream from dvb source to Darwin, and then use an rtsp client to ask Darwin for the stream.<br><br>To interact with Darwin, you have to write your own darwin plugin. <br>Have a look at the QTSS documentation http://www.apple.com/quicktime/streamingserver/<br><br>A better solution would be to write a gstreamer plugin acting as a RTSP server ...<br><br>Aurelien<br><br><div style="font-family: arial,helvetica,sans-serif; font-size: 13px;">----- Message d'origine ----<br>De : Rémi BUISSON <remi.buisson@viotech.net><br>À : Discussion of the development of GStreamer
<gstreamer-devel@lists.sourceforge.net><br>Envoyé le : Jeudi, 7 Août 2008, 17h14mn 30s<br>Objet : Re: [gst-devel] Re : Help with RTP<br><br>Thanks.<br><br>With this server, can I stream from dvb source for instance ?<br><br>In fact, I am in an internship and the program I have to write must <br>exchange SIP message for the commands. So my question is: can I interact <br>easily with Darwin from my C program ? with a plugin ?<br><br>Thanks again !<br><br>Rémi<br><br>Aurelien Grimaud wrote:<br>> Hi, I think you need a server here to handle the negotiations (sdp <br>> exchange).<br>> Then, you can use RTCP sent by client to detect network congestion.<br>><br>> Darwin streaming server is a rtsp server and may be a good start to <br>> handle rtsp.<br>> It also supports easy plugin writting, in which you can alter your <br>> stream the way you want, without re-writting a rtsp stack from scratch.<br>><br>>
Aurelien<br>><br>> ----- Message d'origine ----<br>> De : Rémi BUISSON <<a ymailto="mailto:remi.buisson@viotech.net" href="mailto:remi.buisson@viotech.net">remi.buisson@viotech.net</a>><br>> À : Daiane Angolini <<a ymailto="mailto:daiane.angolini@freescale.com" href="mailto:daiane.angolini@freescale.com">daiane.angolini@freescale.com</a>><br>> Cc : Discussion of the development of GStreamer <br>> <<a ymailto="mailto:gstreamer-devel@lists.sourceforge.net" href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>><br>> Envoyé le : Jeudi, 7 Août 2008, 8h37mn 59s<br>> Objet : Re: [gst-devel] Help with RTP<br>><br>> Hi and thanks for your reply !<br>><br>> OK I will try today and I will tell you if it works for me.<br>><br>> My goal is to write a streaming server which can transcode, transrate,<br>> ... a stream and using a file for clients may be some kind
of a<br>> do-it-yourself solution.<br>><br>> For instance, the client send a message to the server for starting the<br>> streaming of a file. Next, imagine the network is overloaded. Then, the<br>> client will ask to the server to decrease the bitrate of the stream.<br>><br>> The client may be VLC, MPlayer, ...<br>><br>> Have you any idea to do it ?<br>><br>> I think it's possible because the streams are standardized.<br>> Moreover, I tried to read a video streamed by flumotion over HTTP and it<br>> works perfectly with VLC.<br>><br>> Thanks again.<br>><br>> Daiane Angolini wrote:<br>> > Hello Remi!<br>> ><br>> > If you want to see streaming using VLC you can use a SDP file.<br>> ><br>> > Try this:<br>> ><br>> ><br>> > *v=0*<br>> > *o=- 37 614155991 IN IP4 127.0.0.0*<br>> > *s=QuickTime*<br>> > *t=0 0*<br>> >
*a=range:npt=now-*<br>> > *m=audio 5432 RTP/AVP 0*<br>> > *c=IN IP4 10.29.241.6*<br>> > *b=AS:63*<br>> > *m=video 5434 RTP/AVP 96*<br>> > *c=IN IP4 10.29.241.6*<br>> > *a=rtpmap:96 H263-2000/90000*<br>> > *a=fmtp:96 *<br>> > *a=cliprect:0,0,144,176*<br>> > *a=framesize:96 176-144*<br>> ><br>> > In this file you have both audio and video.<br>> > But if you need a simple one, you can try this:<br>> ><br>> > *v=0*<br>> > *m=video 5434 RTP/AVP 96*<br>> > *c=IN IP4 10.29.240.186*<br>> > *a=rtpmap:96 MP4V-ES/90000 *<br>> > *a=fmtp:96<br>> > <br>> config=000001b002000001b59113000001000000012000c488800f50584121443f;profile-level-id=2*<br>> ><br>> > The *config* and *profile-level-id* you can get from the return value<br>> > of *gst-lauch -v* command. In the caps property.<br>> ><br>> > Ah!<br>> >
To use VLC with SDP file use:<br>> ><br>> > vlc -vvv file.sdp<br>> ><br>> > I hope it works<br>> ><br>> ><br>> > Daiane<br>> ><br>> ><br>> ><br>> ><br>> > On Wed, 2008-08-06 at 16:21 +0200, Rémi BUISSON wrote:<br>> >> Hi everyone,<br>> >><br>> >> I have some troubles streaming video file over the network.<br>> >><br>> >> First of all, I tired this :<br>> >><br>> >> server:<br>> >> gst-launch filesrc location=../partage/Videos/superman_originale.avi !<br>> >> decodebin ! x264enc ! video/x-h264 ! rtph264pay pt=96 ! udpsink<br>> >> host=192.168.1.2 port=5000 sync=false<br>> >><br>> >> client:<br>> >> gst-launch udpsrc port=5000 ! rtph264depay ! decodebin ! xvimagesink<br>> >><br>> >> It works but I can't read it with vlc : vlc udp://@:5000 : is
it <br>> normal ?<br>> >><br>> >> Now I would like to do the same thing with an RTP pipeline :<br>> >><br>> >> server:<br>> >> gst-launch filesrc location=../partage/Videos/superman_originale.avi !<br>> >> decodebin ! x264enc ! rtph264pay ! rtpbin localport=5001<br>> >> destinations=127.0.0.1:5000<br>> >><br>> >> client:<br>> >> gst-launch udpsrc port=5000 ! rtph264depay ! decodebin ! <br>> xvimagesink --><br>> >> this works<br>> >> gst-launch rtpbin localport=5000 ! rtph264depay ! decodebin !<br>> >> xvimagesink --> this doesn't work<br>> >><br>> >> How can I receive my stream ? with vlc ?<br>> >><br>> >> Thanks in advance !<br>> >><br>> >> <br>> -------------------------------------------------------------------------<br>> >> This SF.Net email is sponsored by
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