Hi Ross, Thanks for the concern, now I am able to stream h.264 with correct SDP. I've corrected the SDP. Now it's streaming.
Regards, Soumya LGSI -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, July 12, 2008 4:25 AM To: [EMAIL PROTECTED] Subject: live-devel Digest, Vol 57, Issue 13 Send live-devel mailing list submissions to [email protected] To subscribe or unsubscribe via the World Wide Web, visit http://lists.live555.com/mailman/listinfo/live-devel or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of live-devel digest..." Today's Topics: 1. Re: H264 SDP (Ross Finlayson) 2. live555 Test Programs (Hannah) 3. Re: H264 SDP (Jerry Johns) 4. Re: live555 Test Programs (Ross Finlayson) 5. Re: H264 SDP (Ross Finlayson) 6. Creating a RTSP stream from DVB-T hardware card (Ryan Walklin) ---------------------------------------------------------------------- Message: 1 Date: Thu, 10 Jul 2008 23:12:53 -0700 From: Ross Finlayson <[EMAIL PROTECTED]> Subject: Re: [Live-devel] H264 SDP To: LIVE555 Streaming Media - development & use <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"; Format="flowed" >I have attached our sdp and NAL frame (contains NAL units 7 and 8) >alongwith the mail. We may be missing some needed information in sdp >or we might be communicating it in the wrong format or I might be >totally mistakenJ. Could you tell us the correct sdp fmtp line to >generate No I can't, because only you know the details of your H.264 stream. Are you sure you are encoding the sps and pps NAL units correctly using Base64 (for the "sprop_parameter_sets_str" parameter)? I suggest that you go through VLC's LIVE555 interface code (in "modules/demux/live555.cpp") to make sure that the SDP fmtp string is decoded into the correct NAL unit data (see the function "parseH264ConfigStr()"). If you're still having problems, then you may need to make your RTSP stream publically available, and post to the VLC mailing list, asking them why VLC cannot play it. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.live555.com/pipermail/live-devel/attachments/20080710/52c0de0b /attachment-0001.html> ------------------------------ Message: 2 Date: Fri, 11 Jul 2008 20:23:38 +0800 From: Hannah <[EMAIL PROTECTED]> Subject: [Live-devel] live555 Test Programs To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Hello, Could I ask you this question? I compiled the source code already and now testing the test programs. I try the MP3 Audio test programs as first example. + For the testMP3Streamer, the output receiver is "Unable to open file "test.mp3" as a MP3 file source" + For testMP3Receiver, output is "Beginning receiving multicast stream..." The other test programs have the same output, I can't stream or play any file. What's the problem and how to solve? *Secondly*, I run the testOnDemandRTSPServer, and it returns me the URL links of the audio/video test files, for example : "mpeg4ESVideoTest" stream, from the file "test.m4e" Play this stream using the URL "rtsp://10.10.10.103:8554/mpeg4ESVideoTest" "mpeg1or2AudioVideoTest" stream, from the file "test.mpg" Play this stream using the URL "rtsp:// 10.10.10.103:8554/mpeg1or2AudioVideoTest" Is it the URL link to use with openRTSP client, that I ask below? *Thirdly*, I run the RTSP client, by the command : *./openRTSP,* and I see the usage instructions : Usage : ./openRTSP [-p <startPortNum>] [-r|-q|-4|-i] [-a|-v] [-V] [-d <duration>] [-D <max-inter-packet-gap-time> [-c] [-S <offset>] [-n] [-O] [-t|-T <http-port>] [-u <username> <password>] [-s <initial-seek-time>] [-z <scale>] [-w <width> -h <height>] [-f <frames-per-second>] [-y] [-H] [-Q [<measurement-interval>]] [-F <filename-prefix>] [-b <file-sink-buffer-size>] [-B <input-socket-buffer-size>] [-I <input-interface-ip-address>] [-m] <url> *(or ./openRTSP -o [-V] <url>) * I have tried to use the bold part, for instance *./openRTSP -o [-V] rtsp://10.10.10.103:8554/mpeg1or2AudioVideoTest *But I received nothing. The terminal screen return the usage instructions again. Did I do wrongly? Or do I misunderstand any part? Is there any solutions or how to solve the situation? Thanks a lot for your help on this. Hannah -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.live555.com/pipermail/live-devel/attachments/20080711/e3274f5f /attachment-0001.html> ------------------------------ Message: 3 Date: Fri, 11 Jul 2008 06:44:31 -0700 From: "Jerry Johns" <[EMAIL PROTECTED]> Subject: Re: [Live-devel] H264 SDP To: <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Are you sure you're creating your SDP properly? We've managed to get our streams working with VLC/Quicktime using proper SDP params Is your profile-level-id correct? Check the RFC spec on exact details As for Base64, try using this site to check your vals: http://www.paulschou.com/tools/xlate/ is your format type (97) correct? We use 96 Hope it helps, Jerry Johns Design Engineer Nuvation Research Corp - Canada Tel: (519) 746-2304 ext. 225 www.nuvation.com <http://www.nuvation.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.live555.com/pipermail/live-devel/attachments/20080711/0197ef2f /attachment-0001.html> ------------------------------ Message: 4 Date: Fri, 11 Jul 2008 07:12:17 -0700 From: Ross Finlayson <[EMAIL PROTECTED]> Subject: Re: [Live-devel] live555 Test Programs To: LIVE555 Streaming Media - development & use <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"; Format="flowed" >+ For the testMP3Streamer, the output receiver is "Unable to open >file "test.mp3" as a MP3 file source" Do you have a MP3 file named "test.mp3" in the same directory? >+ For testMP3Receiver, output is "Beginning receiving multicast stream..." Note that "testMP3Receiver" outputs its received (MP3) data to 'stdout'. (If you don't understand what 'stdout' means, then this software is probably not for you.) >Secondly, I run the testOnDemandRTSPServer, and it returns me the >URL links of the audio/video test files, for example : > >"mpeg4ESVideoTest" stream, from the file "test.m4e" >Play this stream using the URL >"rtsp://<http://10.10.10.103:8554/mpeg4ESVideoTest>10.10.10.103:8554/mpeg4E SVideoTest" > >"mpeg1or2AudioVideoTest" stream, from the file "test.mpg" >Play this stream using the URL >"rtsp://<http://10.10.10.103:8554/mpeg1or2AudioVideoTest>10.10.10.103:8554/ mpeg1or2AudioVideoTest" > >Is it the URL link to use with openRTSP client, that I ask below? Yes, but you need to have the appropriate file - e.g., a MPEG-1 or 2 Program Stream file named "test.mpg" - in the same directory. > >Thirdly, I run the RTSP client, by the command : ./openRTSP, and I >see the usage instructions : > >Usage : ./openRTSP [-p <startPortNum>] [-r|-q|-4|-i] [-a|-v] [-V] >[-d <duration>] [-D <max-inter-packet-gap-time> [-c] [-S <offset>] >[-n] [-O] [-t|-T <http-port>] [-u <username> <password>] [-s ><initial-seek-time>] [-z <scale>] [-w <width> -h <height>] [-f ><frames-per-second>] [-y] [-H] [-Q [<measurement-interval>]] [-F ><filename-prefix>] [-b <file-sink-buffer-size>] [-B ><input-socket-buffer-size>] [-I <input-interface-ip-address>] [-m] ><url> (or ./openRTSP -o [-V] <url>) > > I have tried to use the bold part, for instance ./openRTSP -o [-V] >rtsp://<http://10.10.10.103:8554/mpeg1or2AudioVideoTest>10.10.10.103:8554/m peg1or2AudioVideoTest No, just run ./openRTSP rtsp://<http://10.10.10.103:8554/mpeg1or2AudioVideoTest>10.10.10.103:8554/mp eg1or2AudioVideoTest See the "openRTSP" instructions <http://www.live555.com/openRTSP/> to understand the command-line options. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.live555.com/pipermail/live-devel/attachments/20080711/60079f15 /attachment-0001.html> ------------------------------ Message: 5 Date: Fri, 11 Jul 2008 07:14:30 -0700 From: Ross Finlayson <[EMAIL PROTECTED]> Subject: Re: [Live-devel] H264 SDP To: LIVE555 Streaming Media - development & use <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"; Format="flowed" >is your format type (97) correct? >We use 96 Either is correct (provided that it doesn't clash with another substream (e.g., audio) that you're also sending. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.live555.com/pipermail/live-devel/attachments/20080711/6369416e /attachment-0001.html> ------------------------------ Message: 6 Date: Sat, 12 Jul 2008 10:52:11 +1200 From: Ryan Walklin <[EMAIL PROTECTED]> Subject: [Live-devel] Creating a RTSP stream from DVB-T hardware card To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"; Format="flowed"; DelSp="yes" Hi, I'm using an Elgato DVB-T USB tuner (EyeTV for DTT) in combination with the livemedia library to stream a MPEG2-TS stream containing h264 video and AAC audio (LATM encapsulation) via RTSP over my LAN. I've based my streaming server on the testMPEG2TransportStreamer sample. The EyeTV plugin SDK provides a callback which is activated when ~100 or so packets have arrived, and sends the raw TS data via a pipe to the server code, running in another thread. I've modified the code to read from the other end of the pipe I created. I've also removed the MPEG2Framer from the chain as I presumed the packet stream was already in this format. This done, I'm able to launch VLC and see the stream for 1-2 seconds, however VLC proceeds to crash with the console errors: MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (48). 1388 bytes of trailing data will be dropped! MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (112). 1324 bytes of trailing data will be dropped! MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (60). 1376 bytes of trailing data will be dropped! MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (8). 1428 bytes of trailing data will be dropped! From my reading of the mailing list archives, I understand that somewhere along the encoding chain (Presumably in ByteStreamFileSource (MultiFramedRTPSource) a buffer is being over-run, but I cannot fathom where. All the buffers seem to default to ~50-60kb, however from those errors it seems to be under 100 bytes most of the time. I note the sum of the streamed+dropped packets is 1440, so presumably this has something to do with network framing. I've been playing with this all week and ended up more and more confused reading the source. I'd apprecitate it if anyone is able to shed light. I've attached my server thread source for your perusal. (createRTSPserver is essentially main() from testtMPEG2TransportStreamer.cpp with my tweaks). Regards, Ryan Walklin -------------- next part -------------- A non-text attachment was scrubbed... Name: MPEG2RTSPServer.cpp Type: application/octet-stream Size: 6079 bytes Desc: not available URL: <http://lists.live555.com/pipermail/live-devel/attachments/20080712/0fac44d3 /attachment.obj> -------------- next part -------------- - stream structure - using pipe to copy packets - raw off card - how to relay mpeg2 ts as live to rtsp - buffer size ------------------------------ _______________________________________________ live-devel mailing list [email protected] http://lists.live555.com/mailman/listinfo/live-devel End of live-devel Digest, Vol 57, Issue 13 ****************************************** _______________________________________________ live-devel mailing list [email protected] http://lists.live555.com/mailman/listinfo/live-devel
