From d.fischer at ids-imaging.de Mon May 2 00:08:37 2011 From: d.fischer at ids-imaging.de (Fischer Daniel) Date: Mon, 2 May 2011 09:08:37 +0200 Subject: [Live-devel] How to bypass streamlimit causes by EventTriggerIDs Message-ID: <4CB3F3483954BD4ABD0B4DBB35C7E33801D663AC@exchsrv.idszentral.local> Hello, I am using live555 to stream several network cameras. For that I generate one RTSP-Server and for every camera a subsession on this server with a new URL. To signal the TaskScheduler, that there is a new frame for a stream, I use a EventTriggerID. Every stream has his own EventTriggerID. Now I got the problem, that the EventTriggerID is generated by a bitmask (0x80000000), and the line ?m_EventTriggerId = envir().taskScheduler().createEventTrigger(deliverFrame0);? generates only 32 EventTriggerID?s, so that I have a maximum of 32 stream receivers. Now my question: Is it possible to solve that problem without generating more RTSP-Server with different TaskScheduler on different ports? Can you make me a suggestion how I bypass these limit of maximum stream receivers? Thank you for your efforts With greet Daniel Fischer Viele Gr??e / Best regards - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Dipl.-Inf. (FH) Daniel Fischer iGuard - Development IDS Imaging Development Systems GmbH Dimbacher Strasse 6-8 ? D-74182 Obersulm Handelsregister: Stuttgart HRB 106225 Gesch?ftsf?hrer: J?rgen Hartmann, Armin Vogt, Torsten Wiesinger T : +49 (0)7134 / 961 96-0 F : +49 (0)7134 / 961 96-99 E : d.fischer at ids-imaging.de Web: www.ids-imaging.com - www.iguard.de - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - P Please consider the environment before printing this e-mail -------------- next part -------------- An HTML attachment was scrubbed... URL: From c_hess at null.net Mon May 2 12:42:11 2011 From: c_hess at null.net (Chuck Hess) Date: Mon, 02 May 2011 19:42:11 +0000 Subject: [Live-devel] wis-streamer and the -d option for DarwinInjector Message-ID: <20110502194211.287260@gmx.com> I was looking at the wis-streamer source and I'm intrigued by the -d option, but I can't seem to get it to work. I can use wis-streamer as-is and view the live stream from a remote quicktime player, but not a cell phone. I'm guessing that this is due to the lack of rtsp over http support. I'm able to use the test utility in the live project to stream and view a test.m4e file via DarwinInjector. When I use wis-streamer with the -d option, I get a test.sdp file in my movies directory, but both quicktime, cell phones, and vlc say that the file is not found after buffering. Any ideas? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From arp at softtelecom.es Tue May 3 03:14:00 2011 From: arp at softtelecom.es (=?ISO-8859-1?Q?=C1lvaro_Rodr=EDguez_P=E9rez?=) Date: Tue, 03 May 2011 12:14:00 +0200 Subject: [Live-devel] RTSP decoding In-Reply-To: References: Message-ID: <4DBFD568.5010409@softtelecom.es> Hello, I?m trying to receive a RTSP Stream with live555 and decode the stream with ffmpeg. I?ve seen on some messages that there is information about this issue in the FAQ but I can?t find the correct reference. ?Can someone help me? Thank you very much. ?lvaro. -- ?lvaro Rodr?guez SoftTelecom www.softtelecom.es UPM Campus de Montegancedo PARQUE CIENT?FICO Y TECNOL?GICO UPM 28223 POZUELO MADRID From bruno.abreu at livingdata.pt Tue May 3 03:21:30 2011 From: bruno.abreu at livingdata.pt (Bruno Abreu) Date: Tue, 3 May 2011 11:21:30 +0100 Subject: [Live-devel] RTSPServer crash with simultaneous identical requests Message-ID: <20110503094608.M21495@livingdata.pt> Hello, we've been using the RTSPServer to stream both live and stored video. Because of our client's specific needs, particularly when asking for stored videos, sometimes we get identical requests from 2 clients simultaneously (i.e. a few milliseconds apart) and, whenever that happens, the server crashes. Because we've extended the RTSPServer (as in the DynamicRTSPServer of the Media Server app) to parse the clients' requests, we went back to test some basic things to make sure that the error was not introduced by our enhancements. So we ran testOnDemandRTSPServer with some of the videos you supply for tests and created a simple shell script with the following lines for the clients: #!/bin/sh ./openRTSP -F "0-" rtsp://127.0.0.1:8554/h264ESVideoTest > openRTSP-0.log 2>&1 & ./openRTSP -F "1-" rtsp://127.0.0.1:8554/h264ESVideoTest > openRTSP-1.log 2>&1 & This should launch (almost simultaneously) two instances of openRTSP that ask for the same video segment from the server. And we got similar results to our app. Either the server crashes with a segfault or it delivers the video correctly to one of the clients but it delivers a SDP description containing errors to the 2nd (or so openRTSP complains). From that point on every new request from a client receives a corrupted SDP description. Below is the stack trace from one of the server crashes: gdb ./testOnDemandRTSPServer core ... Core was generated by `./testOnDemandRTSPServer'. Program terminated with signal 11, Segmentation fault. #0 0xb73bc000 in ?? () (gdb) bt #0 0xb73bc000 in ?? () #1 0x08069969 in MultiFramedRTPSink::ourHandleClosure (clientData=0x89124d8) at MultiFramedRTPSink.cpp:413 #2 0x0804f9a8 in H264VideoFileServerMediaSubsession::checkForAuxSDPLine1 (this=0x8905bf8) at H264VideoFileServerMediaSubsession.cpp:61 #3 0x0804f9f9 in checkForAuxSDPLine (clientData=0x8905bf8) at H264VideoFileServerMediaSubsession.cpp:57 #4 0x08077f5f in AlarmHandler::handleTimeout (this=0x8912780) at BasicTaskScheduler0.cpp:34 #5 0x08076a72 in DelayQueue::handleAlarm (this=0x32482036) at DelayQueue.cpp:180 #6 0x080760e5 in BasicTaskScheduler::SingleStep (this=0x8905008, maxDelayTime=0) at BasicTaskScheduler.cpp:189 #7 0x08077617 in BasicTaskScheduler0::doEventLoop (this=0x8905008, watchVariable=0x0) at BasicTaskScheduler0.cpp:80 #8 0x0804a011 in main (argc=, argv=) at testOnDemandRTSPServer.cpp:264 And here is the part of the log from one of the openRTSP clients: Opened URL "rtsp://127.0.0.1:8554/h264ESVideoTest", returning a SDP description: v=0^M o=- 1304352757340991 1 IN IP4 10.10.9.211^M s=Session streamed by "testOnDemandRTSPServer"^M i=h264ESVideoTest^M t=0 0^M a=tool:LIVE555 Streaming Media v2011.03.14^M a=type:broadcast^M a=control:*^M a=range:npt=0-^M a=x-qt-text-nam:Session streamed by "testOnDemandRTSPServer"^M a=x-qt-text-inf:h264ESVideoTest^M m=video 0 RTP/AVP 96^M c=IN IP4 0.0.0.0^M b=AS:500^M a=rtpmap:96 H264/90000^M ?f?^H^Xd?^HliveMedia33a=control:track1^M Failed to create a MediaSession object from the SDP description: Invalid SDP line: ?f?^H^Xd?^HliveMedia33a=control:track1^M I'd like to point out that we're using the latest version of live555 and that, for these tests, not a single line of code as been touched. We would like to know if this is a bug that will get fixed in the future or, simply, something that is not allowed. Any help would be appreciated Thank you Bruno Abreu -- Living Data - Sistemas de Informa??o e Apoio ? Decis?o, Lda. Rua Lu?s de Cam?es, N? 133, 1? B Phone: +351 213622163 1300-357 LISBOA Fax: +351 213622165 Portugal URL: www.livingdata.pt From ferry at bertin.fr Tue May 3 04:40:34 2011 From: ferry at bertin.fr (Guillaume Ferry) Date: Tue, 03 May 2011 13:40:34 +0200 Subject: [Live-devel] Questions about MPEG2 transport streams Message-ID: <4DBFE9B2.5000303@bertin.fr> Hi Ross, I'm currently using openRTSP to receive MPEG2 transport streams. I'd rather have N subsessions for N streams, but I'm not responsible for these servers, so I must stick with TS packets. And here are my two questions : * Is it possible, in a RTSP client, to select a specific stream among the others, and only receive this one ? I think it's not possible, but maybe you have some code to so in liveMedia APIs ? * Otherwise, are there some methods available, always on client' side, to parse a TS packet, list its inner streams, and extract one ? I think I could do the job with FFmpeg, but it would be lighter / simpler if I could avoid it ! Thanks in advance for your insights, Best regards, Guillaume. -- Guillaume FERRY Bertin Technologies D?partement Bertin Conseil Activit? Traitement de l'Information et du Contenu /T?l/ 01.39.30.62.09 /Fax/ 01.39.30.62.45 /Mail/ ferry at bertin.fr /Web/ www.bertin.fr -------------- next part -------------- An HTML attachment was scrubbed... URL: From felix at embedded-sol.com Tue May 10 23:27:54 2011 From: felix at embedded-sol.com (Felix Radensky) Date: Wed, 11 May 2011 09:27:54 +0300 Subject: [Live-devel] Strange problem with h264 streaming Message-ID: <4DCA2C6A.90301@embedded-sol.com> Hi, I have a strange problem with latest version of live555. I'm streaming h264 via RTSP. The video is produced by hardware encoder. In my code I use H264VideoStreamDiscreteFramer class and event triggers to notify the streamer thread about the availability of new h264 buffer. I use VLC as a viewer. The streaming can run four hours without problems, but at some point streamer starts sending tons of RTCP Receiver Report messages, which hang the CPU and the viewer completely. Any ideas what can cause such behaviour will be greatly appreciated. Thanks a lot. Felix. From Bruno.Basilio at brisa.pt Wed May 11 06:35:03 2011 From: Bruno.Basilio at brisa.pt (Bruno Filipe Basilio) Date: Wed, 11 May 2011 13:35:03 +0000 Subject: [Live-devel] Strange problem with h264 streaming Message-ID: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA861904376B@SRVBRIEXC012.brisa.pt> Hi Felix, I have a similar problem. After 3 days playing I couldn't access the RTP/RTCP server PC anymore. Pinging didn't work either. Only after a reboot everything is ok. A few days later the same happened. And now I discovered something even stranger: I sniffed the network traffic going to and going from the server to see if I was missing anything: Analyzing the traffic I saw a stream coming from the server that was 35Mbps analyzing it some more it is a RTCP stream -> this doesn't seems OK (a rtcp stream that is 20 times bigger than the actual video stream) Seeing this I did some more experiments: 1) first I power off power on the server and put it in a separate lan. After sniffing the network I found 4 multicast streams coming out of it: - 232.29.51.215:18888 -> rtp video stream from 2Mbps - 232.29.51.215:18889 -> rtcp sender reports (a few of them) - 232.218.63.39:18890 -> rtp video stream from about 200kbps - 232.218.63.39:18891 -> rtcp sender reports (just a few of them) 2) A placed a 2 decoders in the network and ask the rtsp multicast stream for both of them I received the stream and could decode it (the stream was 232.29.51.215:18888) But the rtcp stream 232.29.51.215:18889 suddenly became enormous about 35 Mbps I think this is a bug and this is causing eventually the strange behavior I mention earlier (not reacting on ping anymore) Best regards, Bruno Basilio Brisa Inova??o e Tecnologia, S.A. -------------------------------------------------------------------------------- Declara??o: A informa??o contida nesta mensagem, e os ficheiros anexos, ? privilegiada e confidencial, destinando-se exclusivamente ao(s) destinat?rio(s).Se n?o ? o destinat?rio (ou o respons?vel pela sua entrega ao destinat?rio) e recebeu a mesma por engano, fica notificado que ? estritamente proibido reproduzir, guardar ou distribuir toda ou qualquer parte desta mensagem e ficheiros anexos.Por favor reencaminhe a mensagem para o respons?vel pelo seu envio ou contacte-nos por telefone e elimine a mensagem e ficheiros anexos do seu computador,sem os reproduzir. Disclaimer: The information contained in this message, and any files attached, is privileged and confidential, and intended exclusively for the included addresses.If you are not the intended recipient (or the person responsible for delivering to the intended recipient) and received this message by mistake, be aware that copy, storage, distribution or any other use of all or part of this message and the files attached is strictly prohibited. Please notify the sender by reply e-mail or contact us by telephone and delete this message and the files attached, without retaining a copy. -------------------------------------------------------------------------------- From Bruno.Basilio at brisa.pt Wed May 11 09:30:09 2011 From: Bruno.Basilio at brisa.pt (Bruno Filipe Basilio) Date: Wed, 11 May 2011 16:30:09 +0000 Subject: [Live-devel] Strange problem with h264 streaming In-Reply-To: <4DCAB27A.2070302@embedded-sol.com> References: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA861904376B@SRVBRIEXC012.brisa.pt> <4DCAB27A.2070302@embedded-sol.com> Message-ID: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA86190437D5@SRVBRIEXC012.brisa.pt> >> I'm streaming h264 via RTSP. The video is produced by >> hardware encoder. In my code I use H264VideoStreamDiscreteFramer >> class and event triggers to notify the streamer thread about >> the availability of new h264 buffer. > What is your video source ? Is it h264 or something else ? My video source is also a live encoder with h264. But instead of using event triggers to notify new frames we are waiting for it using TaskScheduler::turnOnBackgroundReadHandling(). Bruno -------------------------------------------------------------------------------- Declara??o: A informa??o contida nesta mensagem, e os ficheiros anexos, ? privilegiada e confidencial, destinando-se exclusivamente ao(s) destinat?rio(s).Se n?o ? o destinat?rio (ou o respons?vel pela sua entrega ao destinat?rio) e recebeu a mesma por engano, fica notificado que ? estritamente proibido reproduzir, guardar ou distribuir toda ou qualquer parte desta mensagem e ficheiros anexos.Por favor reencaminhe a mensagem para o respons?vel pelo seu envio ou contacte-nos por telefone e elimine a mensagem e ficheiros anexos do seu computador,sem os reproduzir. Disclaimer: The information contained in this message, and any files attached, is privileged and confidential, and intended exclusively for the included addresses.If you are not the intended recipient (or the person responsible for delivering to the intended recipient) and received this message by mistake, be aware that copy, storage, distribution or any other use of all or part of this message and the files attached is strictly prohibited. Please notify the sender by reply e-mail or contact us by telephone and delete this message and the files attached, without retaining a copy. -------------------------------------------------------------------------------- From felix at embedded-sol.com Wed May 11 08:59:54 2011 From: felix at embedded-sol.com (Felix Radensky) Date: Wed, 11 May 2011 18:59:54 +0300 Subject: [Live-devel] Strange problem with h264 streaming In-Reply-To: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA861904376B@SRVBRIEXC012.brisa.pt> References: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA861904376B@SRVBRIEXC012.brisa.pt> Message-ID: <4DCAB27A.2070302@embedded-sol.com> Hi Bruno, On 05/11/2011 04:35 PM, Bruno Filipe Basilio wrote: > Hi Felix, > > I have a similar problem. > After 3 days playing I couldn't access the RTP/RTCP server PC anymore. Pinging didn't work either. Only after a reboot everything is ok. > A few days later the same happened. > > And now I discovered something even stranger: > I sniffed the network traffic going to and going from the server to see if I was missing anything: > > Analyzing the traffic I saw a stream coming from the server that was 35Mbps analyzing it some more it is a RTCP stream -> this doesn't seems OK (a rtcp stream that is 20 times bigger than the actual video stream) > > Seeing this I did some more experiments: > 1) first I power off power on the server and put it in a separate lan. > After sniffing the network I found 4 multicast streams coming out of it: > - 232.29.51.215:18888 -> rtp video stream from 2Mbps > - 232.29.51.215:18889 -> rtcp sender reports (a few of them) > - 232.218.63.39:18890 -> rtp video stream from about 200kbps > - 232.218.63.39:18891 -> rtcp sender reports (just a few of them) > 2) A placed a 2 decoders in the network and ask the rtsp multicast stream for both of them > I received the stream and could decode it (the stream was 232.29.51.215:18888) > But the rtcp stream 232.29.51.215:18889 suddenly became enormous about 35 Mbps > I think this is a bug and this is causing eventually the strange behavior I mention earlier (not reacting on ping anymore) > > What is your video source ? Is it h264 or something else ? Felix. From felix at embedded-sol.com Wed May 11 10:15:15 2011 From: felix at embedded-sol.com (Felix Radensky) Date: Wed, 11 May 2011 20:15:15 +0300 Subject: [Live-devel] Strange problem with h264 streaming In-Reply-To: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA86190437D5@SRVBRIEXC012.brisa.pt> References: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA861904376B@SRVBRIEXC012.brisa.pt> <4DCAB27A.2070302@embedded-sol.com> <3EC6EEDBCBD4AC4A9FBE5F1F5EEA86190437D5@SRVBRIEXC012.brisa.pt> Message-ID: <4DCAC423.7010408@embedded-sol.com> Hi Bruno, On 05/11/2011 07:30 PM, Bruno Filipe Basilio wrote: >>> I'm streaming h264 via RTSP. The video is produced by >>> hardware encoder. In my code I use H264VideoStreamDiscreteFramer >>> class and event triggers to notify the streamer thread about >>> the availability of new h264 buffer. >> What is your video source ? Is it h264 or something else ? > My video source is also a live encoder with h264. > But instead of using event triggers to notify new frames we are waiting for it using TaskScheduler::turnOnBackgroundReadHandling(). Thanks. BTW, in my case the problem is immediately reproducible with smplayer. Felix. From thomas.maier at uni-kassel.de Wed May 11 04:04:50 2011 From: thomas.maier at uni-kassel.de (Thomas Maier) Date: Wed, 11 May 2011 13:04:50 +0200 Subject: [Live-devel] Problems on iPhone when on cell network Message-ID: <6B2A0D7D-39ED-44D9-80F1-215D73DC5ECF@uni-kassel.de> Hi gurus, I am using the LIVE555 Streaming Media libs on the iPhone. As long as I am on WiFi, everything works fine. However, as soon as I try to use the cell network (2G or 3G), I get the following error: Unable to determine our source address: This computer has an invalid IP address: 0x0 I grepped the sources and found "This computer has an invalid IP address" in groupsock/GroupsockHelper.cpp. Alas, my network hacking skills are zero. I see the comment "(This code, like many others, won't handle IPv6)", so maybe (wildly guessing here) some info about my carrier is relevant? It is the German Telekom in, surprise, Germany. If it's of any use, when I start my app when WiFi is active, get some RTP data, stop, switch to cell in the iPhone settings, and then try to get data, I do not get the above error but a different one: RTSP 'OPTIONS' request failed: Failed to find network address for "my.camera.url" This happens in continueAfterOPTIONS() (my code is quite a copy of openRTSP and works fine on WiFi). Has anybody experienced that problem? Any pointers to what I should read or what I could do? I am using the sources as of 2010-10-28. I have read http://live555.com/liveMedia/public/changelog.txt but could not find a relevant change that might fix the problem. Should I upgrade anyway? I am on Xcode 3.2.6 with iOS SDK 4.3. gcc --version says i686-apple-darwin10-gcc-4.2.1 (GCC) 4.2.1 (Apple Inc. build 5666) (dot 3) I am not sure what other info you need but I will happily provide it. Thanks in advance, Thomas From Autuori.Gianluigi.Wintime at ansaldobreda.it Wed May 11 06:53:41 2011 From: Autuori.Gianluigi.Wintime at ansaldobreda.it (Autuori Gianluigi) Date: Wed, 11 May 2011 15:53:41 +0200 Subject: [Live-devel] how to capture H264 stream and save it in a avi file Message-ID: Hello, I'm trying to capture a video stream from Axis IP camera with openRTSP. I get a correct file if I use: openRTSP -d 20 -4 -f 3 -w 640 -h 480 -b 400000 "rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3" >video.avi but I try this: openRTSP -d 20 -f 3 -w 640 -h 480 -b 400000 "rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3" I obtain this file: video-H264-1 but I can't open it. What is wrong? I read this post: http://lists.live555.com/pipermail/live-devel/2007-January/005886.html is it the same problem? thanks Gianluigi -------------------------------------------------------------------------------- Questo messaggio e-mail e ogni documento ad esso eventualmente allegato puo' avere carattere riservato ed essere tutelato da segreto. Esso,comunque, e' ad esclusivo utilizzo del destinatario in indirizzo. Qualora non foste il destinatario del messaggio vi preghiamo di volerci avvertire immediatamente per e-mail o telefono e di cancellare il presente messaggio e ogni eventuale allegato dal vostro sistema. E' vietata la duplicazione o l'utilizzo per qualunque fine del messaggio e di ogni allegato, nonche' la loro divulgazione, distribuzione o inoltro a terzi senza l'espressa autorizzazione del mittente. In ragione del mezzo di trasmissione utilizzato, il mittente non assume alcuna responsabilita' sulla segretezza/riservatezza delle informazioni contenute nel messaggio e nei relativi allegati. This e-mail and any file transmitted with it may contain material that is confidential, privileged and/or attorney work product for the sole use of the intended recipient. If you are not the intended recipient of this e-mail, please do not read it, notify us immediately by e-mail or by telephone and then delete this message and any file attached from your system. You should not copy or use it for any purpose, disclose the contents of the same to any other person or forward it without express permission. Considering the means of transmission, we do not undertake any liability with respect to the secrecy and confidentiality of the information contained in this e-mail and its attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From xzsiro00 at stud.fit.vutbr.cz Wed May 11 10:51:58 2011 From: xzsiro00 at stud.fit.vutbr.cz (=?windows-1250?Q?Anton_Zs=EDros?=) Date: Wed, 11 May 2011 19:51:58 +0200 Subject: [Live-devel] multiple streams with WindowsAudioInputDevice Message-ID: <4DCACCBE.6060600@stud.fit.vutbr.cz> Hello, at first, I would like to apologize for my English, but I hope that, someone will understand. I have some audio inputs (real and virtual) and I want to stream them. My question is: it is posible to stream from multipe audio input devices on Windows (7) at same time? I am using WindowsAudioInputDevice with noMixer suffix. When I stream only one input it works fine, but when I want to start another stream from other input, the first stream will be terminated and I dont know why. As sink I use SimpleRTPSink. I try do this in single event loop, or run multiple event loops for each stream in separate thread (each thread has own TaskSheduler and UsageEnvironment), the result is same. For testing purpose I try run multiple streams from MP3 files using MP3FileSource and MPEG1or2AudioRTPSink, and it works fine. I need it for my master thesis and the time is running out :( Thanks for any help. Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: From felix at embedded-sol.com Wed May 11 23:33:31 2011 From: felix at embedded-sol.com (Felix Radensky) Date: Thu, 12 May 2011 09:33:31 +0300 Subject: [Live-devel] Problems on iPhone when on cell network In-Reply-To: <6B2A0D7D-39ED-44D9-80F1-215D73DC5ECF@uni-kassel.de> References: <6B2A0D7D-39ED-44D9-80F1-215D73DC5ECF@uni-kassel.de> Message-ID: <4DCB7F3B.4040308@embedded-sol.com> Hi Thomas, On 5/11/2011 2:04 PM, Thomas Maier wrote: > Hi gurus, > > I am using the LIVE555 Streaming Media libs on the iPhone. As long as I am on WiFi, everything works fine. However, as soon as I try to use the cell network (2G or 3G), I get the following error: > > Unable to determine our source address: This computer has an invalid IP address: 0x0 > > I grepped the sources and found "This computer has an invalid IP address" in groupsock/GroupsockHelper.cpp. Alas, my network hacking skills are zero. I see the comment "(This code, like many others, won't handle IPv6)", so maybe (wildly guessing here) some info about my carrier is relevant? It is the German Telekom in, surprise, Germany. > > If it's of any use, when I start my app when WiFi is active, get some RTP data, stop, switch to cell in the iPhone settings, and then try to get data, I do not get the above error but a different one: > > RTSP 'OPTIONS' request failed: Failed to find network address for "my.camera.url" > > This happens in continueAfterOPTIONS() (my code is quite a copy of openRTSP and works fine on WiFi). > > Has anybody experienced that problem? Any pointers to what I should read or what I could do? > > I am using the sources as of 2010-10-28. I have read http://live555.com/liveMedia/public/changelog.txt but could not find a relevant change that might fix the problem. Should I upgrade anyway? > > I am on Xcode 3.2.6 with iOS SDK 4.3. gcc --version says > > i686-apple-darwin10-gcc-4.2.1 (GCC) 4.2.1 (Apple Inc. build 5666) (dot 3) > > I am not sure what other info you need but I will happily provide it. > > Thanks in advance, > > Thomas > > This error usually means that you don't have default gateway defined. Felix. From Bruno.Basilio at brisa.pt Thu May 12 01:39:53 2011 From: Bruno.Basilio at brisa.pt (Bruno Filipe Basilio) Date: Thu, 12 May 2011 08:39:53 +0000 Subject: [Live-devel] Problems on iPhone when on cell network Message-ID: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA86190438E9@SRVBRIEXC012.brisa.pt> > Unable to determine our source address: This computer has an invalid IP address: 0x0 I had this problem on my linux box when the live555 RTSP server is launched on boot, more than one network interface is configured and one of them has a dynamic IP served by a DHCP. I don't know if this is your problem but in my case it was enough simply to remove the second interface with dynamic IP, it isn't a definitive solution but it can help understand the symptom. > Has anybody experienced that problem? Any pointers to what I should read or what I could do? Sorry but I don't have the knowledge to help in the other issues. Best regards, Bruno Basilio Brisa Inovacao e Tecnologia, S.A. -------------------------------------------------------------------------------- Declara??o: A informa??o contida nesta mensagem, e os ficheiros anexos, ? privilegiada e confidencial, destinando-se exclusivamente ao(s) destinat?rio(s).Se n?o ? o destinat?rio (ou o respons?vel pela sua entrega ao destinat?rio) e recebeu a mesma por engano, fica notificado que ? estritamente proibido reproduzir, guardar ou distribuir toda ou qualquer parte desta mensagem e ficheiros anexos.Por favor reencaminhe a mensagem para o respons?vel pelo seu envio ou contacte-nos por telefone e elimine a mensagem e ficheiros anexos do seu computador,sem os reproduzir. Disclaimer: The information contained in this message, and any files attached, is privileged and confidential, and intended exclusively for the included addresses.If you are not the intended recipient (or the person responsible for delivering to the intended recipient) and received this message by mistake, be aware that copy, storage, distribution or any other use of all or part of this message and the files attached is strictly prohibited. Please notify the sender by reply e-mail or contact us by telephone and delete this message and the files attached, without retaining a copy. -------------------------------------------------------------------------------- From thomas.maier at uni-kassel.de Thu May 12 03:24:49 2011 From: thomas.maier at uni-kassel.de (Thomas Maier) Date: Thu, 12 May 2011 12:24:49 +0200 Subject: [Live-devel] Problems on iPhone when on cell network In-Reply-To: <4DCB7F3B.4040308@embedded-sol.com> References: <6B2A0D7D-39ED-44D9-80F1-215D73DC5ECF@uni-kassel.de> <4DCB7F3B.4040308@embedded-sol.com> Message-ID: <097A9A5A-F6BA-4BE4-8D6B-058746534D15@uni-kassel.de> Hi Felix, Am 12.05.2011 um 08:33 schrieb Felix Radensky: > Hi Thomas, > > On 5/11/2011 2:04 PM, Thomas Maier wrote: >> Hi gurus, >> >> I am using the LIVE555 Streaming Media libs on the iPhone. As long as I am on WiFi, everything works fine. However, as soon as I try to use the cell network (2G or 3G), I get the following error: >> >> Unable to determine our source address: This computer has an invalid IP address: 0x0 >> >> I grepped the sources and found "This computer has an invalid IP address" in groupsock/GroupsockHelper.cpp. Alas, my network hacking skills are zero. I see the comment "(This code, like many others, won't handle IPv6)", so maybe (wildly guessing here) some info about my carrier is relevant? It is the German Telekom in, surprise, Germany. >> >> If it's of any use, when I start my app when WiFi is active, get some RTP data, stop, switch to cell in the iPhone settings, and then try to get data, I do not get the above error but a different one: >> >> RTSP 'OPTIONS' request failed: Failed to find network address for "my.camera.url" >> >> This happens in continueAfterOPTIONS() (my code is quite a copy of openRTSP and works fine on WiFi). >> >> Has anybody experienced that problem? Any pointers to what I should read or what I could do? >> >> I am using the sources as of 2010-10-28. I have read http://live555.com/liveMedia/public/changelog.txt but could not find a relevant change that might fix the problem. Should I upgrade anyway? >> >> I am on Xcode 3.2.6 with iOS SDK 4.3. gcc --version says >> >> i686-apple-darwin10-gcc-4.2.1 (GCC) 4.2.1 (Apple Inc. build 5666) (dot 3) >> >> I am not sure what other info you need but I will happily provide it. >> >> Thanks in advance, >> >> Thomas >> >> > > This error usually means that you don't have default gateway defined. > > Felix. Thanks for the swift reply. Wouldn't that mean that I could not access the cell network at all? (Beware, my knowledge about networks is shallow.) However, other programs (like Safari and Mail) work fine. It's only my own app and only the code using the Live555 libs and only on the cell network. Other network access in my code (like downloading an NSURLRequest from the same host I try to connect to using the Live555 libs) works fine, even on the cell network. Maybe this is all irrelevant because my understanding of a default gateway is all wrong? Anything I can do to analyze the problem to provide you with further information? Cheers, Thomas From thomas.maier at uni-kassel.de Thu May 12 03:33:56 2011 From: thomas.maier at uni-kassel.de (Thomas Maier) Date: Thu, 12 May 2011 12:33:56 +0200 Subject: [Live-devel] Problems on iPhone when on cell network In-Reply-To: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA86190438E9@SRVBRIEXC012.brisa.pt> References: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA86190438E9@SRVBRIEXC012.brisa.pt> Message-ID: <8D680F20-2096-4D0A-9084-B71CEB146BDB@uni-kassel.de> Am 12.05.2011 um 10:39 schrieb Bruno Filipe Basilio: >> Unable to determine our source address: This computer has an invalid IP address: 0x0 > > I had this problem on my linux box when the live555 RTSP server is launched on boot, more than one network interface is configured and one of them has a dynamic IP served by a DHCP. > I don't know if this is your problem but in my case it was enough simply to remove the second interface with dynamic IP, it isn't a definitive solution but it can help understand the symptom. Hi Bruno, thanks for the reply. My code is running on an iPhone and I suppose I am not able to mess with the network interfaces. And I guess nor should I be :). Have you ever had this problem on a client trying to connect to an RTSP server? Your post made me realize that I did not explicitly state that I am trying to make my iPhone be a client accessing a remote RTSP server. The server works fine and is not under my control, but I am getting the errors for client code running on my iPhone. Any ideas? Thanks, Thomas From Autuori.Gianluigi.Wintime at ansaldobreda.it Thu May 12 04:55:26 2011 From: Autuori.Gianluigi.Wintime at ansaldobreda.it (Autuori Gianluigi) Date: Thu, 12 May 2011 13:55:26 +0200 Subject: [Live-devel] problem using openRTSP Message-ID: Hello, I'm using openRTSP under Linux I'm capturing H264 stream from axis camera: openRTSP -d 20 -4 -f 3 -w 640 -h 480 -b 400000 "rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3" >video.avi sometimes the video is grey the output on the shell is: ************************************************************************ ************************************************************************ ********************************** Sending request: OPTIONS rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 RTSP/1.0 CSeq: 1 User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09) Received OPTIONS response: RTSP/1.0 200 OK CSeq: 1 Public: DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, SET_PARAMETER, TEARDOWN Date: Thu, 12 May 2011 11:50:57 GMT Sending request: DESCRIBE rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 RTSP/1.0 CSeq: 2 Accept: application/sdp User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09) Received DESCRIBE response: RTSP/1.0 200 OK CSeq: 2 Content-Type: application/sdp Content-Base: rtsp://10.10.1.61/axis-media/media.amp/ Date: Thu, 12 May 2011 11:50:57 GMT Content-Length: 496 Need to read 496 extra bytes Read 496 extra bytes: v=0 o=- 1305201057626077 1305201057626077 IN IP4 10.10.1.61 s=Media Presentation e=NONE c=IN IP4 0.0.0.0 b=AS:50000 t=0 0 a=control:rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 a=range:npt=0.000000- m=video 0 RTP/AVP 96 b=AS:50000 a=framerate:3.0 a=control:rtsp://10.10.1.61/axis-media/media.amp/trackID=1?compression=3 5&fps=3 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1; profile-level-id=420029; sprop-parameter-sets=Z0IAKeKQGQJvy4C3AQEBpB4kRUA=,aM48gA== Opened URL "rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3", returning a SDP description: v=0 o=- 1305201057626077 1305201057626077 IN IP4 10.10.1.61 s=Media Presentation e=NONE c=IN IP4 0.0.0.0 b=AS:50000 t=0 0 a=control:rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 a=range:npt=0.000000- m=video 0 RTP/AVP 96 b=AS:50000 a=framerate:3.0 a=control:rtsp://10.10.1.61/axis-media/media.amp/trackID=1?compression=3 5&fps=3 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1; profile-level-id=420029; sprop-parameter-sets=Z0IAKeKQGQJvy4C3AQEBpB4kRUA=,aM48gA== Created receiver for "video/H264" subsession (client ports 55386-55387) Sending request: SETUP rtsp://10.10.1.61/axis-media/media.amp/trackID=1?compression=35&fps=3 RTSP/1.0 CSeq: 3 Transport: RTP/AVP;unicast;client_port=55386-55387 User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09) Received SETUP response: RTSP/1.0 200 OK CSeq: 3 Session: 9D3B4ABC; timeout=60 Transport: RTP/AVP;unicast;client_port=55386-55387;server_port=50804-50805;ssrc=7BD D6747;mode="PLAY" Date: Thu, 12 May 2011 11:50:57 GMT Setup "video/H264" subsession (client ports 55386-55387) Sending request: PLAY rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 RTSP/1.0 CSeq: 4 Session: 9D3B4ABC Range: npt=0.000-10.000 User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09) Received PLAY response: RTSP/1.0 200 OK CSeq: 4 Session: 9D3B4ABC Range: npt=0.145443- RTP-Info: url=rtsp://10.10.1.61/axis-media/media.amp/trackID=1?compression=35&fps= 3;seq=27911;rtptime=588695951 Date: Thu, 12 May 2011 11:50:57 GMT Started playing session Receiving streamed data (for up to 10.000000 seconds)... Sending request: TEARDOWN rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 RTSP/1.0 CSeq: 5 Session: 9D3B4ABC User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09) Received TEARDOWN response: RTSP/1.0 200 OK CSeq: 5 Session: 9D3B4ABC Date: Thu, 12 May 2011 11:51:07 GMT ************************************************************************ ************************************************************************ ********** Is there something wrong? thank you Gianluigi -------------------------------------------------------------------------------- Questo messaggio e-mail e ogni documento ad esso eventualmente allegato puo' avere carattere riservato ed essere tutelato da segreto. Esso,comunque, e' ad esclusivo utilizzo del destinatario in indirizzo. Qualora non foste il destinatario del messaggio vi preghiamo di volerci avvertire immediatamente per e-mail o telefono e di cancellare il presente messaggio e ogni eventuale allegato dal vostro sistema. E' vietata la duplicazione o l'utilizzo per qualunque fine del messaggio e di ogni allegato, nonche' la loro divulgazione, distribuzione o inoltro a terzi senza l'espressa autorizzazione del mittente. In ragione del mezzo di trasmissione utilizzato, il mittente non assume alcuna responsabilita' sulla segretezza/riservatezza delle informazioni contenute nel messaggio e nei relativi allegati. This e-mail and any file transmitted with it may contain material that is confidential, privileged and/or attorney work product for the sole use of the intended recipient. If you are not the intended recipient of this e-mail, please do not read it, notify us immediately by e-mail or by telephone and then delete this message and any file attached from your system. You should not copy or use it for any purpose, disclose the contents of the same to any other person or forward it without express permission. Considering the means of transmission, we do not undertake any liability with respect to the secrecy and confidentiality of the information contained in this e-mail and its attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From matt at schuckmannacres.com Thu May 12 09:16:04 2011 From: matt at schuckmannacres.com (Matt Schuckmannn) Date: Thu, 12 May 2011 09:16:04 -0700 Subject: [Live-devel] Problems on iPhone when on cell network In-Reply-To: <8D680F20-2096-4D0A-9084-B71CEB146BDB@uni-kassel.de> References: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA86190438E9@SRVBRIEXC012.brisa.pt> <8D680F20-2096-4D0A-9084-B71CEB146BDB@uni-kassel.de> Message-ID: <4DCC07C4.3050306@schuckmannacres.com> Hi, I've had Live555 working on a iPhone as a RTSP client for some time. I have noticed that sometimes on some iPhone's when they are on the cell network I can not connect to the server. When this happens I've always been able to shutdown my app, launch mail or safari to kick the connection into gear and then switch back to my app and things work fine. Since my code is just demo prototype stuff I've never really investigated what is going on when it can't connect but perhaps this is a clue. Matt S. On 5/12/2011 3:33 AM, Thomas Maier wrote: > Am 12.05.2011 um 10:39 schrieb Bruno Filipe Basilio: >>> Unable to determine our source address: This computer has an invalid IP address: 0x0 >> I had this problem on my linux box when the live555 RTSP server is launched on boot, more than one network interface is configured and one of them has a dynamic IP served by a DHCP. >> I don't know if this is your problem but in my case it was enough simply to remove the second interface with dynamic IP, it isn't a definitive solution but it can help understand the symptom. > Hi Bruno, > > thanks for the reply. My code is running on an iPhone and I suppose I am not able to mess with the network interfaces. And I guess nor should I be :). Have you ever had this problem on a client trying to connect to an RTSP server? Your post made me realize that I did not explicitly state that I am trying to make my iPhone be a client accessing a remote RTSP server. The server works fine and is not under my control, but I am getting the errors for client code running on my iPhone. Any ideas? > > Thanks, Thomas > > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel From thomas.maier at uni-kassel.de Thu May 12 09:59:53 2011 From: thomas.maier at uni-kassel.de (Thomas Maier) Date: Thu, 12 May 2011 18:59:53 +0200 Subject: [Live-devel] Problems on iPhone when on cell network In-Reply-To: <4DCC07C4.3050306@schuckmannacres.com> References: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA86190438E9@SRVBRIEXC012.brisa.pt> <8D680F20-2096-4D0A-9084-B71CEB146BDB@uni-kassel.de> <4DCC07C4.3050306@schuckmannacres.com> Message-ID: <07F2F3C9-6E28-46C1-99B3-7A9BE44A74BB@uni-kassel.de> Am 12.05.2011 um 18:16 schrieb Matt Schuckmannn: > Hi, I've had Live555 working on a iPhone as a RTSP client for some time. > I have noticed that sometimes on some iPhone's when they are on the cell network I can not connect to the server. When this happens I've always been able to shutdown my app, launch mail or safari to kick the connection into gear and then switch back to my app and things work fine. > Since my code is just demo prototype stuff I've never really investigated what is going on when it can't connect but perhaps this is a clue. > > Matt S. Hi Matt, thanks for the hint. Just tested what you suggested. Doesn't work for me, unfortunately. All the other apps work fine on the cell network, even the parts of my own app that are not related to Live555. It is only when I try to use the part of my app that uses Live555 that I get the error I mentioned: Unable to determine our source address: This computer has an invalid IP address: 0x0 Thanks, Thomas From Autuori.Gianluigi.Wintime at ansaldobreda.it Fri May 13 00:34:27 2011 From: Autuori.Gianluigi.Wintime at ansaldobreda.it (Autuori Gianluigi) Date: Fri, 13 May 2011 09:34:27 +0200 Subject: [Live-devel] R: problem using openRTSP References: Message-ID: I solved it. It was a network problem thanks Gianluigi _____ Da: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] Per conto di Autuori Gianluigi Inviato: gioved? 12 maggio 2011 13.55 A: live-devel at ns.live555.com Oggetto: [Live-devel] problem using openRTSP Hello, I'm using openRTSP under Linux I'm capturing H264 stream from axis camera: openRTSP -d 20 -4 -f 3 -w 640 -h 480 -b 400000 "rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3" >video.avi sometimes the video is grey the output on the shell is: ********************************************************************************************************************************************************************************** Sending request: OPTIONS rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 RTSP/1.0 CSeq: 1 User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09) Received OPTIONS response: RTSP/1.0 200 OK CSeq: 1 Public: DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, SET_PARAMETER, TEARDOWN Date: Thu, 12 May 2011 11:50:57 GMT Sending request: DESCRIBE rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 RTSP/1.0 CSeq: 2 Accept: application/sdp User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09) Received DESCRIBE response: RTSP/1.0 200 OK CSeq: 2 Content-Type: application/sdp Content-Base: rtsp://10.10.1.61/axis-media/media.amp/ Date: Thu, 12 May 2011 11:50:57 GMT Content-Length: 496 Need to read 496 extra bytes Read 496 extra bytes: v=0 o=- 1305201057626077 1305201057626077 IN IP4 10.10.1.61 s=Media Presentation e=NONE c=IN IP4 0.0.0.0 b=AS:50000 t=0 0 a=control:rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 a=range:npt=0.000000- m=video 0 RTP/AVP 96 b=AS:50000 a=framerate:3.0 a=control:rtsp://10.10.1.61/axis-media/media.amp/trackID=1?compression=35&fps=3 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1; profile-level-id=420029; sprop-parameter-sets=Z0IAKeKQGQJvy4C3AQEBpB4kRUA=,aM48gA== Opened URL "rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3", returning a SDP description: v=0 o=- 1305201057626077 1305201057626077 IN IP4 10.10.1.61 s=Media Presentation e=NONE c=IN IP4 0.0.0.0 b=AS:50000 t=0 0 a=control:rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 a=range:npt=0.000000- m=video 0 RTP/AVP 96 b=AS:50000 a=framerate:3.0 a=control:rtsp://10.10.1.61/axis-media/media.amp/trackID=1?compression=35&fps=3 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1; profile-level-id=420029; sprop-parameter-sets=Z0IAKeKQGQJvy4C3AQEBpB4kRUA=,aM48gA== Created receiver for "video/H264" subsession (client ports 55386-55387) Sending request: SETUP rtsp://10.10.1.61/axis-media/media.amp/trackID=1?compression=35&fps=3 RTSP/1.0 CSeq: 3 Transport: RTP/AVP;unicast;client_port=55386-55387 User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09) Received SETUP response: RTSP/1.0 200 OK CSeq: 3 Session: 9D3B4ABC; timeout=60 Transport: RTP/AVP;unicast;client_port=55386-55387;server_port=50804-50805;ssrc=7BDD6747;mode="PLAY" Date: Thu, 12 May 2011 11:50:57 GMT Setup "video/H264" subsession (client ports 55386-55387) Sending request: PLAY rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 RTSP/1.0 CSeq: 4 Session: 9D3B4ABC Range: npt=0.000-10.000 User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09) Received PLAY response: RTSP/1.0 200 OK CSeq: 4 Session: 9D3B4ABC Range: npt=0.145443- RTP-Info: url=rtsp://10.10.1.61/axis-media/media.amp/trackID=1?compression=35&fps=3;seq=27911;rtptime=588695951 Date: Thu, 12 May 2011 11:50:57 GMT Started playing session Receiving streamed data (for up to 10.000000 seconds)... Sending request: TEARDOWN rtsp://10.10.1.61/axis-media/media.amp?compression=35&fps=3 RTSP/1.0 CSeq: 5 Session: 9D3B4ABC User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09) Received TEARDOWN response: RTSP/1.0 200 OK CSeq: 5 Session: 9D3B4ABC Date: Thu, 12 May 2011 11:51:07 GMT ********************************************************************************************************************************************************** Is there something wrong? thank you Gianluigi _____ Questo messaggio e-mail e ogni documento ad esso eventualmente allegato puo' avere carattere riservato ed essere tutelato da segreto. Esso,comunque, e' ad esclusivo utilizzo del destinatario in indirizzo. Qualora non foste il destinatario del messaggio vi preghiamo di volerci avvertire immediatamente per e-mail o telefono e di cancellare il presente messaggio e ogni eventuale allegato dal vostro sistema. E' vietata la duplicazione o l'utilizzo per qualunque fine del messaggio e di ogni allegato, nonche' la loro divulgazione, distribuzione o inoltro a terzi senza l'espressa autorizzazione del mittente. In ragione del mezzo di trasmissione utilizzato, il mittente non assume alcuna responsabilita' sulla segretezza/riservatezza delle informazioni contenute nel messaggio e nei relativi allegati. This e-mail and any file transmitted with it may contain material that is confidential, privileged and/or attorney work product for the sole use of the intended recipient. If you are not the intended recipient of this e-mail, please do not read it, notify us immediately by e-mail or by telephone and then delete this message and any file attached from your system. You should not copy or use it for any purpose, disclose the contents of the same to any other person or forward it without express permission. Considering the means of transmission, we do not undertake any liability with respect to the secrecy and confidentiality of the information contained in this e-mail and its attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From johnmcnamara at sendmode.com Tue May 17 15:37:27 2011 From: johnmcnamara at sendmode.com (John McNamara) Date: Tue, 17 May 2011 23:37:27 +0100 Subject: [Live-devel] Problem with RTSP Audio Message-ID: <292BAFDD44734E7994A86DAB23184AD1@BUSINESS> Hi, I have integrated Live555Media into an iPhone app and wish to play an audio (.ACC) file from our RTSP server ( rtsp://91.123.225.100:8554/channel1 .. this is live server publicly viewable ) We are successfully handshaking with the server with DESCRIBE and PLAY commands and from the console can see loads of data coming through, problem is I'm new to all this and don't really know how to read and process these packets and send to iPhone speakers. Has anyone done this before and could help us? All help is really appreciated. Thanks, John John McNamara, Sendmode.com Colab, LYIT, Letterkenny, Co. Donegal. t: 074 9116059 m: 086 1524741 w: www.sendmode.com -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 4113 bytes Desc: not available URL: From steve at stevemcfarlin.com Tue May 17 16:26:37 2011 From: steve at stevemcfarlin.com (Steve McFarlin) Date: Tue, 17 May 2011 16:26:37 -0700 Subject: [Live-devel] Problem with RTSP Audio In-Reply-To: <292BAFDD44734E7994A86DAB23184AD1@BUSINESS> References: <292BAFDD44734E7994A86DAB23184AD1@BUSINESS> Message-ID: <47DE0891-F7D7-4C04-8880-B4C211E4B071@stevemcfarlin.com> On May 17, 2011, at 3:37 PM, John McNamara wrote: > Hi, > > I have integrated Live555Media into an iPhone app and wish to play an audio (.ACC) file from our RTSP server ( rtsp://91.123.225.100:8554/channel1 .. this is live server publicly viewable ) > > We are successfully handshaking with the server with DESCRIBE and PLAY commands and from the console can see loads of data coming through, problem is I?m new to all this and don?t really know how to read and process these packets and send to iPhone speakers. > > Has anyone done this before and could help us? > > All help is really appreciated. > > Thanks, > John > > John McNamara, > Sendmode.com > Colab, LYIT, > Letterkenny, > Co. Donegal. > t: 074 9116059 > m: 086 1524741 > w: www.sendmode.com > > > > > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel Hello John, Your question is not appropriate for this list. I would recommend asking at stackoverflow.com. Here is my response to your future stackoverflow question. Contact me directly if you want further information. I have tested Live555 to both broadcast and receive audio and video on the iPhone. I do not recommend using it if you are going to be streaming AV in and out of the phone. It is not meant for this, and will only frustrate you trying to get it to work. You should be fine if all you need is to stream audio into the phone. WIth this said... I am assuming you are streaming RTP packets with a single frame of AAC data. You need to push the packet payload into a FIFO. Then pull the AAC frames from the FIFO in the callback for an AudioQueue. You could also go down a level and use AudioUnits along with the AudioConverter services. Thanks, Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: From johnmcnamara at sendmode.com Tue May 17 16:39:16 2011 From: johnmcnamara at sendmode.com (John McNamara) Date: Wed, 18 May 2011 00:39:16 +0100 Subject: [Live-devel] Problem with RTSP Audio In-Reply-To: <47DE0891-F7D7-4C04-8880-B4C211E4B071@stevemcfarlin.com> References: <292BAFDD44734E7994A86DAB23184AD1@BUSINESS> <47DE0891-F7D7-4C04-8880-B4C211E4B071@stevemcfarlin.com> Message-ID: Thanks Steve, Much appreciated. John McNamara, Sendmode.com Colab, LYIT, Letterkenny, Co. Donegal. t: 074 9116059 m: 086 1524741 w: www.sendmode.com _____ From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Steve McFarlin Sent: 18 May 2011 00:27 To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Problem with RTSP Audio On May 17, 2011, at 3:37 PM, John McNamara wrote: Hi, I have integrated Live555Media into an iPhone app and wish to play an audio (.ACC) file from our RTSP server ( rtsp://91.123.225.100:8554/channel1 .. this is live server publicly viewable ) We are successfully handshaking with the server with DESCRIBE and PLAY commands and from the console can see loads of data coming through, problem is I'm new to all this and don't really know how to read and process these packets and send to iPhone speakers. Has anyone done this before and could help us? All help is really appreciated. Thanks, John John McNamara, Sendmode.com Colab, LYIT, Letterkenny, Co. Donegal. t: 074 9116059 m: 086 1524741 w: www.sendmode.com _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel Hello John, Your question is not appropriate for this list. I would recommend asking at stackoverflow.com. Here is my response to your future stackoverflow question. Contact me directly if you want further information. I have tested Live555 to both broadcast and receive audio and video on the iPhone. I do not recommend using it if you are going to be streaming AV in and out of the phone. It is not meant for this, and will only frustrate you trying to get it to work. You should be fine if all you need is to stream audio into the phone. WIth this said... I am assuming you are streaming RTP packets with a single frame of AAC data. You need to push the packet payload into a FIFO. Then pull the AAC frames from the FIFO in the callback for an AudioQueue. You could also go down a level and use AudioUnits along with the AudioConverter services. Thanks, Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 4113 bytes Desc: not available URL: From finlayson at live555.com Tue May 17 15:47:10 2011 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 17 May 2011 18:47:10 -0400 Subject: [Live-devel] Problem with RTSP Audio In-Reply-To: <47DE0891-F7D7-4C04-8880-B4C211E4B071@stevemcfarlin.com> References: <292BAFDD44734E7994A86DAB23184AD1@BUSINESS> <47DE0891-F7D7-4C04-8880-B4C211E4B071@stevemcfarlin.com> Message-ID: >Your question is not appropriate for this list. John's question was *somewhat* appropriate for the list, because he's using the LIVE555 library for his iPhone RTSP client application. However, because our library does not include any decoding functionality, we can't help you figure out how to decode the incoming audio (in this case, MPEG-4 audio) data on the iPhone (or on any other client). >I have tested Live555 to both broadcast and receive audio and video >on the iPhone. I do not recommend using it if you are going to be >streaming AV in and out of the phone. It is not meant for this That's not true. There's no inherent reason why our library can't be used to receive/play RTSP audio+video streams. In fact, several clients out there already do this. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From steve at stevemcfarlin.com Tue May 17 23:08:52 2011 From: steve at stevemcfarlin.com (Steve McFarlin) Date: Tue, 17 May 2011 23:08:52 -0700 Subject: [Live-devel] Problem with RTSP Audio In-Reply-To: References: <292BAFDD44734E7994A86DAB23184AD1@BUSINESS> <47DE0891-F7D7-4C04-8880-B4C211E4B071@stevemcfarlin.com> Message-ID: <9E89F3FF-E899-4653-B465-A3D2D0728276@stevemcfarlin.com> >> I have tested Live555 to both broadcast and receive audio and video on the iPhone. I do not recommend using it if you are going to be streaming AV in and out of the phone. It is not meant for this > > That's not true. There's no inherent reason why our library can't be used to receive/play RTSP audio+video streams. In fact, several clients out there already do this. > -- I tried for some time to get this working properly. It worked wonderfully for streaming AVC/AAC out of the iPhone. For an incoming stream my decoder data queue was starved if I had outgoing data. Apparently it was how I was using the library. Maybe I should revisit that code. In any case the library is very good. I would have preferred to use it. Thanks, Steve From felix at embedded-sol.com Tue May 17 23:20:07 2011 From: felix at embedded-sol.com (Felix Radensky) Date: Wed, 18 May 2011 09:20:07 +0300 Subject: [Live-devel] Strange problem with h264 streaming In-Reply-To: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA861904376B@SRVBRIEXC012.brisa.pt> References: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA861904376B@SRVBRIEXC012.brisa.pt> Message-ID: <4DD36517.4040106@embedded-sol.com> Hi Ross, On 05/11/2011 04:35 PM, Bruno Filipe Basilio wrote: > Hi Felix, > > I have a similar problem. > After 3 days playing I couldn't access the RTP/RTCP server PC anymore. Pinging didn't work either. Only after a reboot everything is ok. > A few days later the same happened. > > And now I discovered something even stranger: > I sniffed the network traffic going to and going from the server to see if I was missing anything: > > Analyzing the traffic I saw a stream coming from the server that was 35Mbps analyzing it some more it is a RTCP stream -> this doesn't seems OK (a rtcp stream that is 20 times bigger than the actual video stream) > > Seeing this I did some more experiments: > 1) first I power off power on the server and put it in a separate lan. > After sniffing the network I found 4 multicast streams coming out of it: > - 232.29.51.215:18888 -> rtp video stream from 2Mbps > - 232.29.51.215:18889 -> rtcp sender reports (a few of them) > - 232.218.63.39:18890 -> rtp video stream from about 200kbps > - 232.218.63.39:18891 -> rtcp sender reports (just a few of them) > 2) A placed a 2 decoders in the network and ask the rtsp multicast stream for both of them > I received the stream and could decode it (the stream was 232.29.51.215:18888) > But the rtcp stream 232.29.51.215:18889 suddenly became enormous about 35 Mbps > I think this is a bug and this is causing eventually the strange behavior I mention earlier (not reacting on ping anymore) > > > Do you have any insights what could be the problem here ? It seems to happen on different devices using live555 to stream h264 via RTSP. I use ARM based DM6467 and Bruno is running live555 application on a PC. Your feedback is very much appreciated. Thanks a lot. Felix. From finlayson at live555.com Wed May 18 02:14:56 2011 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 18 May 2011 05:14:56 -0400 Subject: [Live-devel] Problem with RTSP Audio In-Reply-To: <9E89F3FF-E899-4653-B465-A3D2D0728276@stevemcfarlin.com> References: <292BAFDD44734E7994A86DAB23184AD1@BUSINESS> <47DE0891-F7D7-4C04-8880-B4C211E4B071@stevemcfarlin.com> <9E89F3FF-E899-4653-B465-A3D2D0728276@stevemcfarlin.com> Message-ID: > >> I have tested Live555 to both broadcast and receive audio and >video on the iPhone. I do not recommend using it if you are going to >be streaming AV in and out of the phone. It is not meant for this >> >> That's not true. There's no inherent reason why our library can't >>be used to receive/play RTSP audio+video streams. In fact, several >>clients out there already do this. >> -- > >I tried for some time to get this working properly. It worked >wonderfully for streaming AVC/AAC out of the iPhone. For an incoming >stream my decoder data queue was starved if I had outgoing data. >Apparently it was how I was using the library. Maybe I should >revisit that code. In any case the library is very good. I would >have preferred to use it. To handle incoming data, you should implement a "jitter buffer", to allow for network 'jitter' (variability in the delay of incoming packets), and also possible packet loss. You should also take into account the "presentation time" of each incoming piece of data (to decide when to decode/play it). (Also note ) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From Bruno.Basilio at brisa.pt Wed May 18 03:37:35 2011 From: Bruno.Basilio at brisa.pt (Bruno Filipe Basilio) Date: Wed, 18 May 2011 10:37:35 +0000 Subject: [Live-devel] Strange problem with h264 streaming In-Reply-To: <4DD36517.4040106@embedded-sol.com> References: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA861904376B@SRVBRIEXC012.brisa.pt> <4DD36517.4040106@embedded-sol.com> Message-ID: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA86190441E6@SRVBRIEXC012.brisa.pt> > Do you have any insights what could be the problem here ? > It seems to happen on different devices using live555 to stream > h264 via RTSP. I use ARM based DM6467 and Bruno is running live555 > application on a PC. My setup is the Ubuntu Server Linux 10.04 LTS. I believe that system instability caused by the RTCP stream increased traffic is related to a kernel issue mentioned here: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/472057 Best regards, Bruno Basilio -------------------------------------------------------------------------------- Declara??o: A informa??o contida nesta mensagem, e os ficheiros anexos, ? privilegiada e confidencial, destinando-se exclusivamente ao(s) destinat?rio(s).Se n?o ? o destinat?rio (ou o respons?vel pela sua entrega ao destinat?rio) e recebeu a mesma por engano, fica notificado que ? estritamente proibido reproduzir, guardar ou distribuir toda ou qualquer parte desta mensagem e ficheiros anexos.Por favor reencaminhe a mensagem para o respons?vel pelo seu envio ou contacte-nos por telefone e elimine a mensagem e ficheiros anexos do seu computador,sem os reproduzir. Disclaimer: The information contained in this message, and any files attached, is privileged and confidential, and intended exclusively for the included addresses.If you are not the intended recipient (or the person responsible for delivering to the intended recipient) and received this message by mistake, be aware that copy, storage, distribution or any other use of all or part of this message and the files attached is strictly prohibited. Please notify the sender by reply e-mail or contact us by telephone and delete this message and the files attached, without retaining a copy. -------------------------------------------------------------------------------- From johnmcnamara at sendmode.com Wed May 18 03:53:03 2011 From: johnmcnamara at sendmode.com (John McNamara) Date: Wed, 18 May 2011 11:53:03 +0100 Subject: [Live-devel] Problem with RTSP Audio In-Reply-To: References: <292BAFDD44734E7994A86DAB23184AD1@BUSINESS><47DE0891-F7D7-4C04-8880-B4C211E4B071@stevemcfarlin.com><9E89F3FF-E899-4653-B465-A3D2D0728276@stevemcfarlin.com> Message-ID: Thanks so much Ross, Guys I really appreciate all the help... sorry for all the newbie questions... John McNamara, Sendmode.com Colab, LYIT, Letterkenny, Co. Donegal. t: 074 9116059 m: 086 1524741 w: www.sendmode.com -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson Sent: 18 May 2011 10:15 To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Problem with RTSP Audio > >> I have tested Live555 to both broadcast and receive audio and >video on the iPhone. I do not recommend using it if you are going to >be streaming AV in and out of the phone. It is not meant for this >> >> That's not true. There's no inherent reason why our library can't >>be used to receive/play RTSP audio+video streams. In fact, several >>clients out there already do this. >> -- > >I tried for some time to get this working properly. It worked >wonderfully for streaming AVC/AAC out of the iPhone. For an incoming >stream my decoder data queue was starved if I had outgoing data. >Apparently it was how I was using the library. Maybe I should >revisit that code. In any case the library is very good. I would >have preferred to use it. To handle incoming data, you should implement a "jitter buffer", to allow for network 'jitter' (variability in the delay of incoming packets), and also possible packet loss. You should also take into account the "presentation time" of each incoming piece of data (to decide when to decode/play it). (Also note ) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel From sawan.das at videonetics.com Wed May 18 03:41:42 2011 From: sawan.das at videonetics.com (Sawan Das) Date: Wed, 18 May 2011 16:11:42 +0530 Subject: [Live-devel] RTSP Authentication memory leak Message-ID: <1305715302.16174.7.camel@VTPL-Quad2> Hi All, I am using live555 for last few months. Last week I downloaded the latest archive. Now when i integrate it, from valgrind I have some memory leak problem. For RTSPClient.cpp sendPlayCommand, sendSetupCommand, sendOptionsCommand and sendDescribeCommand the statement "if (authenticator != NULL) fCurrentAuthenticator = *authenticator;" allocated the the memory for the overloaded operartor "=". But memory is not deleted. The previous versions don't have this problem. I cannot access the class as i inherited my own client from it. Any idea about this... Regards, Sawan -- Activated By Videonetics Sawan Das Software Engineer Videonetics Technology Pvt. Ltd. sawan.das at videonetics.com +919836275503 www.videonetics.com From KUCALABAL at battelle.org Wed May 18 07:59:51 2011 From: KUCALABAL at battelle.org (Kucalaba, Luke) Date: Wed, 18 May 2011 10:59:51 -0400 Subject: [Live-devel] Small memory leak Message-ID: <92D05FD48DA1CD41AD57254CB23732D70379B5C670@WS-BCO-MBX2.milky-way.battelle.org> Not sure if this has been confirmed by the live555 developers yet, but I believe there is a small memory leak that occurs during destruction of DelayQueue, if there are any entries left on the queue when DelayQueue is destroyed. You can reproduce this by constructing a BasicTaskScheduler and then delete it immediately. You will see the schedulerTickTask AlarmHandler is never destroyed. Thanks, Luke Kucalaba Research Scientist Battelle Memorial Institute Columbus, Ohio -------------- next part -------------- An HTML attachment was scrubbed... URL: From Vincent_Kao at ralinktech.com Thu May 19 05:03:14 2011 From: Vincent_Kao at ralinktech.com (RA-Vincent Kao) Date: Thu, 19 May 2011 12:03:14 +0000 Subject: [Live-devel] About RTSPServer Message-ID: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> Dear all, How do I stop RTSP Server(live555MediaServer.exe) except terminating live555MediaServer.exe? I update libliveMedia.lib by modifying RTSPServer.cpp to let live555MediaServer.exe has chance to call ~RTSPServer(). The scenario is that RTSP server(live555MediaServer.exe) has established the connection with VLC player(running streaming). After live555MediaServer.exe run ~RTSPServer(), the socket still exist. How do I execute RTSP server socket destruction without terminating live555MediaServer.exe? Thanks. CONFIDENTIALITY STATEMENT : The information, attachments and any rights attaching in this e-mail are confidential and privileged; it is intended only for the individual or entity named as the recipient hereof.Any disclosure, copying, distribution, dissemination or use of the contents of this e-mail by persons other than the intended recipient is STRICTLY PROHIBITED and may violate applicable laws.If you have received this e-mail in error, please delete the original message and notify us by return email or collect call immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Thu May 19 11:22:13 2011 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 19 May 2011 13:22:13 -0500 Subject: [Live-devel] About RTSPServer In-Reply-To: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> References: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> Message-ID: >How do I stop RTSP Server(live555MediaServer.exe) except terminating >live555MediaServer.exe? There's nothing wrong with just terminating the "live555MediaServer" application? This will remove the process, and the operating system will do all cleanup necessary (reclaiming memory, closing sockets, etc.). That's what operating systems are for. If, however, you have a "RTSPServer" object running within an application (i.e. process), and you want to delete (and reclaim) this "RTSPServer" object - while keeping the rest of the process running - then you can do this simply by calling Medium::close(pointerToYourRTSPServerObject); You don't need to modify any of the supplied code. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From Vincent_Kao at ralinktech.com Thu May 19 23:02:50 2011 From: Vincent_Kao at ralinktech.com (RA-Vincent Kao) Date: Fri, 20 May 2011 06:02:50 +0000 Subject: [Live-devel] About RTSPServer In-Reply-To: References: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> Message-ID: <42E7479B4A053E44AA8A67EECE348335092C9C4E@TWHCMBS02.tw.ralinktech.ad> Hi, Yes, I called Medium::close(pointerToYourRTSPServerObject) to stop RTSP Server. But I found socket still exist after call that. The status is as following: Before I call Medium::close, RTSP Server connects to VLC media player, and I run Windows system command "netstat -a" to check as following: TCP 0.0.0.0:8554 VincentTestPC:0 LISTENING TCP 192.168.43.70:8554 Software-PC:56139 ESTABLISHED After I call Medium::close, RTSP server still connects to VLC media player, and "netstat -a" shown as following: TCP 192.168.43.70:8554 Software-PC:56139 ESTABLISHED But if I terminate application, port 8554 will destroyed. What do they have any ideas about this? Sorry that I am new to live555 project. Thanks. Vincent Kao -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson Sent: Friday, May 20, 2011 2:22 AM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] About RTSPServer >How do I stop RTSP Server(live555MediaServer.exe) except terminating >live555MediaServer.exe? There's nothing wrong with just terminating the "live555MediaServer" application? This will remove the process, and the operating system will do all cleanup necessary (reclaiming memory, closing sockets, etc.). That's what operating systems are for. If, however, you have a "RTSPServer" object running within an application (i.e. process), and you want to delete (and reclaim) this "RTSPServer" object - while keeping the rest of the process running - then you can do this simply by calling Medium::close(pointerToYourRTSPServerObject); You don't need to modify any of the supplied code. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel CONFIDENTIALITY STATEMENT : The information, attachments and any rights attaching in this e-mail are confidential and privileged; it is intended only for the individual or entity named as the recipient hereof.Any disclosure, copying, distribution, dissemination or use of the contents of this e-mail by persons other than the intended recipient is STRICTLY PROHIBITED and may violate applicable laws.If you have received this e-mail in error, please delete the original message and notify us by return email or collect call immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Fri May 20 03:34:31 2011 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 20 May 2011 05:34:31 -0500 Subject: [Live-devel] About RTSPServer In-Reply-To: <42E7479B4A053E44AA8A67EECE348335092C9C4E@TWHCMBS02.tw.ralinktech.ad> References: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9C4E@TWHCMBS02.tw.ralinktech.ad> Message-ID: >Yes, I called Medium::close(pointerToYourRTSPServerObject) to stop >RTSP Server. >But I found socket still exist after call that. That's odd. The "RTSPServer" destructor definitely closes the server's socket. I don't know why that's not working for you. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From Vincent_Kao at ralinktech.com Mon May 23 00:08:18 2011 From: Vincent_Kao at ralinktech.com (RA-Vincent Kao) Date: Mon, 23 May 2011 07:08:18 +0000 Subject: [Live-devel] About RTSPServer In-Reply-To: References: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9C4E@TWHCMBS02.tw.ralinktech.ad> Message-ID: <42E7479B4A053E44AA8A67EECE348335092C9DF4@TWHCMBS02.tw.ralinktech.ad> Dear all, I post my source as following(refer to Live555MediaServer.cpp): HANDLE g_hStartServerThread; char g_watch; //Used for doEventLoop() volatile BOOL g_bServerStarted = FALSE; void StartServerThread( LPVOID pParam ) { DBGPRINT(RT_DEBUG_TRACE, _T("[RaWLAPI][WDP] --> StartServerThread\n")); TaskScheduler* scheduler = BasicTaskScheduler::createNew(); UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler); UserAuthenticationDatabase* authDB = NULL; #ifdef ACCESS_CONTROL // To implement client access control to the RTSP server, do the following: authDB = new UserAuthenticationDatabase; authDB->addUserRecord("username1", "password1"); // replace these with real strings // Repeat the above with each , that you wish to allow // access to the server. #endif // Create the RTSP server. Try first with the default port number (554), // and then with the alternative port number (8554): RTSPServer* rtspServer; portNumBits rtspServerPortNum = 554; rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB); if (rtspServer == NULL) { rtspServerPortNum = 8554; rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB); } if (rtspServer == NULL) { *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n"; exit(1); } DBGPRINT(RT_DEBUG_TRACE, _T("[RaWLAPI][WDP] LIVE555 Media Server version = %s\n"), MEDIA_SERVER_VERSION_STRING); DBGPRINT(RT_DEBUG_TRACE, _T("[RaWLAPI][WDP] LIVE555 Streaming Media library version = %s\n"), LIVEMEDIA_LIBRARY_VERSION_STRING); char* urlPrefix = rtspServer->rtspURLPrefix(); DBGPRINT(RT_DEBUG_TRACE, _T("[RaWLAPI][WDP] URL = %s\n"), urlPrefix); DBGPRINT(RT_DEBUG_TRACE, _T("[RaWLAPI][WDP] Support file type: aac, amr, m4e, 264, dv, mp3, mpg, ts, wmv\n")); // Also, attempt to create a HTTP server for RTSP-over-HTTP tunneling. // Try first with the default HTTP port (80), and then with the alternative HTTP // port numbers (8000 and 8080). if (rtspServer->setUpTunnelingOverHTTP(80) || rtspServer->setUpTunnelingOverHTTP(8000) || rtspServer->setUpTunnelingOverHTTP(8080)) { DBGPRINT(RT_DEBUG_TRACE, _T("[RaWLAPI][WDP] Use port %d for optional RTSP-over-HTTP tunneling\n"), rtspServer->httpServerPortNum()); } else { DBGPRINT(RT_DEBUG_ERROR, _T("[RaWLAPI][WDP] RTSP-over-HTTP tunneling is not available!\n")); } g_watch = 0; g_bServerStarted = TRUE; env->taskScheduler().doEventLoop(&g_watch); // does not return Medium::close(rtspServer); //Close socket DBGPRINT(RT_DEBUG_TRACE, _T("[RaWLAPI][WDP] <-- StartServerThread\n")); } //************* API ************/ DWORD WINAPI RARTSP_StartOrStopRTSPServer(BOOL bStart, TCHAR *szServerIP, int nServerPort) { DWORD dwRet = RAWL_ERROR_SUCCESS; if(bStart) { g_hStartServerThread = NULL; g_hStartServerThread = (HANDLE)_beginthread(StartServerThread, NULL, NULL); while(FALSE == g_bServerStarted) { Sleep(10); } } else //Stop server { g_watch = 1; //Let doEventLoop() return } return dwRet; } When I want to stop server, I modify watch variable to let event loop return, and then call Media::Close to close server's socket. But socket still exist(netstat -a show socket established) after that call. And though socket can be destroyed after terminate application, debug log also generates many memory leaks. And I wonder why Live555MediaServer.cpp does not delete scheduler and env objects, how are these objects released when application is terminated? Thanks. Vincent Kao -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson Sent: Friday, May 20, 2011 6:35 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] About RTSPServer >Yes, I called Medium::close(pointerToYourRTSPServerObject) to stop >RTSP Server. >But I found socket still exist after call that. That's odd. The "RTSPServer" destructor definitely closes the server's socket. I don't know why that's not working for you. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel CONFIDENTIALITY STATEMENT : The information, attachments and any rights attaching in this e-mail are confidential and privileged; it is intended only for the individual or entity named as the recipient hereof.Any disclosure, copying, distribution, dissemination or use of the contents of this e-mail by persons other than the intended recipient is STRICTLY PROHIBITED and may violate applicable laws.If you have received this e-mail in error, please delete the original message and notify us by return email or collect call immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Mon May 23 23:14:47 2011 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 24 May 2011 00:14:47 -0600 Subject: [Live-devel] About RTSPServer In-Reply-To: <42E7479B4A053E44AA8A67EECE348335092C9DF4@TWHCMBS02.tw.ralinktech.ad> References: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9C4E@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9DF4@TWHCMBS02.tw.ralinktech.ad> Message-ID: >I post my source Unfortunately few - if any - people on the mailing list have the time to look through your source code. However, You Have Complete Source Code for the "LIVE555 Streaming Media" libraries. You can see where the RTSP server's socket gets closed (hint: line 220 of "liveMedia/RTSPServer.cpp"), so you should be able to figure out why that is not working for you. >And I wonder why Live555MediaServer.cpp does not delete scheduler >and env objects, >how are these objects released when application is terminated? The ***operating system*** releases all of the application's memory (and sockets), when the application is terminated. Unfortunately, you seem to have a very limited understanding of "processes", and how operating systems handle them. (This will be my last posting on this topic.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From Priya_Raghavendra at mindtree.com Mon May 23 23:23:09 2011 From: Priya_Raghavendra at mindtree.com (Priya Raghavendra) Date: Tue, 24 May 2011 06:23:09 +0000 Subject: [Live-devel] Live streaming stops after half an hour Message-ID: Hi, We have been facing this problem of live streaming stopping exactly after half an hour. The same code works on Linux OS but windows this issue is there. The readData function does not get called, And it does not get into the loop isCurrentlyAwaitingData. So incomingDataHandler1 itself is not getting called. Under what conditions will live stop calling incomingDataHandler1 ? void LiveInputSource:: incomingDataHandler1() { int ret; if (!isCurrentlyAwaitingData()) { printf("isCurrentlyAwaitingData\n"); return ; } ret = readData(); if (ret < 0 ) { handleClosure (this); printf("handleClosure (this)\n"); } else if (ret == 0 ) { int uSecsToDelay = 50000 ; nextTask() = envir().taskScheduler().scheduleDelayedTask (uSecsToDelay, (TaskFunc*)incomingDataHandler, this); } else { nextTask() = envir().taskScheduler().scheduleDelayedTask (0, (TaskFunc*)afterGetting, this); } } int LiveInputVideoSource:: readData() { /* Get frame from queue */ queue_get (frame, &frame_len); if (frame_len == 0) { return -1; } /* Copy frame to fTo */ memcpy (fTo, frame, frame_len); fFrameSize = frame_len; /* Presentation Time */ gettimeofday (&fPresentationTime); return 1; } Thanks, Priya ________________________________ http://www.mindtree.com/email/disclaimer.html -------------- next part -------------- An HTML attachment was scrubbed... URL: From Vincent_Kao at ralinktech.com Mon May 23 23:57:59 2011 From: Vincent_Kao at ralinktech.com (RA-Vincent Kao) Date: Tue, 24 May 2011 06:57:59 +0000 Subject: [Live-devel] About RTSPServer In-Reply-To: References: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9C4E@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9DF4@TWHCMBS02.tw.ralinktech.ad> Message-ID: <42E7479B4A053E44AA8A67EECE348335092C9F2D@TWHCMBS02.tw.ralinktech.ad> Thanks for your response. I am sure that ~RTSPServer is called(even run to ::closeSocket) when I call Medium::Close, but socket still exists (check netstat -a). And I know resources will automatically released by OS even if application does not release them before termination. But I mean I just wonder Live555MediaServer.cpp sample does not do release resources actions before application termination. Maybe it is just a "sample" so it does not need to do those. Therefore, I investigated on Internet everywhere and tried to use Medium::Close, because I don't want to stop RTSP server by application termination for my cause. But it was still invalid, so I posted my part of codes and maybe I need to do more things besides Medium::Close. Additionally, I call Medium::Close when streaming is running. Thanks. Vincent Kao -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson Sent: Tuesday, May 24, 2011 2:15 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] About RTSPServer >I post my source Unfortunately few - if any - people on the mailing list have the time to look through your source code. However, You Have Complete Source Code for the "LIVE555 Streaming Media" libraries. You can see where the RTSP server's socket gets closed (hint: line 220 of "liveMedia/RTSPServer.cpp"), so you should be able to figure out why that is not working for you. >And I wonder why Live555MediaServer.cpp does not delete scheduler >and env objects, >how are these objects released when application is terminated? The ***operating system*** releases all of the application's memory (and sockets), when the application is terminated. Unfortunately, you seem to have a very limited understanding of "processes", and how operating systems handle them. (This will be my last posting on this topic.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel CONFIDENTIALITY STATEMENT : The information, attachments and any rights attaching in this e-mail are confidential and privileged; it is intended only for the individual or entity named as the recipient hereof.Any disclosure, copying, distribution, dissemination or use of the contents of this e-mail by persons other than the intended recipient is STRICTLY PROHIBITED and may violate applicable laws.If you have received this e-mail in error, please delete the original message and notify us by return email or collect call immediately. Thank you. From warren at etr-usa.com Tue May 24 08:25:15 2011 From: warren at etr-usa.com (Warren Young) Date: Tue, 24 May 2011 09:25:15 -0600 Subject: [Live-devel] About RTSPServer In-Reply-To: <42E7479B4A053E44AA8A67EECE348335092C9F2D@TWHCMBS02.tw.ralinktech.ad> References: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9C4E@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9DF4@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9F2D@TWHCMBS02.tw.ralinktech.ad> Message-ID: <4DDBCDDB.3080004@etr-usa.com> On 5/24/2011 12:57 AM, RA-Vincent Kao wrote: > but socket still exists (check netstat -a). If you're seeing a TIME_WAIT state, that's normal and unavoidable: http://tangentsoft.net/wskfaq/articles/debugging-tcp.html If it's a FIN_WAITx or SYN_x state, one of the ends of the connection isn't closing down properly. See the state-transition diagram linked from that article to debug it. If the core problem is that the server can't restart for some time after being stopped due to the TIME_WAIT state, add a setsockopt(SO_REUSEADDR) after the socket() call. Ross, it would be nice to see this in core. I know the standards say there's a risk of misdelivered data if you don't wait the full 2*MSS for rogue packets, but it's a pretty common thing to do in a server. From runet at innovsys.com Tue May 24 10:44:27 2011 From: runet at innovsys.com (Rune Torgersen) Date: Tue, 24 May 2011 17:44:27 +0000 Subject: [Live-devel] About RTSPServer In-Reply-To: <4DDBCDDB.3080004@etr-usa.com> References: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9C4E@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9DF4@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9F2D@TWHCMBS02.tw.ralinktech.ad> <4DDBCDDB.3080004@etr-usa.com> Message-ID: > -----Original Message----- > On 5/24/2011 12:57 AM, RA-Vincent Kao wrote: > > but socket still exists (check netstat -a). > > If you're seeing a TIME_WAIT state, that's normal and unavoidable: > > http://tangentsoft.net/wskfaq/articles/debugging-tcp.html > > If it's a FIN_WAITx or SYN_x state, one of the ends of the > connection > isn't closing down properly. See the state-transition diagram > linked > from that article to debug it. > > If the core problem is that the server can't restart for some time > after > being stopped due to the TIME_WAIT state, add a > setsockopt(SO_REUSEADDR) > after the socket() call. Or call shutdown() on the socket before the close. From p.zebelloni at c-labs.it Tue May 24 13:53:00 2011 From: p.zebelloni at c-labs.it (Paolo Zebelloni) Date: Tue, 24 May 2011 22:53:00 +0200 Subject: [Live-devel] MediaServer Message-ID: <201105242253.00637.p.zebelloni@c-labs.it> Scenario: an ARM-Linux box with 2 ETH (1on "external" network, 1 on "internal" network), must open a RTSP session on the "external" network and make this session available to several RTSP clients on the "internal" network. On the "external" network we shall have just one stream, on the "internal" one, several concurrent streams. How I can bind OpenRTSP and MediaServer together? I made some tries with named pipes, without success. From Priya_Raghavendra at mindtree.com Tue May 24 22:54:26 2011 From: Priya_Raghavendra at mindtree.com (Priya Raghavendra) Date: Wed, 25 May 2011 05:54:26 +0000 Subject: [Live-devel] Live streaming stops after half an hour Message-ID: Hi, On further testing I noticed that the stream works for 35 minutes, stops for 35 minutes and again recovers after 35 minutes. And this happens continuously. What could be the problem? Could there be some overflow of some timing parameters? Please help me debug this issue. Regards, Priya From: Priya Raghavendra Sent: Tuesday, May 24, 2011 11:53 AM To: 'live-devel at lists.live555.com' Subject: Live streaming stops after half an hour Hi, We have been facing this problem of live streaming stopping exactly after half an hour. The same code works on Linux OS but windows this issue is there. The readData function does not get called, And it does not get into the loop isCurrentlyAwaitingData. So incomingDataHandler1 itself is not getting called. Under what conditions will live stop calling incomingDataHandler1 ? void LiveInputSource:: incomingDataHandler1() { int ret; if (!isCurrentlyAwaitingData()) { printf("isCurrentlyAwaitingData\n"); return ; } ret = readData(); if (ret < 0 ) { handleClosure (this); printf("handleClosure (this)\n"); } else if (ret == 0 ) { int uSecsToDelay = 50000 ; nextTask() = envir().taskScheduler().scheduleDelayedTask (uSecsToDelay, (TaskFunc*)incomingDataHandler, this); } else { nextTask() = envir().taskScheduler().scheduleDelayedTask (0, (TaskFunc*)afterGetting, this); } } int LiveInputVideoSource:: readData() { /* Get frame from queue */ queue_get (frame, &frame_len); if (frame_len == 0) { return -1; } /* Copy frame to fTo */ memcpy (fTo, frame, frame_len); fFrameSize = frame_len; /* Presentation Time */ gettimeofday (&fPresentationTime); return 1; } Thanks, Priya ________________________________ http://www.mindtree.com/email/disclaimer.html -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastien-devel at celeos.eu Tue May 24 23:58:14 2011 From: sebastien-devel at celeos.eu (=?ISO-8859-1?Q?S=E9bastien?= Escudier) Date: Wed, 25 May 2011 08:58:14 +0200 Subject: [Live-devel] bug with Authentication Message-ID: <1306306694.8917.4.camel@stim-desktop> Hi There seems to be a bug which crash the library with athentication. The bug can be reproduced with "openRTSP" and this url : rtsp://212.199.182.21/crashme the backtrace is : #0 rawmemchr () at ../sysdeps/i386/rawmemchr.S:116 #1 0x0017ac07 in _IO_str_init_static_internal (sf=0xbf8c0258, ptr=0x0, size=0, pstart=0x0) at strops.c:45 #2 0x0016e7b3 in _IO_vsscanf (string=0x0, format=0x807b990 "Digest realm=\"%[^\"]\", nonce=\"%[^\"]\"", args=0xbf8c0328 "") at iovsscanf.c:44 #3 0x0015d71b in __sscanf (s=0x0, format=0x807b990 "Digest realm=\"%[^\"]\", nonce=\"%[^\"]\"") at sscanf.c:34 #4 0x08053e5b in RTSPClient::handleAuthenticationFailure (this=0x9959720, paramsStr=0x0) at RTSPClient.cpp:1110 #5 0x08057488 in RTSPClient::handleResponseBytes (this=0x9959720, newBytesRead=160) at RTSPClient.cpp:1521 #6 0x080577a3 in RTSPClient::incomingDataHandler1 (this=0x9959720) at RTSPClient.cpp:1310 #7 0x080577cb in RTSPClient::incomingDataHandler (instance=0x9959720) at RTSPClient.cpp:1303 #8 0x0807785a in BasicTaskScheduler::SingleStep (this=0x9959008, maxDelayTime=0) at BasicTaskScheduler.cpp:130 #9 0x08078e97 in BasicTaskScheduler0::doEventLoop (this=0x9959008, watchVariable=0x0) at BasicTaskScheduler0.cpp:80 #10 0x0804c40a in main (argc=2, argv=0xbf8c0834) at playCommon.cpp:509 Best regards, Sebastien. From Bruno.Basilio at brisa.pt Wed May 25 01:18:49 2011 From: Bruno.Basilio at brisa.pt (Bruno Filipe Basilio) Date: Wed, 25 May 2011 08:18:49 +0000 Subject: [Live-devel] Live streaming stops after half an hour Message-ID: <3EC6EEDBCBD4AC4A9FBE5F1F5EEA861904497F@SRVBRIEXC012.brisa.pt> Hi Praya, >> We have been facing this problem of live streaming stopping exactly after half an hour. >> The same code works on Linux OS but windows this issue is there. > On further testing I noticed that the stream works for 35 minutes, stops for 35 minutes > and again recovers after 35 minutes. And this happens continuously. Have you tried the latest version of the live555 library? Because I had a similar problem and it has been solved, please see the following link for details: http://lists.live555.com/pipermail/live-devel/2010-December/012949.html > What could be the problem? > Could there be some overflow of some timing parameters? I think the following changelog line resumes the solution. ? 2010.12.05: - Added a sanity check to "MultiFramedRTPSink" and "BasicUDPSink" to allow for the possibility of the system clock jumping ahead in time, and thereby messing up the calculation of how long to wait before sending the next packet. (Thanks to Anders Chen for noting this issue.) ? Best regards, Bruno Basilio Brisa Inova??o e Tecnologia, S.A. -------------------------------------------------------------------------------- Declara??o: A informa??o contida nesta mensagem, e os ficheiros anexos, ? privilegiada e confidencial, destinando-se exclusivamente ao(s) destinat?rio(s).Se n?o ? o destinat?rio (ou o respons?vel pela sua entrega ao destinat?rio) e recebeu a mesma por engano, fica notificado que ? estritamente proibido reproduzir, guardar ou distribuir toda ou qualquer parte desta mensagem e ficheiros anexos.Por favor reencaminhe a mensagem para o respons?vel pelo seu envio ou contacte-nos por telefone e elimine a mensagem e ficheiros anexos do seu computador,sem os reproduzir. Disclaimer: The information contained in this message, and any files attached, is privileged and confidential, and intended exclusively for the included addresses.If you are not the intended recipient (or the person responsible for delivering to the intended recipient) and received this message by mistake, be aware that copy, storage, distribution or any other use of all or part of this message and the files attached is strictly prohibited. Please notify the sender by reply e-mail or contact us by telephone and delete this message and the files attached, without retaining a copy. -------------------------------------------------------------------------------- From warren at etr-usa.com Wed May 25 08:09:47 2011 From: warren at etr-usa.com (Warren Young) Date: Wed, 25 May 2011 09:09:47 -0600 Subject: [Live-devel] About RTSPServer In-Reply-To: References: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9C4E@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9DF4@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9F2D@TWHCMBS02.tw.ralinktech.ad> <4DDBCDDB.3080004@etr-usa.com> Message-ID: <4DDD1BBB.6010702@etr-usa.com> On 5/24/2011 11:44 AM, Rune Torgersen wrote: >> add a >> setsockopt(SO_REUSEADDR) >> after the socket() call. > > Or call shutdown() on the socket before the close. No, shutdown() will not let you skip past TIME_WAIT. The only thing that will is slamming the connection shut with RST, and that's no good solution at all. I've only ever done that when writing code to interoperate with a broken peer which I cannot fix. From warren at etr-usa.com Wed May 25 08:14:40 2011 From: warren at etr-usa.com (Warren Young) Date: Wed, 25 May 2011 09:14:40 -0600 Subject: [Live-devel] About RTSPServer In-Reply-To: <42E7479B4A053E44AA8A67EECE348335092C9FBC@TWHCMBS02.tw.ralinktech.ad> References: <42E7479B4A053E44AA8A67EECE348335092C9B75@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9C4E@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9DF4@TWHCMBS02.tw.ralinktech.ad> <42E7479B4A053E44AA8A67EECE348335092C9F2D@TWHCMBS02.tw.ralinktech.ad> <4DDBCDDB.3080004@etr-usa.com> <42E7479B4A053E44AA8A67EECE348335092C9FBC@TWHCMBS02.tw.ralinktech.ad> Message-ID: <4DDD1CE0.5020502@etr-usa.com> On 5/24/2011 9:14 PM, RA-Vincent Kao wrote: > netstat -a show socket state as ESTABLISHED. > So that means RTSP server does not send FIN bit and state does not transit to FIN_WAIT_1? Yes. That means that either a) you are misdiagnosing this, or b) neither closesocket() nor shutdown() is being called on that connection's socket. P.S. Please keep replies on the list. From finlayson at live555.com Wed May 25 11:03:46 2011 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 25 May 2011 12:03:46 -0600 Subject: [Live-devel] bug with Authentication In-Reply-To: <1306306694.8917.4.camel@stim-desktop> References: <1306306694.8917.4.camel@stim-desktop> Message-ID: >There seems to be a bug which crash the library with athentication. >The bug can be reproduced with "openRTSP" and this url : >rtsp://212.199.182.21/crashme Thanks for the bug report. The problem was caused by the server returning a 401 response, but without a "WWW-Authenticate:" header. The client was not handling this properly. I have now installed a new version (2011.05.25) of the "LIVE555 Streaming Media" code that fixes this. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From Brad.Lackey at schneider-electric.com Thu May 26 17:56:54 2011 From: Brad.Lackey at schneider-electric.com (Lackey, Brad) Date: Thu, 26 May 2011 17:56:54 -0700 Subject: [Live-devel] How to specify an rtp client without rtsp Message-ID: Hello Everyone! I've been playing with Live555 for a while now and it seems pretty awesome, good job guys. By 'playing' I mean I've developed around live555 to send out an rtp stream from a live source (mpeg4, mjpeg, h264, etc) all via rtsp, and it works wonderfully. The problem I'm having is I want to be able to specify a client listening on a certain port and have live555 send a video stream to the specified computer and port WITHOUT having to go through the RTSP process. So I'm hoping to do the following? 1. Get a computers ip and port where they are listening for rtp packets. 2. I would give live555 the client's ip and port and specify a stream 3. live555 would send send the rtp stream out to the client no questions asked, no need for rtsp 4. happy times =] Thanks for the help! ______________________________________________________________________ Bradley Lackey | Pelco by Schneider Electric | Buildings Business | United States | Software Engineer II Phone: +559.292.1981 | Toll Free: +800.289.9100 ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From sokratis.barmpounakis at unige.ch Thu May 26 02:48:34 2011 From: sokratis.barmpounakis at unige.ch (Sokratis Barmpounakis) Date: Thu, 26 May 2011 11:48:34 +0200 Subject: [Live-devel] cannot view rtsp stream with vlc Message-ID: Hello, when I am opening vlc to view a rtsp stream (raw h264 video) from the rtsp server I can watch it with no problem. (I just open rtsp://x.x.x.x:8554/streamName) But when I am trying that from another computer connected directly to the same router as the server, vlc can open the stream with success, however the video won't play. No error messages in the vlc messages window. And now incoming data from the stream. Any ideas? Thank you in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: From renaud.guezennec.ext at valeo.com Fri May 27 05:08:17 2011 From: renaud.guezennec.ext at valeo.com (Renaud GUEZENNEC) Date: Fri, 27 May 2011 14:08:17 +0200 Subject: [Live-devel] subsession->readSource() is 0x0 Message-ID: Hi all, I'm trying to use live555 to make an rtsp client. So, I think i'm really close to succeed but I'm getting errors. When I call > subsession->sink->startPlaying(*(subsession->readSource()), > subsessionAfterPlaying, subsession); > subsession->readSource() returns a null pointer so, how can make it right ? I implemented my own sink (My goal is to store data into a Qt contener, to perform rendering into Qt application) Am i missing something ? Regards Renaud This e-mail message is intended only for the use of the intended recipient(s). The information contained therein may be confidential or privileged, and its disclosure or reproduction is strictly prohibited. If you are not the intended recipient, please return it immediately to its sender at the above address and destroy it. -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Mon May 30 20:33:49 2011 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 30 May 2011 20:33:49 -0700 Subject: [Live-devel] cannot view rtsp stream with vlc In-Reply-To: References: Message-ID: >Hello, >when I am opening vlc to view a rtsp stream (raw h264 video) from >the rtsp server I can watch it with no problem. (I just open >rtsp://x.x.x.x:8554/streamName) > >But when I am trying that from another computer connected directly >to the same router as the server, vlc can open the stream with >success, however the video won't play. No error messages in the vlc >messages window. And now incoming data from the stream. > >Any ideas? Something in your network (either your router, or a firewall somewhere) is blocking UDP packets. In any case, I suggest testing first with the "openRTSP" command-line RTSP client , rather than VLC. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From TWiser at logostech.net Tue May 31 12:14:55 2011 From: TWiser at logostech.net (Wiser, Tyson) Date: Tue, 31 May 2011 12:14:55 -0700 Subject: [Live-devel] Possible bugs in ByteStreamFileSource and MPEG2TransportStreamFramer Message-ID: <8CD7A9204779214D9FDC255DE48B952150CDF897@EXPMBX105-1.exch.logostech.net> I am using the live555 library in a Linux application that receives a low frame rate MPEG2-TS from a live source one frame at a time. The source is not represented as a file by the OS. To make the interface to the live555 library easier, I opened a pipe and write each frame to it as I receive them. The read end of the pipe is given as the source to a ByteStreamFileSource instance which then feeds an instance of MPEG2TransportStreamFramer, which in turn feeds a BasicUDPSink (unfortunately my requirements do not allow me to use RTP/RTCP, which I would have preferred). This seems to work fairly well with the following exceptions. 1. BasicUDPSink has a default packet size of 1450. MPEG2TransportStreamFramer asks its source (ByteStreamFileSource in this case) for 1450 bytes and then, in afterGettingFrame1, correctly limits the actual packet size to 1316 (7 188-byte TS packets). However, it seems to discard the remaining 134 bytes (or less if less than 1450 bytes were read), which causes sync problems on almost every packet after that and wreaks havoc on the receiving end of the stream. If I change the default packet size in BasicUDPSink to 1316, these problems go away. This discarding of data seems like a bug to me, but I may just be using it incorrectly. 2. Since data is given to me and thus written to the pipe frame-by-frame, it is very common that a single write to the pipe will not be a multiple of 7 188-byte TS packets (e.g. a frame might be 25192 bytes = 134 188-byte TS packets = 19 UDP packets containing 7 TS packets each + 1 UDP packet containing 1 TS packet). While the MPEG2-TS source continues to produce frames this does not cause any problems. However, when the source is paused/stopped it can cause the event loop to hang indefinitely. This is due to the fact that ByteStreamFileSource::doReadFromFile is trying to read more data than it can receive and thus blocks in the "fFrameSize = fread(fTo, 1, fMaxSize, fFid);" line. A fix that worked for me was to replace that line with "fFrameSize = read(fileno(fFid), fTo, fMaxSize);". Since I am only working in Linux, this fix may not be cross-platform without some massaging. Again, I consider this a bug because it doesn't behave the way I think it should, but you may never have intended ByteStreamFileSource to be used with a pipe as a source. Thanks for the great work on this library. Tyson Wiser -------------- next part -------------- An HTML attachment was scrubbed... URL: From Szoke at BeeComputing.com Mon May 30 06:09:59 2011 From: Szoke at BeeComputing.com (Szoke) Date: Mon, 30 May 2011 15:09:59 +0200 Subject: [Live-devel] VLC stops working due to live555 breakdown after 2147 seconds Message-ID: <4DE39727.6040506@BeeComputing.com> Hi everyone, I have started a discussion on the VLC forum, which explains the problem I have with Live555: http://forum.videolan.org/viewtopic.php?f=14&t=90813 In short: 1. I am catching a rtsp stream from an IP camera attached to a kestrel nest. For those who are interested in birds, just for fun, have a look here: http://www.justin.tv/szokeszaniszlo/videos?frh=1306760409 this system is also in two embedded systems used in the woods, catching amazing images from black storks : http://www.solon.be/3-combat-violent-entre-la-cigogne-et-la-martre-violent-struggle-between-black-stork-and-pine-marten/ 2. Unfortunately, VLC stops working after about 35 minutes. Debugging VLC showed that the stream is broken by live555 after exactly 2147 seconds, or 2^31, which sounds like a 32 bit signed integer being used to store micro seconds... All details are given on the VLC forum at the URL given above. 3. In embedded systems using linutop computers (running Ubuntu), VLC stops reading the stream for the same reason. Using different picture resolution doesn't help. We could bypass the problem by restarting VLC every 15 minutes. But under Windows, I cannot restart VLC as the Dyyno Universal Broadcaster sending the images live to justin.tv looses the source image in that case. 4. I'm doing my best to have all these systems up and running full time, as even one national television broadcast (TV5) is following these videos. So any help will be greatly appreciated ! Thanks a lot in advance, Stan From wew at bearnet.com Tue May 31 10:55:46 2011 From: wew at bearnet.com (Bill Weinman) Date: Tue, 31 May 2011 10:55:46 -0700 Subject: [Live-devel] RTSP/RTP streaming from camera Message-ID: <4DE52BA2.5010500@bearnet.com> I'm writing an application to stream from a camera to a wowza server for re-broadcast. I have most of it working. I have encoded video (x264) and audio (aac) frames -- I'm using ffmpeg's libav libraries for this -- and am successfully communicating with the wowza server using RTSP. I'm sending ANNOUNCE and SETUP requests and getting the expected responses (this part is easy -- it's a simple protocol). Where I'm stuck is with RTP. I have the results from my RTSP SETUP request, and I have video and audio frames, now I need to establish the RTP session, create the streams, and send them to the server. I see that the live555 code has classes like H264VideoRTPSource and DynamicRTSPServer and these look helpful. I'm just not seeing clearly where to start. Any suggestions? Thanks! //Bill From nacho at vartificial.com Mon May 30 03:07:06 2011 From: nacho at vartificial.com (Ignacio Rodriguez Torres) Date: Mon, 30 May 2011 12:07:06 +0200 Subject: [Live-devel] openRTSP h264 decoding Message-ID: Hello, first of all thank you all for maintaining this forum so active I am using openRTSP to recieve a stream with a .sdp this stream contains video and audio streams, but I?m only interested in the video, which is encoded with h264, when i am trying to decode with ffmpeg I obtain NAL errors. I also tried to write the stream in a file with FileSink and open it with vlc or ffmpeg but i obtained the same result, ffmpeg -i "video-H264-1" video-H264-1: Unknown format I saw that H264VideoFileSink put the header 0x00000001 in each frame but seems that don?t work properly. I read the FAQ and some post in the mailing list and I saw similar problems in other persons, Can someone help me? Thank you very much. Attached the trace of execution Opening connection to 10.138.255.8, port 554... ...remote connection opened Sending request: OPTIONS rtsp:// 10.138.255.8/broadcast/mediapro/vivo/goltv/mediapro_vivo_goltv_h264_1.sdpRTSP/1.0 CSeq: 2 User-Agent: ./openRTSP (LIVE555 Streaming Media v2010.12.05) Received 175 new bytes of response data. Received a complete OPTIONS response: RTSP/1.0 200 OK CSeq: 2 Date: Mon, 30 May 2011 09:55:48 GMT Public: OPTIONS, DESCRIBE, SETUP, PLAY, PAUSE, TEARDOWN Server: ERICSSON EMTV 1.0 Via: RTSP/1.0 bgustrnin2 Sending request: DESCRIBE rtsp:// 10.138.255.8/broadcast/mediapro/vivo/goltv/mediapro_vivo_goltv_h264_1.sdpRTSP/1.0 CSeq: 3 User-Agent: ./openRTSP (LIVE555 Streaming Media v2010.12.05) Accept: application/sdp Received 1024 new bytes of response data. Received a complete DESCRIBE response: RTSP/1.0 200 OK Content-Base: rtsp:// 10.138.255.8/broadcast/mediapro/vivo/goltv/mediapro_vivo_goltv_h264_1.sdp/ Content-Language: en-US Content-Length: 675 Content-Type: application/sdp CSeq: 3 Date: Mon, 30 May 2011 09:55:49 GMT Public: OPTIONS, DESCRIBE, SETUP, PLAY, PAUSE, TEARDOWN Server: ERICSSON EMTV 1.0 Via: RTSP/1.0 bgustrnin2 v=0 o=Ericsson 3515738112 3515738112 IN IP4 10.132.149.44 s=mediapro_vivo_goltv_h264_1.sdp c=IN IP4 0.0.0.0 t=0 0 a=control:* m=video 0 RTP/AVP 105 b=AS:222 a=control:streamid=1 a=range:npt=0- a=Width:integer;320 a=Height:integer;240 a=cliprect:0,0,240,320 a=rtpmap:105 H264/90000 a=fmtp:105 profile-level-id=42E00C; sprop-parameter-sets=Z0LgDJZUCg/I,aM4BrFCA; packetization-mode=1 a=mpeg4-esid:11 a=3GPP-Adaptation-Support:1 m=audio 0 RTP/AVP 110 b=AS:20 a=control:streamid=2 a=range:npt=0- a=rtpmap:110 MP4A-LATM/32000/1 a=mpeg4-esid:12 a=fmtp:110 profile-level-id=15;object=2;cpresent=0;config=400028103FC0;SBR-enabled=1 a=3GPP-Adaptation-Support:1 Opened URL "rtsp:// 10.138.255.8/broadcast/mediapro/vivo/goltv/mediapro_vivo_goltv_h264_1.sdp", returning a SDP description: v=0 o=Ericsson 3515738112 3515738112 IN IP4 10.132.149.44 s=mediapro_vivo_goltv_h264_1.sdp c=IN IP4 0.0.0.0 t=0 0 a=control:* m=video 0 RTP/AVP 105 b=AS:222 a=control:streamid=1 a=range:npt=0- a=Width:integer;320 a=Height:integer;240 a=cliprect:0,0,240,320 a=rtpmap:105 H264/90000 a=fmtp:105 profile-level-id=42E00C; sprop-parameter-sets=Z0LgDJZUCg/I,aM4BrFCA; packetization-mode=1 a=mpeg4-esid:11 a=3GPP-Adaptation-Support:1 m=audio 0 RTP/AVP 110 b=AS:20 a=control:streamid=2 a=range:npt=0- a=rtpmap:110 MP4A-LATM/32000/1 a=mpeg4-esid:12 a=fmtp:110 profile-level-id=15;object=2;cpresent=0;config=400028103FC0;SBR-enabled=1 a=3GPP-Adaptation-Support:1 Created receiver for "video/H264" subsession (client ports 38330-38331) Created receiver for "audio/MP4A-LATM" subsession (client ports 45668-45669) Sending request: SETUP rtsp:// 10.138.255.8/broadcast/mediapro/vivo/goltv/mediapro_vivo_goltv_h264_1.sdp/streamid=1RTSP/1.0 CSeq: 4 User-Agent: ./openRTSP (LIVE555 Streaming Media v2010.12.05) Transport: RTP/AVP;unicast;client_port=38330-38331 Received 298 new bytes of response data. Received a complete SETUP response: RTSP/1.0 200 OK CSeq: 4 Date: Mon, 30 May 2011 09:55:50 GMT Public: OPTIONS, DESCRIBE, SETUP, PLAY, PAUSE, TEARDOWN Server: ERICSSON EMTV 1.0 Session: 1306749349803963000-38658;timeout=80 Transport: RTP/AVP;unicast;client_port=38330-38331;server_port=20150-20151 Via: RTSP/1.0 bgustrnin2 Setup "video/H264" subsession (client ports 38330-38331) Sending request: SETUP rtsp:// 10.138.255.8/broadcast/mediapro/vivo/goltv/mediapro_vivo_goltv_h264_1.sdp/streamid=2RTSP/1.0 CSeq: 5 User-Agent: ./openRTSP (LIVE555 Streaming Media v2010.12.05) Transport: RTP/AVP;unicast;client_port=45668-45669 Session: 1306749349803963000-38658 Received 287 new bytes of response data. Received a complete SETUP response: RTSP/1.0 200 OK CSeq: 5 Date: Mon, 30 May 2011 09:55:51 GMT Public: OPTIONS, DESCRIBE, SETUP, PLAY, PAUSE, TEARDOWN Server: ERICSSON EMTV 1.0 Session: 1306749349803963000-38658 Transport: RTP/AVP;unicast;client_port=45668-45669;server_port=20154-20155 Via: RTSP/1.0 bgustrnin2 Setup "audio/MP4A-LATM" subsession (client ports 45668-45669) Created output file : "video-H264-1" Sending request: PLAY rtsp:// 10.138.255.8/broadcast/mediapro/vivo/goltv/mediapro_vivo_goltv_h264_1.sdp/RTSP/1.0 CSeq: 6 User-Agent: ./openRTSP (LIVE555 Streaming Media v2010.12.05) Session: 1306749349803963000-38658 Range: npt=0.000- Received 555 new bytes of response data. Received a complete PLAY response: RTSP/1.0 200 OK Content-Base: rtsp:// 10.138.255.8/broadcast/mediapro/vivo/goltv/mediapro_vivo_goltv_h264_1.sdp/ CSeq: 6 Date: Mon, 30 May 2011 09:55:51 GMT Public: OPTIONS, DESCRIBE, SETUP, PLAY, PAUSE, TEARDOWN RTP-Info: url=rtsp:// 10.138.255.8/broadcast/mediapro/vivo/goltv/mediapro_vivo_goltv_h264_1.sdp/streamid=1;seq=0;rtptime=1000000,url=rtsp://10.138.255.8/broadcast/mediapro/vivo/goltv/mediapro_vivo_goltv_h264_1.sdp/streamid=2;seq=0;rtptime=1000000 Server: ERICSSON EMTV 1.0 Session: 1306749349803963000-38658 Via: RTSP/1.0 bgustrnin2 Started playing session Receiving streamed data (signal with "kill -HUP 3638" or "kill -USR1 3638" to terminate)... [h264 @ 0x2cf1550] AVC: nal size 1121979542 [h264 @ 0x2cf1550] no frame! [h264 @ 0x2cf1550] AVC: nal size -838751152 [h264 @ 0x2cf1550] no frame! [h264 @ 0x2cf1550] AVC: nal size -2004873472 [h264 @ 0x2cf1550] concealing 300 DC, 300 AC, 300 MV errors [h264 @ 0x2cf1550] AVC: nal size -1711143936 [h264 @ 0x2cf1550] concealing 282 DC, 282 AC, 282 MV errors [h264 @ 0x2cf1550] AVC: nal size -1711011839 [h264 @ 0x2cf1550] concealing 300 DC, 300 AC, 300 MV errors [h264 @ 0x2cf1550] AVC: nal size -1710879743 [h264 @ 0x2cf1550] concealing 300 DC, 300 AC, 300 MV errors [h264 @ 0x2cf1550] AVC: nal size -1710747647 [h264 @ 0x2cf1550] concealing 300 DC, 300 AC, 300 MV errors [h264 @ 0x2cf1550] AVC: nal size -1710615551 [h264 @ 0x2cf1550] concealing 300 DC, 300 AC, 300 MV errors [h264 @ 0x2cf1550] AVC: nal size -1710483455 [h264 @ 0x2cf1550] concealing 300 DC, 300 AC, 300 MV errors [h264 @ 0x2cf1550] AVC: nal size -1710351359 [h264 @ 0x2cf1550] concealing 300 DC, 300 AC, 300 MV errors -------------- next part -------------- An HTML attachment was scrubbed... URL: From digitalqtum at gmail.com Wed May 18 21:39:23 2011 From: digitalqtum at gmail.com (Digital Qtum) Date: Thu, 19 May 2011 04:39:23 -0000 Subject: [Live-devel] how to detect frame loss Message-ID: Hello, I?m doing this almost 2 weeks it is somewhat difficult to me.. My embedded board receive h.264 baseline stream from IP camera. During the system overhead or temporary network disconnection, P frame (or I frame) is lost, and then screen display breaks until next I frame received. So, I wanna detect (P) frame loss and skip P frames to next I frame. Can this be done by live555? Or need some application side process? To summarise 1. How can I detect P frame loss? 2. If above is possible, where should it be done to skip P frames? (Live555 or application) Any advice will be welcomed. -------------- next part -------------- An HTML attachment was scrubbed... URL: From hongfeng_wang at yahoo.com Wed May 18 07:57:28 2011 From: hongfeng_wang at yahoo.com (Hongfeng WANG) Date: Wed, 18 May 2011 14:57:28 -0000 Subject: [Live-devel] testH264VideoStreamer not working properly Message-ID: <574948.81249.qm@web81306.mail.mud.yahoo.com> It's compiled on Ubuntu 11.04/x86 machine, the video is tc10.264 downloaded from the website. Now when the test program runs, client can not view it. I tried Movie player of Ubuntu, it complains about "stream in wrong format", VLC on windows 7 will flicker for seconds and crash. Since this is provided as example, it should be working fine out of the box. Did I miss something? FYI: the live555MediaServer streams same video just fine. Thanks. Hongfeng Wang -------------- next part -------------- An HTML attachment was scrubbed... URL: From infinitebuzz at gmail.com Thu May 26 18:56:24 2011 From: infinitebuzz at gmail.com (James Brooks) Date: Thu, 26 May 2011 21:56:24 -0400 Subject: [Live-devel] config.macosx change Message-ID: Hi All I could not get the macos version to compile with LIBRARY_LINK = ar cr so I used LIBRARY_LINK = libtool -s -o which seems to work fine. Cheers, James -- James Brooks, Manager R&D VVW Tel: +1 416 255 5636, ext 226 http://www.drastictech.com/contacts/james.vcf -------------- next part -------------- An HTML attachment was scrubbed... URL: From jafrado at gmail.com Thu May 12 15:45:14 2011 From: jafrado at gmail.com (James Dougherty) Date: Thu, 12 May 2011 22:45:14 -0000 Subject: [Live-devel] how to capture H264 stream and save it in a avi file In-Reply-To: References: Message-ID: Hi Gianluigi, Please provide the -i option and review the options on the main site, below: http://www.live555.com/openRTSP/#quicktime Those options all work for me, and very well! Here is another recipe I use for additional cross-validation. If you have L16 media, put a riff header on it: sox -B -t raw -s -b 16 -c 2 -r 48k audio-L16-2 audio-L16-2.wav For MP4 (openRTSP -4 option): mp4box -fps 30 -add video-H264-1.264 -add audio-L16-2.wav -new myfile.mp4 For AVI (openRTSP -i option) ffmpeg -r 30 -i video-H264-1 -i audio-L16-2.wav -vcodec copy -acodec copy myfile.avi I use the openRTSP options, but at times, I have used the above to verify I didn't break something :-) YMMV best regards -james On Wed, May 11, 2011 at 6:53 AM, Autuori Gianluigi < Autuori.Gianluigi.Wintime at ansaldobreda.it> wrote: > Hello, > I'm trying to capture a video stream from Axis IP camera with openRTSP. > I get a correct file if I use: > openRTSP -d 20 -4 -f 3 -w 640 -h 480 -b 400000 "rtsp:// > 10.10.1.61/axis-media/media.amp?compression=35&fps=3" >video.avi > but I try this: > openRTSP -d 20 -f 3 -w 640 -h 480 -b 400000 "rtsp:// > 10.10.1.61/axis-media/media.amp?compression=35&fps=3" > I obtain this file: video-H264-1 > but I can't open it. > What is wrong? > I read this post: > http://lists.live555.com/pipermail/live-devel/2007-January/005886.html > is it the same problem? > > thanks > Gianluigi > > ------------------------------ > Questo messaggio e-mail e ogni documento ad esso eventualmente allegato > puo' avere carattere riservato ed essere tutelato da segreto. Esso,comunque, > e' > ad esclusivo utilizzo del destinatario in indirizzo. Qualora non foste il > destinatario del messaggio vi preghiamo di volerci avvertire immediatamente > per e-mail o telefono e di cancellare il presente messaggio e ogni eventuale > allegato dal vostro sistema. E' vietata la duplicazione o l'utilizzo per > qualunque fine del messaggio e di ogni allegato, nonche' la loro > divulgazione, distribuzione o inoltro a terzi senza l'espressa > autorizzazione del mittente. In ragione del mezzo di trasmissione > utilizzato, il mittente non assume alcuna responsabilita' sulla > segretezza/riservatezza delle informazioni contenute nel messaggio e nei > relativi allegati. > > This e-mail and any file transmitted with it may contain material that is > confidential, privileged and/or attorney work product for the sole use of > the intended recipient. If you are not the intended recipient of this > e-mail, please do not read it, notify us immediately by e-mail or by > telephone and then delete this message and any file attached from your > system. You should not copy or use it for any purpose, disclose the contents > of the same to any other person or forward it without express permission. > Considering the means of transmission, we do not undertake any liability > with respect to the secrecy and confidentiality of the information contained > in this e-mail and its attachments. > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From panuse1123 at gmail.com Sun May 22 00:13:16 2011 From: panuse1123 at gmail.com (nick pan) Date: Sun, 22 May 2011 07:13:16 -0000 Subject: [Live-devel] How to deliver video frames to our codec? Message-ID: Hi,everyone, Sorry,please forgive me not good English, For liveMedia library, I have some questions need to understand. we want to do streaming the H264 video from liveMedia Server. and I want to decode the H264 frames by our platform, and display to component video output. If we don't need to do for h264 vdieo decoding ,but wish to obtain video frame data,and deliver to our codec. so, following is my questions: 1. I wish to obtain video frame data,and directly deliver to our codec, which I need to add or modify that part? (openRTSP example) Do I need go modify H264videoRTPSink::afterGettingFrame ? In that function,Is the data already been decoded or not? or just get data from FileSink and deiver that? 2. How can i control the frames arrived or not? thanks, Best Regards, Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: From socbar at gmail.com Fri May 13 05:53:31 2011 From: socbar at gmail.com (sokratis) Date: Fri, 13 May 2011 12:53:31 -0000 Subject: [Live-devel] building testH264VideoStreamer in MSVS Message-ID: Hi all, I would like to use some parts of the testH264VideoStreamer.cpp code in my app in order to stream over rtp the output after encoding some data. I read in the documentation that I can also change the input of testH264VideoStreamer and instead of .264 files from disk, give directly input from the encoder output (live stream). That's all cool. But. I guess that I have to build all BasicUsageEnvironment, groupsock, liveMedia and UsageEnvironment before using testH264VideoStreamer , right. The problem is that I don't know how to do that in Visual Studio. I need to build it there, as my application is implemented already there. I tried some things but I get a looot of errors. What should be the procedure? Thank you very much in advance. Sokratis -------------- next part -------------- An HTML attachment was scrubbed... URL: From stew.paddaso at gmail.com Fri May 13 17:00:38 2011 From: stew.paddaso at gmail.com (Stew Paddaso) Date: Sat, 14 May 2011 00:00:38 -0000 Subject: [Live-devel] Raw RTP Packet Info Message-ID: Is there a way to get information about the RTP packets that make up each media frame (client-side)? I want to know the sequence number of each packet, the size of each packet, and the time that each one arrived. I plan to write my own MediaSink for "frame" statistics, but I'd also like information about the packets that compose that frame. I've run openRTSP and studied the code. I've also looked through the code for the inheritance tree from H264VideoRTPSource through Medium. I'm not sure if I need to implement an auxiliary read handler for the source or if there's an easier option. Thx From vptruman at gmail.com Wed May 25 12:10:45 2011 From: vptruman at gmail.com (Mark Dean) Date: Wed, 25 May 2011 12:10:45 -0700 Subject: [Live-devel] Integrating OnDemandRTSPServer Message-ID: Was wondering if anyone here had an example or knew if it would be relatively painless to integrate the Live555 RTSP server with an application that ends up with video information in either an openGL PBO or via directShow. Any pointers? Thanks guys. Mark From ydgoo9 at gmail.com Wed May 25 23:21:59 2011 From: ydgoo9 at gmail.com (YD) Date: Thu, 26 May 2011 15:21:59 +0900 Subject: [Live-devel] Question about trickmode with liveMediaserver Message-ID: Hello all, I am testing RTSP/RTP with liveMediaserver with MPEG2-TS. I made index file and the file for trickmode using testMPEG2TransportStreamTrickPlay.exe For example, original contents is avatar.ts index file is avatar.tsx and file for trickplay is avatar2.ts I am using the VLC and my player for test. Seek is OK but I cannot do the fast forward. My question is 1. Is it possible the trickplay on VLC with liveMediaserver ? 2. During normal play(speed 1x), If I send command below, Should server send the stream using avatar2.ts not avatar.ts ? PLAY rtsp://xxx.xxx.xxx.xxx:554/avatar.ts RTSP/1.0 CSeq: 6 Session: 0000532C Scale: 2.000000 User-Agent: ip player (LIVE555 Streaming Media v2006.09.20) Thanks in advance, -------------- next part -------------- An HTML attachment was scrubbed... URL: From zhonghochen at gmail.com Thu May 12 02:34:43 2011 From: zhonghochen at gmail.com (zhongho chen) Date: Thu, 12 May 2011 09:34:43 -0000 Subject: [Live-devel] live555 latency Message-ID: Hi, I use live555 mediaserver and VLC player. When I play a 264 file (slamtv60.264), it gets about 10 sec latency. My VLC play have 1.2 sec latency. Does any other source cause the latency? Thank You, Zhong-Ho