From ali.muzzaffar at verticity.com Mon Aug 4 23:48:42 2008 From: ali.muzzaffar at verticity.com (ali.muzzaffar at verticity.com) Date: Tue, 5 Aug 2008 02:48:42 -0400 (Eastern Daylight Time) Subject: [Live-devel] [Fwd: Required ltl help --- problem in LIVE555] Message-ID: <.202.5.140.130.1217918922.squirrel@mail.verticity.com> Hello, Ross, I am writing because we are currently experimenting with LIVE555 and Darwin Server v5.5.4 I am having problem in streaming mpeg4 to darwin server using testMpeg4toDarwin. I have analyzed the packet through wireshark. and in closer looking we come to know that this Darwin Server closes connection on second Announce Method. On Command prompt we get this error: >testMPEG4VideoToDarwin.exe 192.168.2.63 Beginning streaming... Beginning to read from file... injector->setDestination() failed: cannot handle ANNOUNCE response: RTSP/1.0 412 ?Precondition Failed We have searched the internet thorougly but was not able to find an answer which can resolve our issue. Let me also tell you that we have allowed user in qtaccess file write permissions. We have similar problem which I think you have faced in 2002. The link is given below http://lists.apple.com/archives/Streaming-server-dev/2002/May/msg00104.html It is I think problem Digest authentication.....but I am still not sure......what should I do to make it work. Thanks in Advance. Ali Muzzaffar Software Engineer Verticity.com From Melvin_Raj at satyam.com Tue Aug 5 03:04:13 2008 From: Melvin_Raj at satyam.com (Melvin_Raj) Date: Tue, 5 Aug 2008 18:04:13 +0800 Subject: [Live-devel] Help with link error Message-ID: hello....im new to this forum...im trying to build live555's test prog, specificly the openrtsp...i am able to build the mpegsender,mp3sender and receiver respectively witout any errors but when i try to build openrtsp, i get the following errors: 1>------ Rebuild All started: Project: rtsp, Configuration: Debug Win32 ------ 1>Deleting intermediate and output files for project 'rtsp', configuration 'Debug|Win32' 1>Compiling... 1>openRTSP.cpp 1>Compiling manifest to resources... 1>Linking... 1>openRTSP.obj : error LNK2001: unresolved external symbol "unsigned short tunnelOverHTTPPortNum" (?tunnelOverHTTPPortNum@@3GA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "unsigned int statusCode" (?statusCode@@3IA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double duration" (?duration@@3NA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double scale" (?scale@@3NA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double initialSeekTime" (?initialSeekTime@@3NA) 1>liveMedia.lib(RTSPClient.obj) : error LNK2019: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) referenced in function "char * __cdecl createScaleString(float,float)" (?createScaleString@@YAPADMM at Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2001: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) 1>liveMedia.lib(RTSPCommon.obj) : error LNK2001: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) 1>liveMedia.lib(RTSPClient.obj) : error LNK2019: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) referenced in function "char * __cdecl createScaleString(float,float)" (?createScaleString@@YAPADMM at Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2001: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) 1>liveMedia.lib(RTSPCommon.obj) : error LNK2001: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2019: unresolved external symbol "public: static class H263plusVideoRTPSource * __cdecl H263plusVideoRTPSource::createNew(class UsageEnvironment &,class Groupsock *,unsigned char,unsigned int)" (?createNew at H263plusVideoRTPSource@@SAPAV1 at AAVUsag eEnvironment@@PAVGroupsock@@EI at Z) referenced in function "public: unsigned int __thiscall MediaSubsession::initiate(int)" (?initiate at MediaSubsession@@QAEIH at Z) 1>MSVCRTD.lib(crtexe.obj) : error LNK2019: unresolved external symbol _main referenced in function ___tmainCRTStartup 1>C:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\Debug\rtsp.exe : fatal error LNK1120: 9 unresolved externals 1>Build log was saved at "file://c:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\rtsp\Debug\BuildLog.htm" 1>rtsp - 14 error(s), 0 warning(s) ========== Rebuild All: 0 succeeded, 1 failed, 0 skipped i've already added the required lib: wsock32.lib BasicUsageEnvironment.lib groupsock.lib liveMedia.lib UsageEnvironment.lib any help in pointing me towards the proper direction will be appreciated [cid:image001.gif at 01C8F723.C1B5C3D0] thank you in advance....God bless... ps: im trying to build an rtp receiver that will be able to play streaming files from a server... ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 174 bytes Desc: image001.gif URL: From icotoi at rolabs.com Tue Aug 5 06:59:19 2008 From: icotoi at rolabs.com (Ionut Cotoi) Date: Tue, 05 Aug 2008 16:59:19 +0300 Subject: [Live-devel] Win32 SendingInterfaceAddr Message-ID: <48985CB7.1010804@rolabs.com> Hi, I am trying to select the multicast output interface on windows by setting the SendingInterfaceAddr variable. The result is that Groupsock will not send packets on the network after this. The sendto() in GroupsockHelper.cpp :: witeSocket() always returns -1. On Unix it works like a charm (I tried it on Mac OSX 10.5.4). Do you have any ideas on this behaviour? Best regards, Ionut Cotoi From roland at wingmanteam.com Tue Aug 5 11:24:17 2008 From: roland at wingmanteam.com (Roland) Date: Tue, 5 Aug 2008 11:24:17 -0700 Subject: [Live-devel] Help with link error In-Reply-To: References: Message-ID: <000a01c8f728$789f7dd0$69de7970$@com> Hi Melvin Do a FindInFiles for the symbols that aren't being resolved, e.g. "tunnelOverHTTPPortNum" or "statusCode". Then figure out if that source file where these variables are being defined is included in the compilation of the .LIB or the final .EXE. Add the corresponding source file to one of those projects and recompile. E.g. for the "tunnelOverHTTPPortNum" variable, you'll find "extern" references and one real declaration in testProgs\playCommon.cpp. Add playCommon.cpp to resolve this issue. Same goes for the other unresolved symbols. e.g. class "Locale" is defined in "liveMedia\Locale.cpp"... cu Roland From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Melvin_Raj Sent: Tuesday, August 05, 2008 3:04 AM To: live-devel at lists.live555.com Subject: [Live-devel] Help with link error hello....im new to this forum...im trying to build live555's test prog, specificly the openrtsp...i am able to build the mpegsender,mp3sender and receiver respectively witout any errors but when i try to build openrtsp, i get the following errors: 1>------ Rebuild All started: Project: rtsp, Configuration: Debug Win32 ------ 1>Deleting intermediate and output files for project 'rtsp', configuration 'Debug|Win32' 1>Compiling... 1>openRTSP.cpp 1>Compiling manifest to resources... 1>Linking... 1>openRTSP.obj : error LNK2001: unresolved external symbol "unsigned short tunnelOverHTTPPortNum" (?tunnelOverHTTPPortNum@@3GA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "unsigned int statusCode" (?statusCode@@3IA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double duration" (?duration@@3NA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double scale" (?scale@@3NA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double initialSeekTime" (?initialSeekTime@@3NA) 1>liveMedia.lib(RTSPClient.obj) : error LNK2019: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) referenced in function "char * __cdecl createScaleString(float,float)" (?createScaleString@@YAPADMM at Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2001: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) 1>liveMedia.lib(RTSPCommon.obj) : error LNK2001: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) 1>liveMedia.lib(RTSPClient.obj) : error LNK2019: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) referenced in function "char * __cdecl createScaleString(float,float)" (?createScaleString@@YAPADMM at Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2001: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) 1>liveMedia.lib(RTSPCommon.obj) : error LNK2001: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2019: unresolved external symbol "public: static class H263plusVideoRTPSource * __cdecl H263plusVideoRTPSource::createNew(class UsageEnvironment &,class Groupsock *,unsigned char,unsigned int)" (?createNew at H263plusVideoRTPSource@@SAPAV1 at AAVUsag eEnvironment@@PAVGroupsock@@EI at Z) referenced in function "public: unsigned int __thiscall MediaSubsession::initiate(int)" (?initiate at MediaSubsession@@QAEIH at Z) 1>MSVCRTD.lib(crtexe.obj) : error LNK2019: unresolved external symbol _main referenced in function ___tmainCRTStartup 1>C:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\Debug\rtsp.exe : fatal error LNK1120: 9 unresolved externals 1>Build log was saved at "file://c:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\rtsp\Debug\BuildLog.htm" 1>rtsp - 14 error(s), 0 warning(s) ========== Rebuild All: 0 succeeded, 1 failed, 0 skipped i've already added the required lib: wsock32.lib BasicUsageEnvironment.lib groupsock.lib liveMedia.lib UsageEnvironment.lib any help in pointing me towards the proper direction will be appreciated http://www.gidforums.com/images/gid/smilies/icon_smile.gifthank you in advance....God bless... ps: im trying to build an rtp receiver that will be able to play streaming files from a server... _____ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. _____ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 174 bytes Desc: not available URL: From nshamshiva at gmail.com Tue Aug 5 11:52:57 2008 From: nshamshiva at gmail.com (Shiva Shankar N) Date: Tue, 5 Aug 2008 15:52:57 -0300 Subject: [Live-devel] Problem in executing openRTSP Message-ID: <4f96b010808051152y1af616ecj78a515a8879126bb@mail.gmail.com> Hi all, I am trying to grab video streaming from axis camera using openRTSP. I have download the source code from live555 and built make file and executed the all the make files and built project files in Visual c++ 6.0 (for usageenv, basicusageenv, groupsock) and then finally i built the testprg to get openRTSP.exe. With this i am trying to grab the mpeg4 video but it gives me error saying " * Failed to get a SDP description from URL "rtsp://128.xxx.xxx.xxx/mpeg.mp4": cannot handle DESCRIBE response: RTSP/1.0 404 Not Found* " please can someone tell me where exactly i am going wrong. i am getting error when i type the following command in command prompt : *d:\Media\live\live\testProgs>openRTSP -t rtsp://**128.xxx.xxx.1xx* * Sending request: OPTIONS rtsp://**128.xxx.xxx.1xx** RTSP/1.0 CSeq: 1 User-Agent: openRTSP (LIVE555 Streaming Media v2008.07.24) Received OPTIONS response: RTSP/1.0 200 OK CSeq: 1 Public: DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, TEARDOWN Sending request: DESCRIBE rtsp://**128.xxx.xxx.1xx** RTSP/1.0 CSeq: 2 Accept: application/sdp User-Agent: openRTSP (LIVE555 Streaming Media v2008.07.24) Received DESCRIBE response: RTSP/1.0 404 Not Found CSeq: 2 Failed to get a SDP description from URL "rtsp://128.xxx.xxx.1xx": cannot handle DESCRIBE response: RTSP/1.0 404 Not Found *I tried without -t option but still it gives me same problem, is there anything that i am missing in this ? Thank sham -------------- next part -------------- An HTML attachment was scrubbed... URL: From Melvin_Raj at satyam.com Tue Aug 5 19:22:05 2008 From: Melvin_Raj at satyam.com (Melvin_Raj) Date: Wed, 6 Aug 2008 10:22:05 +0800 Subject: [Live-devel] Help with link error In-Reply-To: <000a01c8f728$789f7dd0$69de7970$@com> References: <000a01c8f728$789f7dd0$69de7970$@com> Message-ID: Hello..thank you for your help...I managed to reduce the errors down to 3!! :) 1>------ Build started: Project: rtsp, Configuration: Debug Win32 ------ 1>Compiling... 1>Locale.cpp 1>Linking... 1>playCommon.obj : error LNK2019: unresolved external symbol "public: static class H264VideoFileSink * __cdecl H264VideoFileSink::createNew(class UsageEnvironment &,char const *,unsigned int,unsigned int)" (?createNew at H264VideoFileSink@@SAPAV1 at AAVUsageEnvironment@@PBDII at Z) referenced in function _main 1>liveMedia.lib(MediaSession.obj) : error LNK2019: unresolved external symbol "public: static class H263plusVideoRTPSource * __cdecl H263plusVideoRTPSource::createNew(class UsageEnvironment &,class Groupsock *,unsigned char,unsigned int)" (?createNew at H263plusVideoRTPSource@@SAPAV1 at AAVUsageEnvironment@@PAVGroupsock@@EI at Z) referenced in function "public: unsigned int __thiscall MediaSubsession::initiate(int)" (?initiate at MediaSubsession@@QAEIH at Z) 1>C:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\Debug\rtsp.exe : fatal error LNK1120: 2 unresolved externals 1>Build log was saved at "file://c:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\rtsp\Debug\BuildLog.htm" 1>rtsp - 3 error(s), 0 warning(s) ========== Build: 0 succeeded, 1 failed, 0 up-to-date, 0 skipped ========== As you said, I tried doin a search for the H264VideoFileSink but it all in the livemedia.hh and also H264VideoFileSink.hh(I have included then in the preprocessor)....how do I go about corercting this error??and btw, once I compile openrtsp, will I be able to receive,play,records the video streams from mpegstreamer?? Thank You soo much for your time in helping :) God blesss... From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Roland Sent: Wednesday, August 06, 2008 2:24 AM To: 'LIVE555 Streaming Media - development & use' Subject: Re: [Live-devel] Help with link error Hi Melvin Do a FindInFiles for the symbols that aren't being resolved, e.g. "tunnelOverHTTPPortNum" or "statusCode". Then figure out if that source file where these variables are being defined is included in the compilation of the .LIB or the final .EXE. Add the corresponding source file to one of those projects and recompile. E.g. for the "tunnelOverHTTPPortNum" variable, you'll find "extern" references and one real declaration in testProgs\playCommon.cpp. Add playCommon.cpp to resolve this issue. Same goes for the other unresolved symbols. e.g. class "Locale" is defined in "liveMedia\Locale.cpp"... cu Roland From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Melvin_Raj Sent: Tuesday, August 05, 2008 3:04 AM To: live-devel at lists.live555.com Subject: [Live-devel] Help with link error hello....im new to this forum...im trying to build live555's test prog, specificly the openrtsp...i am able to build the mpegsender,mp3sender and receiver respectively witout any errors but when i try to build openrtsp, i get the following errors: 1>------ Rebuild All started: Project: rtsp, Configuration: Debug Win32 ------ 1>Deleting intermediate and output files for project 'rtsp', configuration 'Debug|Win32' 1>Compiling... 1>openRTSP.cpp 1>Compiling manifest to resources... 1>Linking... 1>openRTSP.obj : error LNK2001: unresolved external symbol "unsigned short tunnelOverHTTPPortNum" (?tunnelOverHTTPPortNum@@3GA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "unsigned int statusCode" (?statusCode@@3IA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double duration" (?duration@@3NA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double scale" (?scale@@3NA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double initialSeekTime" (?initialSeekTime@@3NA) 1>liveMedia.lib(RTSPClient.obj) : error LNK2019: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) referenced in function "char * __cdecl createScaleString(float,float)" (?createScaleString@@YAPADMM at Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2001: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) 1>liveMedia.lib(RTSPCommon.obj) : error LNK2001: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) 1>liveMedia.lib(RTSPClient.obj) : error LNK2019: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) referenced in function "char * __cdecl createScaleString(float,float)" (?createScaleString@@YAPADMM at Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2001: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) 1>liveMedia.lib(RTSPCommon.obj) : error LNK2001: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2019: unresolved external symbol "public: static class H263plusVideoRTPSource * __cdecl H263plusVideoRTPSource::createNew(class UsageEnvironment &,class Groupsock *,unsigned char,unsigned int)" (?createNew at H263plusVideoRTPSource@@SAPAV1 at AAVUsag eEnvironment@@PAVGroupsock@@EI at Z) referenced in function "public: unsigned int __thiscall MediaSubsession::initiate(int)" (?initiate at MediaSubsession@@QAEIH at Z) 1>MSVCRTD.lib(crtexe.obj) : error LNK2019: unresolved external symbol _main referenced in function ___tmainCRTStartup 1>C:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\Debug\rtsp.exe : fatal error LNK1120: 9 unresolved externals 1>Build log was saved at "file://c:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\rtsp\Debug\BuildLog.htm" 1>rtsp - 14 error(s), 0 warning(s) ========== Rebuild All: 0 succeeded, 1 failed, 0 skipped i've already added the required lib: wsock32.lib BasicUsageEnvironment.lib groupsock.lib liveMedia.lib UsageEnvironment.lib any help in pointing me towards the proper direction will be appreciated [cid:image001.gif at 01C8F7AE.0A35D750] thank you in advance....God bless... ps: im trying to build an rtp receiver that will be able to play streaming files from a server... ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 174 bytes Desc: image001.gif URL: From boris.kazakov at gmail.com Wed Aug 6 02:40:32 2008 From: boris.kazakov at gmail.com (=?KOI8-R?B?4s/SydMg68HawcvP1w==?=) Date: Wed, 6 Aug 2008 13:40:32 +0400 Subject: [Live-devel] Live streaming with DeviceSource Message-ID: <681c4a70808060240t5c9931d7gf8c4ccbe756d3755@mail.gmail.com> Hello, I'm trying to stream MPEG Video Elementary Stream encoded with ffmpeg using live. I've written a subclass of FramedSource based on the directions given in DeviceSource. I'm showing video sequence and streaming it at the same time. I encode data into sequence of buffers and then pass this buffers into my subclass. Everything works relatively fine. I'm only confused with the fact that when I'm passing buffers with the frame rate greater or equal than the frame rate of the encoded mpeg, everything works fine, but when the rate of passing buffers is lower that the frame rate of encoded mpeg, VLC plays several frames and than freezes. Why might this happen? (In my subclass I use gettimeofday(&fPresentationTime, NULL) to set presentation time). Best regards, Boris Kazakov. -------------- next part -------------- An HTML attachment was scrubbed... URL: From kumar.bala at DSP-Weuffen.de Wed Aug 6 01:17:59 2008 From: kumar.bala at DSP-Weuffen.de (Kumar Bala) Date: Wed, 06 Aug 2008 10:17:59 +0200 Subject: [Live-devel] Windows vs Linux RTSP/RTP/UDP difference In-Reply-To: <9DEC7033FCB743A2B7A0C228E45B3AC0@DSPWeuffen.local> References: <0883400D15254E839A5F790A703D4C07@DSPWeuffen.local> <9DEC7033FCB743A2B7A0C228E45B3AC0@DSPWeuffen.local> Message-ID: <48995E37.7000102@dsp-weuffen.de> Hi, I have finally able to achieve the same latency on both the linux and windows machine. I had to play with some cacheing parameters in VLC to acheive low latency for MPEG 4 on windows. If anyone is interested I can share the commandline paramters. Cheers Kumar Kumar Bala wrote: > Any one has any ideas on why this couldd be ? Has any one seen this > issue in openSIP related application ? > Kumar Bala wrote: >> Hello All, >> I am facing a bizarre platform dependent problem in streaming real >> time Mpeg 4 (Standard and High Definition). >> >> I have an embedded linux where I capture a live video and convert it >> into MPEG4 frames (using a hardware encoder). I use >> MPEG4DiscreteStreamFramer to stream these frames using modified >> testOnDemandRTSP example. >> I have almost negligible latency when my client (mplayer) runs on a >> linux platform. But the same sources (at least rtp relevant part) when >> compiled on windows using mingw shows a huge latency of about 2 to 2.5 >> sec. >> >> Has anyone else faced these problems before ? Are they related to >> lower level protocol implementation ? Any suggestion would of great help. >> _______________________________________________ >> live-devel mailing list >> live-devel at lists.live555.com >> http://lists.live555.com/mailman/listinfo/live-devel >> > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > -- Mit freundlichen Gr??en / Best regards Kumar Bala (Software Developer and System Designer) DSP-Weuffen Software/Hardware f?r DSP- und MCP-Konzeption und Optimierung ihres Systems DSP-Weuffen GmbH Schomburger Strasse 11 88279 Amtzell Tel: +49 7520 / 96673 - 146 Fax: +49 7520 / 96673 - 124 Mail: kumar.bala at dsp-weuffen.de Web: http://www.dsp-weuffen.de Gesch?ftsf?hrer: Dieter Weuffen | Amtsgericht 89014 ULM | HR-NR: HRB621135 Bitte beachten Sie, dass diese E-Mail einschlie?lich aller eventuell angeh?ngten Dokumente vertrauliche und/oder rechtlich gesch?tzte Informationen enthalten kann. Der Inhalt ist ausschlie?lich f?r den bezeichneten Adressaten bestimmt. Wenn Sie nicht der richtige Adressat oder dessen Vertreter sind, setzen Sie sich bitte mit dem Absender der E-Mail in Verbindung und l?schen Sie die E-Mail sofort. Jede Form der Verwendung, Ver?ffentlichung, Vervielf?ltigung oder Weitergabe des Inhalts fehlgeleiteter E-Mails ist unzul?ssig. From roland at wingmanteam.com Wed Aug 6 11:35:31 2008 From: roland at wingmanteam.com (Roland) Date: Wed, 6 Aug 2008 11:35:31 -0700 Subject: [Live-devel] Help with link error In-Reply-To: References: <000a01c8f728$789f7dd0$69de7970$@com> Message-ID: <001401c8f7f3$3543d290$9fcb77b0$@com> Hi Melvin Add liveMedia\H263plusVideoRTPSource.cpp and liveMedia\H264VideoFileSink.cpp to your project. That should do it. Similar to like you added "Locale.cpp". During linking, the Linker will gather all the object files (generated by compiling C or C++ source code) and all the libraries and do a cross-check against all symbols. E.g. if you call a function Foo() in your code, the compiler only sees the signature of the function in some header file: "foo.h: void Foo(void);" There is no code there, just the information on how to call it. During the linking step, the linker will resolve the references and look for compiled code for Foo(), which is either in another object file, or in a library (which is just a 'zip' of a bunch of object files). cu roland From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Melvin_Raj Sent: Tuesday, August 05, 2008 7:22 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Help with link error Hello..thank you for your help.I managed to reduce the errors down to 3!! J 1>------ Build started: Project: rtsp, Configuration: Debug Win32 ------ 1>Compiling... 1>Locale.cpp 1>Linking... 1>playCommon.obj : error LNK2019: unresolved external symbol "public: static class H264VideoFileSink * __cdecl H264VideoFileSink::createNew(class UsageEnvironment &,char const *,unsigned int,unsigned int)" (?createNew at H264VideoFileSink@@SAPAV1 at AAVUsageEnvironment@@PBDII at Z) referenced in function _main 1>liveMedia.lib(MediaSession.obj) : error LNK2019: unresolved external symbol "public: static class H263plusVideoRTPSource * __cdecl H263plusVideoRTPSource::createNew(class UsageEnvironment &,class Groupsock *,unsigned char,unsigned int)" (?createNew at H263plusVideoRTPSource@@SAPAV1 at AAVUsageEnvironment@@PAVGroupsock @@EI at Z) referenced in function "public: unsigned int __thiscall MediaSubsession::initiate(int)" (?initiate at MediaSubsession@@QAEIH at Z) 1>C:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\Debug\rtsp.exe : fatal error LNK1120: 2 unresolved externals 1>Build log was saved at "file://c:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\rtsp\Debug\BuildLog.htm" 1>rtsp - 3 error(s), 0 warning(s) ========== Build: 0 succeeded, 1 failed, 0 up-to-date, 0 skipped ========== As you said, I tried doin a search for the H264VideoFileSink but it all in the livemedia.hh and also H264VideoFileSink.hh(I have included then in the preprocessor)..how do I go about corercting this error??and btw, once I compile openrtsp, will I be able to receive,play,records the video streams from mpegstreamer?? Thank You soo much for your time in helping J God blesss. From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Roland Sent: Wednesday, August 06, 2008 2:24 AM To: 'LIVE555 Streaming Media - development & use' Subject: Re: [Live-devel] Help with link error Hi Melvin Do a FindInFiles for the symbols that aren't being resolved, e.g. "tunnelOverHTTPPortNum" or "statusCode". Then figure out if that source file where these variables are being defined is included in the compilation of the .LIB or the final .EXE. Add the corresponding source file to one of those projects and recompile. E.g. for the "tunnelOverHTTPPortNum" variable, you'll find "extern" references and one real declaration in testProgs\playCommon.cpp. Add playCommon.cpp to resolve this issue. Same goes for the other unresolved symbols. e.g. class "Locale" is defined in "liveMedia\Locale.cpp"... cu Roland From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Melvin_Raj Sent: Tuesday, August 05, 2008 3:04 AM To: live-devel at lists.live555.com Subject: [Live-devel] Help with link error hello....im new to this forum...im trying to build live555's test prog, specificly the openrtsp...i am able to build the mpegsender,mp3sender and receiver respectively witout any errors but when i try to build openrtsp, i get the following errors: 1>------ Rebuild All started: Project: rtsp, Configuration: Debug Win32 ------ 1>Deleting intermediate and output files for project 'rtsp', configuration 'Debug|Win32' 1>Compiling... 1>openRTSP.cpp 1>Compiling manifest to resources... 1>Linking... 1>openRTSP.obj : error LNK2001: unresolved external symbol "unsigned short tunnelOverHTTPPortNum" (?tunnelOverHTTPPortNum@@3GA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "unsigned int statusCode" (?statusCode@@3IA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double duration" (?duration@@3NA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double scale" (?scale@@3NA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double initialSeekTime" (?initialSeekTime@@3NA) 1>liveMedia.lib(RTSPClient.obj) : error LNK2019: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) referenced in function "char * __cdecl createScaleString(float,float)" (?createScaleString@@YAPADMM at Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2001: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) 1>liveMedia.lib(RTSPCommon.obj) : error LNK2001: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) 1>liveMedia.lib(RTSPClient.obj) : error LNK2019: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) referenced in function "char * __cdecl createScaleString(float,float)" (?createScaleString@@YAPADMM at Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2001: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) 1>liveMedia.lib(RTSPCommon.obj) : error LNK2001: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2019: unresolved external symbol "public: static class H263plusVideoRTPSource * __cdecl H263plusVideoRTPSource::createNew(class UsageEnvironment &,class Groupsock *,unsigned char,unsigned int)" (?createNew at H263plusVideoRTPSource@@SAPAV1 at AAVUsag eEnvironment@@PAVGroupsock@@EI at Z) referenced in function "public: unsigned int __thiscall MediaSubsession::initiate(int)" (?initiate at MediaSubsession@@QAEIH at Z) 1>MSVCRTD.lib(crtexe.obj) : error LNK2019: unresolved external symbol _main referenced in function ___tmainCRTStartup 1>C:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\Debug\rtsp.exe : fatal error LNK1120: 9 unresolved externals 1>Build log was saved at "file://c:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\rtsp\Debug\BuildLog.htm" 1>rtsp - 14 error(s), 0 warning(s) ========== Rebuild All: 0 succeeded, 1 failed, 0 skipped i've already added the required lib: wsock32.lib BasicUsageEnvironment.lib groupsock.lib liveMedia.lib UsageEnvironment.lib any help in pointing me towards the proper direction will be appreciated http://www.gidforums.com/images/gid/smilies/icon_smile.gifthank you in advance....God bless... ps: im trying to build an rtp receiver that will be able to play streaming files from a server... _____ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. _____ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. _____ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 174 bytes Desc: not available URL: From mamille1 at rockwellcollins.com Wed Aug 6 16:24:38 2008 From: mamille1 at rockwellcollins.com (mamille1 at rockwellcollins.com) Date: Wed, 6 Aug 2008 18:24:38 -0500 Subject: [Live-devel] Very large P-frames in a recording In-Reply-To: Message-ID: Roland Thanks for the suggestion. We've confirmed that the large frames contain only a single picture by searching for the MPEG-2 start codes. We're working on a filter that will truncate a frame when it exceeds a given size, so that excessively large frames are not written to the file. What exactly do you look for in Wireshark? I can see that it recognizes the RTP streams, but I didn't see where to get a more detailed analysis -- it's hard for me to tell what the analysis graphs are really measuring. -=- Mike Miller Rockwell Collins, Inc. Cedar Rapids, IA -------------- next part -------------- An HTML attachment was scrubbed... URL: From Melvin_Raj at satyam.com Wed Aug 6 20:02:32 2008 From: Melvin_Raj at satyam.com (Melvin_Raj) Date: Thu, 7 Aug 2008 11:02:32 +0800 Subject: [Live-devel] Help with link error In-Reply-To: <001401c8f7f3$3543d290$9fcb77b0$@com> References: <000a01c8f728$789f7dd0$69de7970$@com> <001401c8f7f3$3543d290$9fcb77b0$@com> Message-ID: Hyee Roland, Thank you for your help..managed to solve the errors and learn new thing at the same time...now im getting another error when running the openrtsp program from command line: [cid:image002.png at 01C8F87D.1737EA40] What do I have to do??thank you for your help...do have a working example that I can see(the print screens perhaps)??i want to see how it works..thank you :) ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 6537 bytes Desc: image002.png URL: From nshamshiva at gmail.com Wed Aug 6 22:04:18 2008 From: nshamshiva at gmail.com (Shiva Shankar N) Date: Thu, 7 Aug 2008 02:04:18 -0300 Subject: [Live-devel] Help with openrtsp Message-ID: <4f96b010808062204u4303381aia0f19fa1f78b15bb@mail.gmail.com> Hi all, I have an mpeg streamer device (axis camera) which delivers a stream via RTSP..I have downloaded source code from live555 website and built an openRTSP.exe using visual c++. When i try to run this openRTSP with an ipaddress it gives an error " Failed to get a SDP description from URL "rtsp://128.197.178.104/mpeg.mp4": cannot handle DESCRIBE response: RTSP/1.0 404 Not Found " Please can some one tell me where exactly i am going wrong. Command Prompt stderr : d:\Media\live\live\testProgs>openRTSP -t rtsp://128.xxx.xxx.1xx Sending request: OPTIONS rtsp://128.xxx.xxx.1xx RTSP/1.0 CSeq: 1 User-Agent: openRTSP (LIVE555 Streaming Media v2008.07.24) Received OPTIONS response: RTSP/1.0 200 OK CSeq: 1 Public: DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, TEARDOWN Sending request: DESCRIBE rtsp://128.xxx.xxx.1xx RTSP/1.0 CSeq: 2 Accept: application/sdp User-Agent: openRTSP (LIVE555 Streaming Media v2008.07.24) Received DESCRIBE response: RTSP/1.0 404 Not Found CSeq: 2 Failed to get a SDP description from URL "rtsp://128.xxx.xxx.1xx": cannot handle DESCRIBE response: RTSP/1.0 404 Not Found I tried without -t option but still it gives me same problem, is there anything that i am missing in this ? Thank sham -------------- next part -------------- An HTML attachment was scrubbed... URL: From robin.penea at gmail.com Wed Aug 6 22:58:21 2008 From: robin.penea at gmail.com (Robin Penea) Date: Thu, 7 Aug 2008 07:58:21 +0200 Subject: [Live-devel] Help with openrtsp In-Reply-To: <4f96b010808062204u4303381aia0f19fa1f78b15bb@mail.gmail.com> References: <4f96b010808062204u4303381aia0f19fa1f78b15bb@mail.gmail.com> Message-ID: Hi there, I think you might have an error in your calling URL to get the RTSP stream. If you take a look at the axis api http://www.axis.com/techsup/cam_servers/dev/cam_rtsp_api.php , they say that the URL looks like rtsp:///mpeg4/media.amp. So your URL should be rtsp://128.197.178.104/mpeg4/media.amp. Hope this helps ! -- Robin Penea -------------- next part -------------- An HTML attachment was scrubbed... URL: From Bernhard at Feiten.de Wed Aug 6 22:59:10 2008 From: Bernhard at Feiten.de (Bernhard Feiten) Date: Thu, 7 Aug 2008 07:59:10 +0200 Subject: [Live-devel] Media stream duplication on a streaming relay References: <1134049882.43983a5aa6b32@imp4-g19.free.fr><6.2.3.4.1.20051208104044.032296d0@localhost> <00e901c8d74d$6de53290$d302a8c0@Popelfranz> Message-ID: <00b201c8f852$b7388340$d302a8c0@Popelfranz> Dear Ross, all, as nobody answered I started to write a StreamDup class following your hints from the e-mail below. Related to that I have two questions? Would it be required that the sequnce number and timestamp of the duplicates are newly initialized? Would RTCP also need to be considered then? Thank you in advance, Bernhard ----- Original Message ----- From: "Bernhard Feiten" To: "LIVE555 Streaming Media - development & use" Sent: Thursday, June 26, 2008 7:28 AM Subject: Re: [Live-devel] Media stream duplication on a streaming relay > Dear Ross, all, > > did somebody wrote the StreamDuplication class you proposed below already? > > Do you have a hint how the getNextFrame functions could be synchronized? > > Thank you very much, > Bernhard > > > > > ----- Original Message ----- > From: "Ross Finlayson" > To: "LIVE555 Streaming Media - development & use" > > Sent: Thursday, December 08, 2005 8:57 PM > Subject: Re: [Live-devel] Media stream duplication on a streaming relay > > >> As you noticed, you can't duplicate a stream by having each recipient >> read from a single object, because "getNextFrame()" can't be called more >> than once on the same object concurrently. >> >> The solution, instead, is to create a separate object (of some >> "FramedSource" subclass) for each recipient. This new class would >> implement the "doGetNextFrame()" virtual function by somehow >> 'registering' with the data source object - to request a copy of the next >> incoming frame. >> >> The data source object (which would *not* be a subclass of >> "FramedSource", and so would not implement "doGetNextFrame()") would >> handle these requests by delivering copies of each incoming frame to each >> recipient, and not ask for a new incoming frame (from its upstream data >> source) until it has finished delivering data to each downstream >> recipient. >> >> For an example of code that is similar to this, note the relationship >> between the "MPEG1or2Demux" and "MPEG1or2DemuxedElementaryStream" >> classes. (In this case, however, we are demultiplexing data to downstream >> recipients, rather than duplicating it.) Note in particular that (i) >> "MPEG1or2Demux" is subclassed from "Medium", not "FramedSource", and (ii) >> "MPEG1or2Demux" implements a "getNextFrame()" function that is similar >> to, but different from "FramedSource::getNextFrame()". >> >> (This functionality (data duplication) is something that should probably >> be added to the library someday. Until then, however, you will need to >> implement it yourself.) >> >> >> Ross Finlayson >> Live Networks, Inc. (LIVE555.COM) >> >> >> _______________________________________________ >> live-devel mailing list >> live-devel at lists.live555.com >> http://lists.live555.com/mailman/listinfo/live-devel > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel From Melvin_Raj at satyam.com Wed Aug 6 19:52:57 2008 From: Melvin_Raj at satyam.com (Melvin_Raj) Date: Thu, 7 Aug 2008 10:52:57 +0800 Subject: [Live-devel] Help with link error In-Reply-To: <001401c8f7f3$3543d290$9fcb77b0$@com> References: <000a01c8f728$789f7dd0$69de7970$@com> <001401c8f7f3$3543d290$9fcb77b0$@com> Message-ID: Hyee Roland, Thank you for your help..managed to solve the errors and learn new thing at the same time...now im getting another error when running the openrtsp program from command line: [cid:image002.png at 01C8F87B.C03BF5C0] What do I have to do??thank you for your help...do have a working example that I can see(the print screens perhaps)??i want to see how it works..thank you :) From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Roland Sent: Thursday, August 07, 2008 2:36 AM To: 'LIVE555 Streaming Media - development & use' Subject: Re: [Live-devel] Help with link error Hi Melvin Add liveMedia\H263plusVideoRTPSource.cpp and liveMedia\H264VideoFileSink.cpp to your project. That should do it. Similar to like you added "Locale.cpp". During linking, the Linker will gather all the object files (generated by compiling C or C++ source code) and all the libraries and do a cross-check against all symbols. E.g. if you call a function Foo() in your code, the compiler only sees the signature of the function in some header file: "foo.h: void Foo(void);" There is no code there, just the information on how to call it. During the linking step, the linker will resolve the references and look for compiled code for Foo(), which is either in another object file, or in a library (which is just a 'zip' of a bunch of object files). cu roland From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Melvin_Raj Sent: Tuesday, August 05, 2008 7:22 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Help with link error Hello..thank you for your help...I managed to reduce the errors down to 3!! :) 1>------ Build started: Project: rtsp, Configuration: Debug Win32 ------ 1>Compiling... 1>Locale.cpp 1>Linking... 1>playCommon.obj : error LNK2019: unresolved external symbol "public: static class H264VideoFileSink * __cdecl H264VideoFileSink::createNew(class UsageEnvironment &,char const *,unsigned int,unsigned int)" (?createNew at H264VideoFileSink@@SAPAV1 at AAVUsageEnvironment@@PBDII at Z) referenced in function _main 1>liveMedia.lib(MediaSession.obj) : error LNK2019: unresolved external symbol "public: static class H263plusVideoRTPSource * __cdecl H263plusVideoRTPSource::createNew(class UsageEnvironment &,class Groupsock *,unsigned char,unsigned int)" (?createNew at H263plusVideoRTPSource@@SAPAV1 at AAVUsageEnvironment@@PAVGroupsock@@EI at Z) referenced in function "public: unsigned int __thiscall MediaSubsession::initiate(int)" (?initiate at MediaSubsession@@QAEIH at Z) 1>C:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\Debug\rtsp.exe : fatal error LNK1120: 2 unresolved externals 1>Build log was saved at "file://c:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\rtsp\Debug\BuildLog.htm" 1>rtsp - 3 error(s), 0 warning(s) ========== Build: 0 succeeded, 1 failed, 0 up-to-date, 0 skipped ========== As you said, I tried doin a search for the H264VideoFileSink but it all in the livemedia.hh and also H264VideoFileSink.hh(I have included then in the preprocessor)....how do I go about corercting this error??and btw, once I compile openrtsp, will I be able to receive,play,records the video streams from mpegstreamer?? Thank You soo much for your time in helping :) God blesss... From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Roland Sent: Wednesday, August 06, 2008 2:24 AM To: 'LIVE555 Streaming Media - development & use' Subject: Re: [Live-devel] Help with link error Hi Melvin Do a FindInFiles for the symbols that aren't being resolved, e.g. "tunnelOverHTTPPortNum" or "statusCode". Then figure out if that source file where these variables are being defined is included in the compilation of the .LIB or the final .EXE. Add the corresponding source file to one of those projects and recompile. E.g. for the "tunnelOverHTTPPortNum" variable, you'll find "extern" references and one real declaration in testProgs\playCommon.cpp. Add playCommon.cpp to resolve this issue. Same goes for the other unresolved symbols. e.g. class "Locale" is defined in "liveMedia\Locale.cpp"... cu Roland From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Melvin_Raj Sent: Tuesday, August 05, 2008 3:04 AM To: live-devel at lists.live555.com Subject: [Live-devel] Help with link error hello....im new to this forum...im trying to build live555's test prog, specificly the openrtsp...i am able to build the mpegsender,mp3sender and receiver respectively witout any errors but when i try to build openrtsp, i get the following errors: 1>------ Rebuild All started: Project: rtsp, Configuration: Debug Win32 ------ 1>Deleting intermediate and output files for project 'rtsp', configuration 'Debug|Win32' 1>Compiling... 1>openRTSP.cpp 1>Compiling manifest to resources... 1>Linking... 1>openRTSP.obj : error LNK2001: unresolved external symbol "unsigned short tunnelOverHTTPPortNum" (?tunnelOverHTTPPortNum@@3GA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "unsigned int statusCode" (?statusCode@@3IA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double duration" (?duration@@3NA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double scale" (?scale@@3NA) 1>openRTSP.obj : error LNK2001: unresolved external symbol "double initialSeekTime" (?initialSeekTime@@3NA) 1>liveMedia.lib(RTSPClient.obj) : error LNK2019: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) referenced in function "char * __cdecl createScaleString(float,float)" (?createScaleString@@YAPADMM at Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2001: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) 1>liveMedia.lib(RTSPCommon.obj) : error LNK2001: unresolved external symbol "public: virtual __thiscall Locale::~Locale(void)" (??1Locale@@UAE at XZ) 1>liveMedia.lib(RTSPClient.obj) : error LNK2019: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) referenced in function "char * __cdecl createScaleString(float,float)" (?createScaleString@@YAPADMM at Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2001: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) 1>liveMedia.lib(RTSPCommon.obj) : error LNK2001: unresolved external symbol "public: __thiscall Locale::Locale(char const *,int)" (??0Locale@@QAE at PBDH@Z) 1>liveMedia.lib(MediaSession.obj) : error LNK2019: unresolved external symbol "public: static class H263plusVideoRTPSource * __cdecl H263plusVideoRTPSource::createNew(class UsageEnvironment &,class Groupsock *,unsigned char,unsigned int)" (?createNew at H263plusVideoRTPSource@@SAPAV1 at AAVUsag eEnvironment@@PAVGroupsock@@EI at Z) referenced in function "public: unsigned int __thiscall MediaSubsession::initiate(int)" (?initiate at MediaSubsession@@QAEIH at Z) 1>MSVCRTD.lib(crtexe.obj) : error LNK2019: unresolved external symbol _main referenced in function ___tmainCRTStartup 1>C:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\Debug\rtsp.exe : fatal error LNK1120: 9 unresolved externals 1>Build log was saved at "file://c:\Documents and Settings\mp74294\My Documents\Visual Studio 2005\Projects\rtsp\rtsp\Debug\BuildLog.htm" 1>rtsp - 14 error(s), 0 warning(s) ========== Rebuild All: 0 succeeded, 1 failed, 0 skipped i've already added the required lib: wsock32.lib BasicUsageEnvironment.lib groupsock.lib liveMedia.lib UsageEnvironment.lib any help in pointing me towards the proper direction will be appreciated [cid:image003.gif at 01C8F87B.C03BF5C0] thank you in advance....God bless... ps: im trying to build an rtp receiver that will be able to play streaming files from a server... ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 6537 bytes Desc: image002.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.gif Type: image/gif Size: 174 bytes Desc: image003.gif URL: From finlayson at live555.com Thu Aug 7 02:05:58 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 7 Aug 2008 10:05:58 +0100 Subject: [Live-devel] Help with openrtsp In-Reply-To: <4f96b010808062204u4303381aia0f19fa1f78b15bb@mail.gmail.com> References: <4f96b010808062204u4303381aia0f19fa1f78b15bb@mail.gmail.com> Message-ID: >I have an mpeg streamer device (axis camera) which delivers a stream >via RTSP..I have downloaded source code from live555 website and >built an openRTSP.exe using visual c++. >When i try to run this openRTSP with an ipaddress it gives an error > >" Failed to get a SDP description from URL >"rtsp://128.197.178.104/mpeg.mp4": >cannot handle DESCRIBE response: RTSP/1.0 404 Not Found " > >Please can some one tell me where exactly i am going wrong. C'mon folks - is it really too hard to figure out what a "Not Found" error message from the server means?? -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From yamini.s at europlex.in Thu Aug 7 02:26:16 2008 From: yamini.s at europlex.in (Yamini S. [EPLX - DCC]) Date: Thu, 7 Aug 2008 14:56:16 +0530 Subject: [Live-devel] Help with openrtsp References: <4f96b010808062204u4303381aia0f19fa1f78b15bb@mail.gmail.com> Message-ID: <6872DF21E3D09046BDF172721037F4FB45EC49@SBTEXCHANGEDB.sbtdats.com> Hi, For axis camera the URL: rtsp://IP:554/mpeg4/cameraNo/media.amp Assign the IP Address in MediaSession.cpp ,initiate(int useSpecialRTPoffset) member function . Thanks & Regards, S.Yamini Programmer R&D Siemens Building Technologies Pvt. Ltd. ________________________________ From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson Sent: Thursday, August 07, 2008 2:36 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Help with openrtsp I have an mpeg streamer device (axis camera) which delivers a stream via RTSP..I have downloaded source code from live555 website and built an openRTSP.exe using visual c++. When i try to run this openRTSP with an ipaddress it gives an error " Failed to get a SDP description from URL "rtsp://128.197.178.104/mpeg.mp4": cannot handle DESCRIBE response: RTSP/1.0 404 Not Found " Please can some one tell me where exactly i am going wrong. C'mon folks - is it really too hard to figure out what a "Not Found" error message from the server means?? -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ This message (including attachments) is confidential and may be privileged. If you have received it by mistake please notify the sender by return e-mail and delete this message from your system. You may not copy, disclose or use the contents in any way. SBTPL does not guarantee integrity of this communication nor is that this communication free of issues, interceptions or interference. This communication does not create or modify any contract, and unless otherwise stated, is not intended to be contractually binding. Views or opinions expressed in this e-mail message are those of the author only. -------------- next part -------------- An HTML attachment was scrubbed... URL: From eyedasan at gmail.com Thu Aug 7 02:53:09 2008 From: eyedasan at gmail.com (Kannadasan S) Date: Thu, 7 Aug 2008 15:23:09 +0530 Subject: [Live-devel] the Test MPEG2 Tranpost Streamer Does not work in ARM Message-ID: <1dcaa6230808070253r7e9f0689i6f344626b7dc7339@mail.gmail.com> Hi, I have cross compiled the live555 to ARM Platform. I can able to test the testOnDemandserver, that stream can able to play in VLC using the rtsp:// But, the VLC is not receiving the stream from testMPEG2TransportStreamer. Can any one help on this. Regards, Kannadasan S, -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Thu Aug 7 03:25:08 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 7 Aug 2008 11:25:08 +0100 Subject: [Live-devel] Live555 injector->setDestination() In-Reply-To: <.202.5.140.130.1217307529.squirrel@mail.verticity.com> References: <.202.5.140.130.1217307529.squirrel@mail.verticity.com> Message-ID: >But I am getting this error. I don't understand what is the problem, >if somebody has encountered similar type of problem then reply us >and let us know how you have resolved the issue. > > > > > >injector->setDestination() failed: cannot handle ANNOUNCE response: >RTSP/1.0 412 > > Precondition Failed This error is coming from the Darwin server - i.e., it's not a problem with the LIVE555 software. The error might mean that you don't have write access to the Darwin server - I'm not sure.... (Note also that we have our own RTSP server implementation, so you really don't need to be using a separate Darwin Streaming Server.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From Chris.Burns at abdeus.com Thu Aug 7 03:31:44 2008 From: Chris.Burns at abdeus.com (Chris Burns) Date: Thu, 7 Aug 2008 22:31:44 +1200 Subject: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header Message-ID: Hi all, I'm trying to figure out how to calculate the RTPTime values in the RTSP/PLAY-Response "RTP-Info" header, based on an incoming live RTP/RTCP stream(s) from a live encoder. I've pulled apart an HUSM packet dump and the critical section looks something like this: 769: RTCP-SR/9301 from LENC NTP: MSW: 3403825895 (0xcae242e7); LSW: 133143986 (0x07ef9db2) ==> Nov 12, 2007 03:11:35.0310 UTC RTP: 670943030 803: RTCP-SR/9303 from LENC NTP: MSW: 3403825896 (0xcae242e8); LSW: 64424509 (0x03d70a3d) ==> Nov 12, 2007 03:11:36.0150 UTC RTP: 383790848 840: RTCP-SR/9301 from LENC NTP: MSW: 3403825897 (0xcae242e9); LSW: 64424509 (0x03d70a3d) ==> Nov 12, 2007 03:11:37.0150 UTC RTP: 671123120 871: RTSP/PLAY-Request from CLIENT 881: RTCP-SR/9303 from LENC NTP: MSW: 3403825898 (0xcae242ea); LSW: 64424509 (0x03d70a3d) ==> Nov 12, 2007 03:11:38.0150 UTC RTP: 383834926 882: RTCP-SR/3457 to CLIENT Same as 881 906: RTCP-RR/3457 from CLIENT 919: RTCP-SR/9301 from LENC NTP: MSW: 3403825899 (0xcae242eb); LSW: 133143986 (0x07ef9db2) ==> Nov 12, 2007 03:11:39.0310 UTC RTP: 671303570 920: RTCP-SR/9301 to CLIENT Same as 919 921: RTP/9302 from LENC Seq: 39932 RTP: 383841717 922: RTP/3456 to CLIENT Same as 921 923: RTP/9300 from LENC Seq: 59095 RTP: 671369087 924: RTP/9302 from LENC Seq: 39933 RTP: 383842741 925: RTSP/PLAY-Response to CLIENT RTP-Info: url=rtsp://172.28.15.239/rtpencoder/Blah.sdp/streamid=1;seq=39932;rtptim e=383841717, url=rtsp://172.28.15.239/rtpencoder/Blah.sdp/streamid=0;seq=59095;rtptim e=671239890 926: RTP/3458 to CLIENT Seq: 59095 RTP: 671369087 927: RTP/3456 to CLIENT Seq: 39933 RTP: 383842741 How in the world did HUSM come up with "rtptime=671239890"????? Is there any reference or commonly cited material about how these values are calculated? Thanks in advance, ChrisB Chris Burns M: +64 21 391 286 chris.burns at abdeus.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From eyedasan at gmail.com Thu Aug 7 05:57:51 2008 From: eyedasan at gmail.com (Kannadasan S) Date: Thu, 7 Aug 2008 18:27:51 +0530 Subject: [Live-devel] the Test MPEG2 Tranpost Streamer Does not work in ARM Message-ID: <1dcaa6230808070557y473563d5h73833519204bb759@mail.gmail.com> Hi all, I am running the testMPEG2TransportStreamer in the ARM Board. ./testMPEG2TransportStreamer The streaming is Happening. And i am using the Mplyer to receive the stream in my x86 Machine. I have run mplayer udp://1234 Its giving some errors, Playing udp://@:1234/. STREAM_UDP, URL: udp://@:1234/ Failed to connect to server udp_streaming_start failed No stream found to handle url udp://@:1234/ Can anyone help on this issue. The RTSP receving is happening in mplayer. Regards, Kannadasan -- Kannadasan S, Cell : 9742078830 -------------- next part -------------- An HTML attachment was scrubbed... URL: From nshamshiva at gmail.com Thu Aug 7 07:09:14 2008 From: nshamshiva at gmail.com (Shiva Shankar N) Date: Thu, 7 Aug 2008 11:09:14 -0300 Subject: [Live-devel] Help with openrtsp In-Reply-To: References: <4f96b010808062204u4303381aia0f19fa1f78b15bb@mail.gmail.com> Message-ID: <4f96b010808070709hc1baad3ma7e86fa93286eb5b@mail.gmail.com> Hi Ross, I knew that my server was not responding , but i wanted know the method in which the server (axis camera) is called. when i trace the code READSOCKET() function was not responding with this i was sure that there is some commincation problem between server and client. Thanks Sham 2008/8/7 Ross Finlayson > I have an mpeg streamer device (axis camera) which delivers a stream via > RTSP..I have downloaded source code from live555 website and built an > openRTSP.exe using visual c++. > When i try to run this openRTSP with an ipaddress it gives an error > > " Failed to get a SDP description from URL "rtsp:// > 128.197.178.104/mpeg.mp4": cannot handle DESCRIBE response: RTSP/1.0 404 > Not Found " > > Please can some one tell me where exactly i am going wrong. > > > C'mon folks - is it really too hard to figure out what a "Not Found" error > message from the server means?? > > -- > > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mdnpfa001 at uct.ac.za Thu Aug 7 08:03:46 2008 From: mdnpfa001 at uct.ac.za (PFARISO MADANDA) Date: Thu, 07 Aug 2008 17:03:46 +0200 Subject: [Live-devel] Live555 with C# Message-ID: <489B2AF20200002D0008EFE0@gwiasmtp.uct.ac.za> Hi, Is it possible to use Live555 with C# and if so, is there an example somewhere? Thanks, From roland at wingmanteam.com Thu Aug 7 11:24:46 2008 From: roland at wingmanteam.com (Roland) Date: Thu, 7 Aug 2008 11:24:46 -0700 Subject: [Live-devel] Very large P-frames in a recording In-Reply-To: References: Message-ID: <005001c8f8ba$deab4170$9c01c450$@com> Hello Mike In my particular case, I was looking for lost packets (RTSP over UDP) in order to figure out how I should deal with those in the codec and also what the RTCP packets were containing, since I couldn't figure out the weird data I was getting. The graphs don't tell me much either. Not only do they contain bars that span the entire Y-axis, they also suffer from severe redraw issues, once you zoom in on some spots... I tend to look mostly at the list of packets and then use "Jump To" to further inspect the packets in the main window. roland From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of mamille1 at rockwellcollins.com Sent: Wednesday, August 06, 2008 4:25 PM To: live-devel at ns.live555.com Subject: [Live-devel] Very large P-frames in a recording Roland Thanks for the suggestion. We've confirmed that the large frames contain only a single picture by searching for the MPEG-2 start codes. We're working on a filter that will truncate a frame when it exceeds a given size, so that excessively large frames are not written to the file. What exactly do you look for in Wireshark? I can see that it recognizes the RTP streams, but I didn't see where to get a more detailed analysis -- it's hard for me to tell what the analysis graphs are really measuring. -=- Mike Miller Rockwell Collins, Inc. Cedar Rapids, IA -------------- next part -------------- An HTML attachment was scrubbed... URL: From roland at wingmanteam.com Thu Aug 7 11:30:02 2008 From: roland at wingmanteam.com (Roland) Date: Thu, 7 Aug 2008 11:30:02 -0700 Subject: [Live-devel] Help with link error In-Reply-To: References: <000a01c8f728$789f7dd0$69de7970$@com> <001401c8f7f3$3543d290$9fcb77b0$@com> Message-ID: <005501c8f8bb$9b3bf5f0$d1b3e1d0$@com> Hi Melvin You could... add the command line parameter as the parameter for debugging in VS2005, then start RTSP.exe from within VisualStudio and step through the code (I think F10 is the default for step, and F11 for step into), set a lot of breakpoints, dig in and prepare to learn a lot from someone else's code. Another easy one is to search for "Failed to get a SDP" in the source code, isolate the print/output statement, and set a breakpoint there (F9). Then work your way backwards in the code and set breakpoints where you think everything should still be ok. Once you hit breakpoints, inspect variables and try to understand if the values stored in them make sense or not. I'll leave it to someone with RTSP experience to troubleshoot the other side of the equation (what's sitting at 239.255.42.42?)... Roland From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Melvin_Raj Sent: Wednesday, August 06, 2008 8:03 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Help with link error Hyee Roland, Thank you for your help..managed to solve the errors and learn new thing at the same time.now im getting another error when running the openrtsp program from command line: cid:image001.png at 01C8F87A.D056B0E0 What do I have to do??thank you for your help.do have a working example that I can see(the print screens perhaps)??i want to see how it works..thank you J _____ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 6537 bytes Desc: not available URL: From roland at wingmanteam.com Thu Aug 7 11:38:23 2008 From: roland at wingmanteam.com (Roland) Date: Thu, 7 Aug 2008 11:38:23 -0700 Subject: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header In-Reply-To: References: Message-ID: <005b01c8f8bc$c6c68c20$5453a460$@com> > Is there any reference or commonly cited material about how these values are calculated? "The RTP timestamp represents the sampling instant of the first octet of data in the frame. It starts from a random initial value and increments at a media-dependent rate." >From "RTP, Audio and Video for the Internet", Colin Perkins, 2003, ISBN: 0-672-32249-8 From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Chris Burns Sent: Thursday, August 07, 2008 3:32 AM To: live-devel at ns.live555.com Subject: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header Hi all, I'm trying to figure out how to calculate the RTPTime values in the RTSP/PLAY-Response "RTP-Info" header, based on an incoming live RTP/RTCP stream(s) from a live encoder. I've pulled apart an HUSM packet dump and the critical section looks something like this: 769: RTCP-SR/9301 from LENC NTP: MSW: 3403825895 (0xcae242e7); LSW: 133143986 (0x07ef9db2) ==> Nov 12, 2007 03:11:35.0310 UTC RTP: 670943030 803: RTCP-SR/9303 from LENC NTP: MSW: 3403825896 (0xcae242e8); LSW: 64424509 (0x03d70a3d) ==> Nov 12, 2007 03:11:36.0150 UTC RTP: 383790848 840: RTCP-SR/9301 from LENC NTP: MSW: 3403825897 (0xcae242e9); LSW: 64424509 (0x03d70a3d) ==> Nov 12, 2007 03:11:37.0150 UTC RTP: 671123120 871: RTSP/PLAY-Request from CLIENT 881: RTCP-SR/9303 from LENC NTP: MSW: 3403825898 (0xcae242ea); LSW: 64424509 (0x03d70a3d) ==> Nov 12, 2007 03:11:38.0150 UTC RTP: 383834926 882: RTCP-SR/3457 to CLIENT Same as 881 906: RTCP-RR/3457 from CLIENT 919: RTCP-SR/9301 from LENC NTP: MSW: 3403825899 (0xcae242eb); LSW: 133143986 (0x07ef9db2) ==> Nov 12, 2007 03:11:39.0310 UTC RTP: 671303570 920: RTCP-SR/9301 to CLIENT Same as 919 921: RTP/9302 from LENC Seq: 39932 RTP: 383841717 922: RTP/3456 to CLIENT Same as 921 923: RTP/9300 from LENC Seq: 59095 RTP: 671369087 924: RTP/9302 from LENC Seq: 39933 RTP: 383842741 925: RTSP/PLAY-Response to CLIENT RTP-Info: url=rtsp://172.28.15.239/rtpencoder/Blah.sdp/streamid=1;seq=39932;rtptime=38 3841717, url=rtsp://172.28.15.239/rtpencoder/Blah.sdp/streamid=0;seq=59095;rtptime=67 1239890 926: RTP/3458 to CLIENT Seq: 59095 RTP: 671369087 927: RTP/3456 to CLIENT Seq: 39933 RTP: 383842741 How in the world did HUSM come up with "rtptime=671239890"????? Is there any reference or commonly cited material about how these values are calculated? Thanks in advance, ChrisB Chris Burns M: +64 21 391 286 chris.burns at abdeus.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Chris.Burns at abdeus.com Thu Aug 7 15:55:26 2008 From: Chris.Burns at abdeus.com (Chris Burns) Date: Fri, 8 Aug 2008 10:55:26 +1200 Subject: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header In-Reply-To: References: Message-ID: Sorry, I should have been more clear in my question... I'm specifically trying to work out how to create the "RTP-Info" header in a RTSP/PLAY-Response. It looks quite simple. Grab the RTP-Time & SeqNum of the first RTP packet sent on each stream to the client, format it correctly and we're done. Unfortunately, this causes low-powered clients (like cell-phones) to have terrible A/V sync. If you look at the example I posted below, the first RTP packets going to the client for each stream (packets 922 & 926) have details like this: 922: RTP/3456 to CLIENT Seq: 39932 RTP: 383841717 926: RTP/3458 to CLIENT Seq: 59095 RTP: 671369087 But the RTSP/PLAY-Response looks like this: 925: RTSP/PLAY-Response to CLIENT RTP-Info: url=rtsp://172.28.15.239/rtpencoder/Blah.sdp/streamid=1;seq=39932;rtptim e=383841717, url=rtsp://172.28.15.239/rtpencoder/Blah.sdp/streamid=0;seq=59095;rtptim e=671239890 There is a delta of -129197 in the RTP-Time specified in the "RTP-Info" header and the RTP-Time in the packet with SeqNum==59095. How is this delta calculated? Anyone know? Cheers, ChrisB Chris Burns M: +64 21 391 286 chris.burns at abdeus.com Sent: Thursday, 07 August 2008 22:32 To: live-devel at ns.live555.com Subject: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info"header Hi all, I'm trying to figure out how to calculate the RTPTime values in the RTSP/PLAY-Response "RTP-Info" header, based on an incoming live RTP/RTCP stream(s) from a live encoder. I've pulled apart an HUSM packet dump and the critical section looks something like this: 769: RTCP-SR/9301 from LENC NTP: MSW: 3403825895 (0xcae242e7); LSW: 133143986 (0x07ef9db2) ==> Nov 12, 2007 03:11:35.0310 UTC RTP: 670943030 803: RTCP-SR/9303 from LENC NTP: MSW: 3403825896 (0xcae242e8); LSW: 64424509 (0x03d70a3d) ==> Nov 12, 2007 03:11:36.0150 UTC RTP: 383790848 840: RTCP-SR/9301 from LENC NTP: MSW: 3403825897 (0xcae242e9); LSW: 64424509 (0x03d70a3d) ==> Nov 12, 2007 03:11:37.0150 UTC RTP: 671123120 871: RTSP/PLAY-Request from CLIENT 881: RTCP-SR/9303 from LENC NTP: MSW: 3403825898 (0xcae242ea); LSW: 64424509 (0x03d70a3d) ==> Nov 12, 2007 03:11:38.0150 UTC RTP: 383834926 882: RTCP-SR/3457 to CLIENT Same as 881 906: RTCP-RR/3457 from CLIENT 919: RTCP-SR/9301 from LENC NTP: MSW: 3403825899 (0xcae242eb); LSW: 133143986 (0x07ef9db2) ==> Nov 12, 2007 03:11:39.0310 UTC RTP: 671303570 920: RTCP-SR/9301 to CLIENT Same as 919 921: RTP/9302 from LENC Seq: 39932 RTP: 383841717 922: RTP/3456 to CLIENT Same as 921 923: RTP/9300 from LENC Seq: 59095 RTP: 671369087 924: RTP/9302 from LENC Seq: 39933 RTP: 383842741 925: RTSP/PLAY-Response to CLIENT RTP-Info: url=rtsp://172.28.15.239/rtpencoder/Blah.sdp/streamid=1;seq=39932;rtptim e=383841717, url=rtsp://172.28.15.239/rtpencoder/Blah.sdp/streamid=0;seq=59095;rtptim e=671239890 926: RTP/3458 to CLIENT Seq: 59095 RTP: 671369087 927: RTP/3456 to CLIENT Seq: 39933 RTP: 383842741 How in the world did HUSM come up with "rtptime=671239890"????? Is there any reference or commonly cited material about how these values are calculated? Thanks in advance, ChrisB Chris Burns M: +64 21 391 286 chris.burns at abdeus.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Thu Aug 7 17:01:45 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Aug 2008 01:01:45 +0100 Subject: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header In-Reply-To: References: Message-ID: >I'm specifically trying to work out how to create the "RTP-Info" >header in a RTSP/PLAY-Response. Our server implementation does this automatically (see "RTSPServer.cpp", and search for "rtpInfo"). You shouldn't have to do anything yourself to generate this. Similarly, at the client end, the information is filled in automatically - if you wish, you can just access the "MediaSubsession::rtpInfo" structure. Most clients, however, will not need to access this structure directly. Instead, if they wish to get the current 'normal play time' (NPT) for the stream, they can just call "MediaSubsession::getNormalPlayTime()", which uses the "rtpInfo" structure. Also (as I have explained several times before), RTSP/RTP/RTCP clients (receivers) should rarely, if ever, need to look at RTP timestamps, sequence numbers, or RTCP packet data. Our receiving code automatically uses this information itself, to give you a properly-synchronized presentation time for each incoming frame of data. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Thu Aug 7 17:02:54 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Aug 2008 01:02:54 +0100 Subject: [Live-devel] Help with openrtsp In-Reply-To: <4f96b010808070709hc1baad3ma7e86fa93286eb5b@mail.gmail.com> References: <4f96b010808062204u4303381aia0f19fa1f78b15bb@mail.gmail.com> <4f96b010808070709hc1baad3ma7e86fa93286eb5b@mail.gmail.com> Message-ID: >I knew that my server was not responding Your server *was* responding. It was responding with a "404 Not Found" error message. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Thu Aug 7 17:06:11 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Aug 2008 01:06:11 +0100 Subject: [Live-devel] Live555 with C# In-Reply-To: <489B2AF20200002D0008EFE0@gwiasmtp.uct.ac.za> References: <489B2AF20200002D0008EFE0@gwiasmtp.uct.ac.za> Message-ID: >Is it possible to use Live555 with C# I don't know, but perhaps. Our code is all C++, so if it's possible to call C++ libraries from C#, then it should be possible. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Thu Aug 7 17:12:03 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Aug 2008 01:12:03 +0100 Subject: [Live-devel] Fwd: tips for building on visual studio 2008 Message-ID: >Subject: tips for building on visual studio 2008 >Date: Tue, 29 Jul 2008 15:31:30 -0400 >Thread-Topic: tips for building on visual studio 2008 >thread-index: AcjxsbOVWgLA2blDQwSSgevvPx7zWQ== >From: "Andrew Buchanan" >To: >X-Spambayes-Classification: ham >X-Spambayes-Spam-Probability: 0.00 >X-Spambayes-MailId: 1217368566 > >I was able to build just fine, I just thought I'd contribute the >required steps for anybody else interested. > >1. Follow instructions located here: >http://letsgoustc.spaces.live.com/blog/cns!89AD27DFB5E249BA!167.entry >2. Modify TOOLS_32 directory to the equivalent directory in a VS2008 >Installation ('C:\Program Files\Microsoft Visual Studio 9.0\VC'). > >Eg batch file is >----------------------------------- >call "C:\Program Files\Microsoft Visual Studio 9.0\VC\vcvarsall.bat" >cd liveMedia >nmake /B -f liveMedia.mak >cd ../groupsock >nmake /B -f groupsock.mak >cd ../UsageEnvironment >nmake /B -f UsageEnvironment.mak >cd ../BasicUsageEnvironment >nmake /B -f BasicUsageEnvironment.mak >cd ../testProgs >nmake /B -f testProgs.mak >cd ../mediaServer >nmake /B -f mediaServer.mak >----------------------------------- > >3. Modify the cpp files in groupsock to point to instead of > or . > >Put the batch file in the main live directory. Run the batch file. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Chris.Burns at abdeus.com Thu Aug 7 18:37:47 2008 From: Chris.Burns at abdeus.com (Chris Burns) Date: Fri, 8 Aug 2008 13:37:47 +1200 Subject: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header In-Reply-To: References: Message-ID: The crux of my question concerns the *delta* that a reference implementation (Helix USM) applies to one of the rtptime=X values that is reported in the RTP-Info header. All of our testing (with ~30 different brand/model combinations) has shown that this is critical for many mobile phones to correctly apply A/V sync. The live555 code (the 2008.07.25 release) does not adjust (calculate a delta) the RTP-Info header in this way. It simply reports the seqNum & rtptime of the first packet on each stream. And while you may think that RTSP/RTP/RTCP clients should not use the RTP timestamps, these phones are doing just that. Wouldn't it be nice if we could play nicely with them too? Cheers & thanks for all your hard work and excellent code. ChrisB Chris Burns M: +64 21 391 286 chris.burns at abdeus.com ________________________________ From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson Sent: Friday, 08 August 2008 12:02 To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header I'm specifically trying to work out how to create the "RTP-Info" header in a RTSP/PLAY-Response. Our server implementation does this automatically (see "RTSPServer.cpp", and search for "rtpInfo"). You shouldn't have to do anything yourself to generate this. Similarly, at the client end, the information is filled in automatically - if you wish, you can just access the "MediaSubsession::rtpInfo" structure. Most clients, however, will not need to access this structure directly. Instead, if they wish to get the current 'normal play time' (NPT) for the stream, they can just call "MediaSubsession::getNormalPlayTime()", which uses the "rtpInfo" structure. Also (as I have explained several times before), RTSP/RTP/RTCP clients (receivers) should rarely, if ever, need to look at RTP timestamps, sequence numbers, or RTCP packet data. Our receiving code automatically uses this information itself, to give you a properly-synchronized presentation time for each incoming frame of data. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From rajeshkumar.r at imimobile.com Fri Aug 8 01:33:08 2008 From: rajeshkumar.r at imimobile.com (rajesh) Date: Fri, 8 Aug 2008 14:03:08 +0530 Subject: [Live-devel] unsubscribe Message-ID: HI Ross , Kindly unsubscribe this email id from ur mailing list . I will add new email Id with ur mailing list . Thanks Thanks and Regards Rajesh Kumar Sr. Software Engineer R & D - Network Group ImiMobile Pvt Ltd +91 40 23555945 - 235 +91 99084 00027 www.imimobile.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrbrain at email.it Fri Aug 8 02:29:14 2008 From: mrbrain at email.it (MrBrain) Date: Fri, 8 Aug 2008 11:29:14 +0200 Subject: [Live-devel] is there a function called whenever a RTP packet is received? Message-ID: <275b82a01a08de59fe6e4be449770400@79.9.244.157> Hello, I need I'd like to know if there is a function called whenever a RTP packet is received. I need it so I can calculate the ratio of the RTP packet received. Thanks in advance. -- Email.it, the professional e-mail, gratis per te: http://www.email.it/f Sponsor: EmailBlog: news, curiosit?, tendenze dalla rete ... e le tue opinioni! Clicca qui: http://adv.email.it/cgi-bin/foclick.cgi?mid=8138&d=20080808 -------------- next part -------------- An HTML attachment was scrubbed... URL: From Melvin_Raj at satyam.com Fri Aug 8 04:22:47 2008 From: Melvin_Raj at satyam.com (Melvin_Raj) Date: Fri, 8 Aug 2008 19:22:47 +0800 Subject: [Live-devel] Help with link error In-Reply-To: <005501c8f8bb$9b3bf5f0$d1b3e1d0$@com> References: <000a01c8f728$789f7dd0$69de7970$@com> <001401c8f7f3$3543d290$9fcb77b0$@com> <005501c8f8bb$9b3bf5f0$d1b3e1d0$@com> Message-ID: Hi Roland, Thank you for your suggestion..btw, 239.255.42.42 stand for the address of the multicast server (broadcasted using mpegstreamer.exe) Im streaming a pre-recorded mpeg file.. Regards Melvin From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Roland Sent: Friday, August 08, 2008 2:30 AM To: 'LIVE555 Streaming Media - development & use' Subject: Re: [Live-devel] Help with link error Hi Melvin You could... add the command line parameter as the parameter for debugging in VS2005, then start RTSP.exe from within VisualStudio and step through the code (I think F10 is the default for step, and F11 for step into), set a lot of breakpoints, dig in and prepare to learn a lot from someone else's code. Another easy one is to search for "Failed to get a SDP" in the source code, isolate the print/output statement, and set a breakpoint there (F9). Then work your way backwards in the code and set breakpoints where you think everything should still be ok. Once you hit breakpoints, inspect variables and try to understand if the values stored in them make sense or not. I'll leave it to someone with RTSP experience to troubleshoot the other side of the equation (what's sitting at 239.255.42.42?)... Roland From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Melvin_Raj Sent: Wednesday, August 06, 2008 8:03 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Help with link error Hyee Roland, Thank you for your help..managed to solve the errors and learn new thing at the same time...now im getting another error when running the openrtsp program from command line: [cid:image001.png at 01C8F98C.23E44800] What do I have to do??thank you for your help...do have a working example that I can see(the print screens perhaps)??i want to see how it works..thank you :) ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 6537 bytes Desc: image001.png URL: From Jerry.Johns at nuvation.com Fri Aug 8 08:07:31 2008 From: Jerry.Johns at nuvation.com (Jerry Johns) Date: Fri, 8 Aug 2008 08:07:31 -0700 Subject: [Live-devel] Proper subnet for multicast streaming Message-ID: <274F7B50569A1B4C9D7BCAB17A9C7BE104ABA4@mailguy3.skynet.nuvation.com> Hello, Until now, I've always been doing my streaming from my embedded liveMedia boxes to my PC, and they all been tied into my company switch on the 192.168.202.xx subnet (255.255.255.0 mask) VLC has always picked up the stream properly, and have had no problems. When I moved my boxes to a local switch on my desk that I plug into my PC using a separate NIC, I've been having problems since. The boxes are all assigned (including the PC NIC) on 10.0.10.xx subnet, with 255.255.255.0 as the mask. I have an encoder box, and a decoder box both of which run liveMedia (server and client), and the receiver can receive the stream from the encoder without problems. However VLC player on my PC seems not to be able to play the stream anymore - it connects to the stream and I can see some initial information about the stream (h.264 stream, it can get the SDP info and all that fine, most probably since its TCP) but it can't seem to play the stream; I used ethereal, and I can see the UDP packets floating by Any help would be appreciated, Jerry Johns Design Engineer Nuvation Research Corp - Canada Tel: (519) 746-2304 ext. 225 www.nuvation.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Jerry.Johns at nuvation.com Fri Aug 8 14:41:52 2008 From: Jerry.Johns at nuvation.com (Jerry Johns) Date: Fri, 8 Aug 2008 14:41:52 -0700 Subject: [Live-devel] Re-architecturing MPEG4 ES Framer Message-ID: <274F7B50569A1B4C9D7BCAB17A9C7BE104ABA6@mailguy3.skynet.nuvation.com> Hello, I'm currently trying to get MPEG4 ES Streaming from a real-time encoder implemented on an embedded ARM platform - did some analysis on the current MPEG4VideoStreamFramer and its associated parsers - although it's a very well designed (and robust) implementation, it slightly inefficient for our needs; 1) There are two levels of memory copying - one from the ByteStreamFileSource (or whatever source you have) to the StreamParser's double banks, and then from there to the actual fTo location where MultiFramedSink will pick it up and pack/send it. This is unacceptable since I'll already be doing another memory copying from the encoder output memory location to a circular buffer from which LiveMedia picks up its data; this amounts to wasteful memory bandwidth 2) The output of our encoder gives us discrete VOP packets (after the initial VOSS, and VOL headers) that we know the size to - this allows for efficient DMA and to a lesser extent, memcopies. The MPEG4VideoStreamParser goes through byte by byte in its traversal for the start codes, which is not required in our application Knowing these two things, what do you suggest as a course of action? I was thinking of making my own MPEG4VideoStreamFramer class (named differently) that instantiates perhaps a BufferStreamParser? (The data input is resident in a FIFO buffer). Ultimately, I would like to copy from the circular buffer directly to the final fTo location, and to do that without going through the stream byte by byte (not needed, since VOP packets can be transferred wholely to the fTo buffer) Any help will be appreciated, Thanks Jerry Johns Design Engineer Nuvation Research Corp - Canada Tel: (519) 746-2304 ext. 225 www.nuvation.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Fri Aug 8 12:02:28 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Aug 2008 12:02:28 -0700 Subject: [Live-devel] Regarding FileSink destructor In-Reply-To: References: Message-ID: Sorry for the delay in responding to this. Yes, this is a bug; it will be fixed in the next release of the software. (Ditto for the "MP3FileSource" issue that you also reported.) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From Jerry.Johns at nuvation.com Fri Aug 8 21:22:09 2008 From: Jerry.Johns at nuvation.com (Jerry Johns) Date: Fri, 8 Aug 2008 21:22:09 -0700 Subject: [Live-devel] Re-architecturing MPEG4 ES Framer Message-ID: <274F7B50569A1B4C9D7BCAB17A9C7BE104ABA9@mailguy3.skynet.nuvation.com> I'm sorry, l was blind enough not to have read the FAQ :P The discrete framers, combined with a suitable FramedSource derivate should do the trick Thanks, Jerry Johns Design Engineer Nuvation Research Corp - Canada Tel: (519) 746-2304 ext. 225 www.nuvation.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Fri Aug 8 22:22:17 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Aug 2008 22:22:17 -0700 Subject: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header In-Reply-To: References: Message-ID: First, let's make sure that we're talking about the same thing. I assume that you're referring to the way that our RTSP *server* implementation fills in the "RTP-Info:" header that it includes in the response to each RTSP "PLAY" request. If you're instead referring to something else, then please correct me. >The crux of my question concerns the *delta* that a reference >implementation (Helix USM) applies to one of the rtptime=X values >that is reported in the RTP-Info header. All of our testing (with >~30 different brand/model combinations) has shown that this is >critical for many mobile phones to correctly apply A/V sync. What exactly is this 'delta', and which RFC (or Internet-Draft) document defines it? >The live555 code (the 2008.07.25 release) does not adjust (calculate >a delta) the RTP-Info header in this way. It simply reports the >seqNum & rtptime of the first packet on each stream. I believe that the information that we put in the "RTP-Info:" header is correct. It is the information defined in section 12.33 of RFC 2326 (the RTSP specification). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Fri Aug 8 22:24:37 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 8 Aug 2008 22:24:37 -0700 Subject: [Live-devel] unsubscribe In-Reply-To: References: Message-ID: Requests to unsubscribe from a mailing list should *never* be sent to the mailing list itself. For this mailing list, you unsubscribe using the list's web page (the one whose address appears at the bottom of every list message): http://lists.live555.com/mailman/listinfo/live-devel -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From pathos821 at hotmail.com Sat Aug 9 02:47:50 2008 From: pathos821 at hotmail.com (John Smith) Date: Sat, 9 Aug 2008 09:47:50 +0000 Subject: [Live-devel] Using OpenRTSP for receiving H264 Message-ID: Hi, I test OpenRTSP for the URL rtsp://207.235.112.116/h264test2.mp4. Two files are created and OpenRTSP displays following messages; Started playing sessionReceiving streamed data (for up to 376.541992 seconds)... But the file size is 0 after the teardown message appeared. If I tried local live55MediaServer for some mp3 files, openRTSP fills mp3 file. What's wrong with my test? Thanks in advance. Yong _________________________________________________________________ Get Windows Live and get whatever you need, wherever you are. Start here. http://www.windowslive.com/default.html?ocid=TXT_TAGLM_WL_Home_082008 -------------- next part -------------- An HTML attachment was scrubbed... URL: From linux_is_next at hotmail.com Sat Aug 9 07:28:50 2008 From: linux_is_next at hotmail.com (bos marcel) Date: Sat, 9 Aug 2008 16:28:50 +0200 Subject: [Live-devel] question In-Reply-To: References: <489B2AF20200002D0008EFE0@gwiasmtp.uct.ac.za> Message-ID: hello My name is marcel and i have a question.. does live555 suport 3G streaming? -------------------------------------------- http://culture.zapto.org allt du vill veta > Date: Fri, 8 Aug 2008 01:06:11 +0100 > To: live-devel at ns.live555.com > From: finlayson at live555.com > Subject: Re: [Live-devel] Live555 with C# > > >Is it possible to use Live555 with C# > > I don't know, but perhaps. Our code is all C++, so if it's possible > to call C++ libraries from C#, then it should be possible. > -- > > 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 _________________________________________________________________ Tr?tt p? jobbet? Hitta nya utmaningar h?r! http://msn.jobbguiden.se/jobseeker/resumes/postresumenew/postresumestart.aspx?sc_cmp2=JS_INT_SEMSN_NLPCV -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Sat Aug 9 08:18:50 2008 From: finlayson at live555.com (Ross Finlayson) Date: Sat, 9 Aug 2008 08:18:50 -0700 Subject: [Live-devel] Using OpenRTSP for receiving H264 In-Reply-To: References: Message-ID: >Hi, > >I test OpenRTSP for the URL rtsp://207.235.112.116/h264test2.mp4. >Two files are created and OpenRTSP displays following messages; > >Started playing session >Receiving streamed data (for up to 376.541992 seconds)... > >But the file size is 0 after the teardown message appeared. > >If I tried local live55MediaServer for some mp3 files, openRTSP >fills mp3 file. > >What's wrong with my test? Please read the FAQ! -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Sat Aug 9 08:24:06 2008 From: finlayson at live555.com (Ross Finlayson) Date: Sat, 9 Aug 2008 08:24:06 -0700 Subject: [Live-devel] Proper subnet for multicast streaming In-Reply-To: <274F7B50569A1B4C9D7BCAB17A9C7BE104ABA4@mailguy3.skynet.nuvation.com> References: <274F7B50569A1B4C9D7BCAB17A9C7BE104ABA4@mailguy3.skynet.nuvation.com> Message-ID: >Content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C8F968.7AE5399F" > >Hello, > Until now, I've always been doing my streaming from my >embedded liveMedia boxes to my PC, and they all been tied into my >company switch on the 192.168.202.xx subnet (255.255.255.0 mask) >VLC has always picked up the stream properly, and have had no problems. > >When I moved my boxes to a local switch on my desk that I plug into >my PC using a separate NIC, I've been having problems since. The >boxes are all assigned (including the PC NIC) on 10.0.10.xx subnet, >with 255.255.255.0 as the mask. I have an encoder box, and a decoder >box both of which run liveMedia (server and client), and the >receiver can receive the stream from the encoder without problems. > >However VLC player on my PC seems not to be able to play the stream >anymore - it connects to the stream and I can see some initial >information about the stream (h.264 stream, it can get the SDP info >and all that fine, most probably since its TCP) but it can't seem to >play the stream; I used ethereal, and I can see the UDP packets >floating by > >Any help would be appreciated, Does *any* multicast traffic reach your destination network from your source network? Perhaps you just have a multicast routing problem between them? If so, then maybe you should switch to using unicast streaming rather than multicast (e.g., use "testOnDemandRTSPServer" as a model). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Sat Aug 9 08:29:27 2008 From: finlayson at live555.com (Ross Finlayson) Date: Sat, 9 Aug 2008 08:29:27 -0700 Subject: [Live-devel] question In-Reply-To: References: <489B2AF20200002D0008EFE0@gwiasmtp.uct.ac.za> Message-ID: >does live555 suport 3G streaming? It depends what you mean by "3G streaming". "3G" is mostly a marketing term, so you need to be more specific about what you want to do. Do you want to receive streams, or send them? -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From pathos821 at hotmail.com Sat Aug 9 13:27:24 2008 From: pathos821 at hotmail.com (John Smith) Date: Sat, 9 Aug 2008 20:27:24 +0000 Subject: [Live-devel] Using OpenRTSP for receiving H264 In-Reply-To: References: Message-ID: Good reply. Thanks> Date: Sat, 9 Aug 2008 08:18:50 -0700> To: live-devel at ns.live555.com> From: finlayson at live555.com> Subject: Re: [Live-devel] Using OpenRTSP for receiving H264> > >Hi,> >> >I test OpenRTSP for the URL rtsp://207.235.112.116/h264test2.mp4.> >Two files are created and OpenRTSP displays following messages;> >> >Started playing session> >Receiving streamed data (for up to 376.541992 seconds)...> >> >But the file size is 0 after the teardown message appeared.> >> >If I tried local live55MediaServer for some mp3 files, openRTSP > >fills mp3 file.> >> >What's wrong with my test?> > Please read the FAQ!> -- > > 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 _________________________________________________________________ Got Game? Win Prizes in the Windows Live Hotmail Mobile Summer Games Trivia Contest http://www.gowindowslive.com/summergames?ocid=TXT_TAGHM -------------- next part -------------- An HTML attachment was scrubbed... URL: From mamille1 at rockwellcollins.com Sun Aug 10 15:41:32 2008 From: mamille1 at rockwellcollins.com (mamille1 at rockwellcollins.com) Date: Sun, 10 Aug 2008 17:41:32 -0500 Subject: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header In-Reply-To: Message-ID: All The mention of "properly-synchronized presentation time" sounds like it might be related to a problem we are having. We have an application using the live555 library running on an embedded processor that record a digital audio/video stream to a file. The stream is generated by a separate DSP that converts analog audio and video and sends the stream out on a private network. Recordings look pretty good when played back on a PC (with VLC, for example). Playing back to the DSP (for conversion from digital to analog) looks much worse -- artifacts that we think are due to dropped frames and packets. We recently confirmed that our embedded processor and associated DSP do not share a time reference, and there is no real-time clock in our embedded processor -- we just use the CPU's clock for our time reference. We think that this is what's happening during a recording: - our DSP generates an RTP stream using its own time reference to create timestamps and measure time intervals - our embedded processor decodes this stream but does not rewrite timestamp info, so that the time info embedded in the file is our DSP's measurement of time During playback on a PC, the timestamp info from the file is used. The file plays back correctly because our DSP's measurement of time is pretty close to the PC's measurement of time, due to the PC's real-time clock. During playback from our embedded processor to our DSP, our embedded processor's measurement of time is used when reading timestamp info from the recorded file. After timing the playback of several files, it looks like our embedded processor is running about 2% faster relative to our DSP. This causes our processor to send packets and frames to our DSP slightly faster than our DSP can process it, leading to dropped packets and frames. We're digging into the liveMedia code to understand where we might make a subclass to compensate for this on playback. MPEG2TransportStreamFramer.cpp seems a good place to start, since it's already working with the PCR. As near as we can tell right now, however, the MPEG2TransportStreamFramer class is used during recording and playback in our system, so we need to be very careful with changes here. We're also looking into implementing our own gettimeofday() function, where we could apply this compensation during a playback. Since this could also affect recordings, we'd need to be sure that we had turned compensation off during recording and turned it back on only during a playback. We would appreciate any advice on where we should focus to solve this problem. Thanks! -=- Mike Miller Rockwell Collins, Inc. Cedar Rapids, IA -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Sun Aug 10 16:21:01 2008 From: finlayson at live555.com (Ross Finlayson) Date: Sun, 10 Aug 2008 16:21:01 -0700 Subject: [Live-devel] RockwellCollins playback problem In-Reply-To: References: Message-ID: (I've changed the Subject: line to something more appropriate.) >when reading timestamp info from the recorded file. After timing the >playback of several files, it looks like our embedded processor is >running about 2% faster relative to our DSP. This causes our >processor to send packets and frames to our DSP slightly faster than >our DSP can process it, leading to dropped packets and frames. It seems unlikely that 'sending packets 2% too fast' should cause these problems. The receiver (which will use the presentation times generated by the sender) would just decode/display the frames 2% faster than it normally would, and this shouldn't be a problem. Are you sure that your problems aren't caused by inadequately large receive buffers in your receiver's OS? See -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From Bernhard at Feiten.de Sun Aug 10 23:39:57 2008 From: Bernhard at Feiten.de (Bernhard Feiten) Date: Mon, 11 Aug 2008 08:39:57 +0200 Subject: [Live-devel] Media stream duplication on a streaming relay References: <1134049882.43983a5aa6b32@imp4-g19.free.fr><6.2.3.4.1.20051208104044.032296d0@localhost><00e901c8d74d$6de53290$d302a8c0@Popelfranz> <00b201c8f852$b7388340$d302a8c0@Popelfranz> Message-ID: <00d301c8fb7d$1389c960$d302a8c0@Popelfranz> Dear Ross, all, attached you find a first uncomplete version of the StreamDup classes. Perhpas you have some comments. Is this in principle a way how to do it? I'm quite unsure what needs to be regarded concerning the getNextFrame, aftergetting filter chain mechanism. Audio is working, but when I also activate the video branch it crashes in H264VideoRTPSink Must be error, perhaps also somwhere else in my code. Thank you for your comments, Bernhard ----- Original Message ----- From: "Bernhard Feiten" To: "LIVE555 Streaming Media - development & use" Sent: Thursday, August 07, 2008 7:59 AM Subject: Re: [Live-devel] Media stream duplication on a streaming relay > Dear Ross, all, > > as nobody answered I started to write a StreamDup class following your > hints from the e-mail below. > > Related to that I have two questions? > Would it be required that the sequnce number and timestamp of the > duplicates are newly initialized? > Would RTCP also need to be considered then? > > Thank you in advance, > Bernhard > > > ----- Original Message ----- > From: "Bernhard Feiten" > To: "LIVE555 Streaming Media - development & use" > > Sent: Thursday, June 26, 2008 7:28 AM > Subject: Re: [Live-devel] Media stream duplication on a streaming relay > > >> Dear Ross, all, >> >> did somebody wrote the StreamDuplication class you proposed below >> already? >> >> Do you have a hint how the getNextFrame functions could be synchronized? >> >> Thank you very much, >> Bernhard >> >> >> >> >> ----- Original Message ----- >> From: "Ross Finlayson" >> To: "LIVE555 Streaming Media - development & use" >> >> Sent: Thursday, December 08, 2005 8:57 PM >> Subject: Re: [Live-devel] Media stream duplication on a streaming relay >> >> >>> As you noticed, you can't duplicate a stream by having each recipient >>> read from a single object, because "getNextFrame()" can't be called more >>> than once on the same object concurrently. >>> >>> The solution, instead, is to create a separate object (of some >>> "FramedSource" subclass) for each recipient. This new class would >>> implement the "doGetNextFrame()" virtual function by somehow >>> 'registering' with the data source object - to request a copy of the >>> next incoming frame. >>> >>> The data source object (which would *not* be a subclass of >>> "FramedSource", and so would not implement "doGetNextFrame()") would >>> handle these requests by delivering copies of each incoming frame to >>> each recipient, and not ask for a new incoming frame (from its upstream >>> data source) until it has finished delivering data to each downstream >>> recipient. >>> >>> For an example of code that is similar to this, note the relationship >>> between the "MPEG1or2Demux" and "MPEG1or2DemuxedElementaryStream" >>> classes. (In this case, however, we are demultiplexing data to >>> downstream recipients, rather than duplicating it.) Note in particular >>> that (i) "MPEG1or2Demux" is subclassed from "Medium", not >>> "FramedSource", and (ii) "MPEG1or2Demux" implements a "getNextFrame()" >>> function that is similar to, but different from >>> "FramedSource::getNextFrame()". >>> >>> (This functionality (data duplication) is something that should probably >>> be added to the library someday. Until then, however, you will need to >>> implement it yourself.) >>> >>> >>> Ross Finlayson >>> Live Networks, Inc. (LIVE555.COM) >>> >>> >>> _______________________________________________ >>> live-devel mailing list >>> live-devel at lists.live555.com >>> http://lists.live555.com/mailman/listinfo/live-devel >> >> _______________________________________________ >> live-devel mailing list >> live-devel at lists.live555.com >> http://lists.live555.com/mailman/listinfo/live-devel > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: StreamDup.h URL: -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: testStreamDup.cpp URL: -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: StreamDup.cpp URL: From prbhagwat at gmail.com Mon Aug 11 01:30:00 2008 From: prbhagwat at gmail.com (Pramod Bhagwat) Date: Mon, 11 Aug 2008 14:00:00 +0530 Subject: [Live-devel] redeclaration in RTSPOverHTTPServer Message-ID: Hi Ross, This bug is similar to the bug reported by Renato MAURO for RTSPServer. In RTSPOverHTTPServer::createNew function present in RTSPOverHTTPServer.cpp file "ourSocket" is declared twice, inner and outer cycle. Warm Regards, pramod From Jerry.Johns at nuvation.com Mon Aug 11 08:21:00 2008 From: Jerry.Johns at nuvation.com (Jerry Johns) Date: Mon, 11 Aug 2008 08:21:00 -0700 Subject: [Live-devel] Proper subnet for multicast streaming Message-ID: <274F7B50569A1B4C9D7BCAB17A9C7BE104ABAE@mailguy3.skynet.nuvation.com> > Does *any* multicast traffic reach your destination network from your > source network? Perhaps you just have a multicast routing problem > between them? What do you mean by traffic reaching from destination network to the source network? The PC and the 2 embedded LiveMedia boxes exist on a single network, i.e the 10.0.10.x subnet. One of the boxes is an RTSP Server, and the other a client. The client is able to receive and decode the stream just fine. Its just VLC on Windows XP that cannot seem to decode the stream. On Ethereal, I can see the flurry of multicast traffic coming from the embedded box, which means it's visible on Windows It just seems VLC cannot receive the stream. The same behavior is noted in QuickTime Note that if I change all the IP addresses (2 boxes + PC) to be on the 192.168.202.x subnet, VLC is able to decode it. Jerry Johns Design Engineer Nuvation Research Corp - Canada Tel: (519) 746-2304 ext. 225 www.nuvation.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From DionG at Viewcast.com Mon Aug 11 08:56:17 2008 From: DionG at Viewcast.com (Dion Galbreath) Date: Mon, 11 Aug 2008 10:56:17 -0500 Subject: [Live-devel] Set Up Darwin for Darwin Injector. Getting Unathorized on setDestination Message-ID: I am trying to get the Darwin injector code to work, I am using the http://www.live555.com/liveMedia/doxygen/html/testMPEG4VideoToDarwin_8cp p-source.html example, however I was getting the following error injector->setDestination() failed: [ cannot handle ANNOUNCE response: RTSP/1.0 401 Unauthorized ] so I then changed it to use a username and password for my admin login on the server, however I still get same err.. is there something I'm missing? -------------- next part -------------- An HTML attachment was scrubbed... URL: From nshamshiva at gmail.com Mon Aug 11 09:58:59 2008 From: nshamshiva at gmail.com (Shiva Shankar N) Date: Mon, 11 Aug 2008 13:58:59 -0300 Subject: [Live-devel] Quick time option and Bitrate Message-ID: <4f96b010808110958l57a48431t302cc5b198833d4b@mail.gmail.com> Hi all, I am facing some problems : 1. If I use the option -q (quicktime format) for openRSTP and record the streaming video. I am not able to play the recorded video in Quicktime player. And my Command line agruments is as follows: openRTSP -b 50000 -q -v -d 20 rtsp://1xx.1xx.1xx.104/mpeg4/media.amp > stream.mp4 2. I am using Axis cameras, so i can select the option of displaying bitrate & time at the bottom of the streaming video. After recording video for 10sec, when i caluclate the downloaded bitrate based on the file size, it doesn't match with the displayed bitrate in the video. why is this ? Thanks sham -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Mon Aug 11 10:43:31 2008 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 11 Aug 2008 10:43:31 -0700 Subject: [Live-devel] Quick time option and Bitrate In-Reply-To: <4f96b010808110958l57a48431t302cc5b198833d4b@mail.gmail.com> References: <4f96b010808110958l57a48431t302cc5b198833d4b@mail.gmail.com> Message-ID: >1. If I use the option -q (quicktime format) for openRSTP and record >the streaming video. >I am not able to play the recorded video in Quicktime player. >And my Command line agruments is as follows: > >openRTSP -b 50000 -q -v -d 20 >rtsp://1xx.1xx.1xx.104/mpeg4/media.amp > stream.mp4 From the FAQ: "If the session contains a video subsession, you should also use the "-w ", "-h " and "-f " options to specify the width and height (in pixels), and frame rate (per-second) of the corresponding video track. (If these options are omitted, then the values width=240 pixels; height=180 pixels; frame-rate=15 are used.) These values are important; if they are not correct, your file might not play at all! " -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Mon Aug 11 10:49:04 2008 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 11 Aug 2008 10:49:04 -0700 Subject: [Live-devel] Set Up Darwin for Darwin Injector. Getting Unathorized on setDestination In-Reply-To: References: Message-ID: >Content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C8FBCA.D2961690" > >I am trying to get the Darwin injector code to work, I am using the >http://www.live555.com/liveMedia/doxygen/html/testMPEG4VideoToDarwin_8cpp-source.html >example, however I was getting the following error > >injector->setDestination() failed: [ cannot handle ANNOUNCE >response: RTSP/1.0 401 Unauthorized ] > >so I then changed it to use a username and password for my admin >login on the server, however I still get >same err.. You need to set up permission to *write* data to the Darwin server (not just to stream data from it). (For some tips on how to do this, see the comment at the top of "liveMedia/include/DarwinInjector.hh".) I note yet again, however, that we have our own RTSP server implementation, so you really don't need to be using a separate Darwin Streaming Server. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Mon Aug 11 10:51:03 2008 From: finlayson at live555.com (Ross Finlayson) Date: Mon, 11 Aug 2008 10:51:03 -0700 Subject: [Live-devel] redeclaration in RTSPOverHTTPServer In-Reply-To: References: Message-ID: >This bug is similar to the bug reported by Renato MAURO for RTSPServer. >In RTSPOverHTTPServer::createNew function present in >RTSPOverHTTPServer.cpp file "ourSocket" is declared twice, inner and >outer cycle. Yes, you're right - thanks for noticing this. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From Jerry.Johns at nuvation.com Mon Aug 11 12:16:14 2008 From: Jerry.Johns at nuvation.com (Jerry Johns) Date: Mon, 11 Aug 2008 12:16:14 -0700 Subject: [Live-devel] Calculating rtptime for RTSP/PLAY-Response "RTP-Info" header Message-ID: <274F7B50569A1B4C9D7BCAB17A9C7BE104ABB2@mailguy3.skynet.nuvation.com> I might be able to help you here - We have a similar setup here where we use an ARM+DSP to do our real-time encoding/decoding work with AV Sync for audio+video (DaVinci platform) First of all, let me get the task partitioning here straight, what are the specific roles of the DSP and your GPP (general purpose processor, i.e ur embedded processor)? A sensible approach would be to relegate decoding/encoding activities to the DSP, and control oriented tasks to run on the GPP (i.e LiveMedia, synchronization, etc) Does the DSP also take care of outputting the analog stream? I'm assuming you have a buffered interface between the GPP and the DSP - that is absolutely required when crossing two clock domains as you have. There should be some feedback from the DSP to the GPP regarding how fast/slow its decoding samples - this should allow the GPP to ensure the buffer never under-runs or over-runs Compensating using hacky techniques like modifying gettimeofday() will only yield further problems down the line. Jerry Johns Design Engineer Nuvation Research Corp - Canada Tel: (519) 746-2304 ext. 225 www.nuvation.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mamille1 at rockwellcollins.com Mon Aug 11 14:11:12 2008 From: mamille1 at rockwellcollins.com (mamille1 at rockwellcollins.com) Date: Mon, 11 Aug 2008 16:11:12 -0500 Subject: [Live-devel] RockwellCollins playback problem In-Reply-To: Message-ID: Jerry > what are the specific roles of the DSP and your GPP (general purpose processor, The DSP handles the encoding of an analog input to a digital stream, and also the decoding of a digital stream to analog output. The GPP handles control tasks -- set up of streams, working with files, etc. > I'm assuming you have a buffered interface between the GPP and the DSP - > that is absolutely required when crossing two clock domains as you have. Yes, we do have buffers. As Mr. Finlayson pointed out, the buffer sizes on the receiving end may not be large enough. > There should be some feedback from the DSP to the GPP regarding how fast/slow its decoding samples We're working on that. At the moment we don't have a accurate feedback from the decoding side. > Compensating using hacky techniques like modifying gettimeofday()... We completely agree. For the moment, however, this gives us a quick ability to get closer to matching rates. We are looking into where frame rates and bit rates on the sending side (the GPP) can be controlled. Once we do get better info from the DSP about how much data it's able to process we want to dynamically adjust the data rates to match. Thanks! -=- Mike Miller Rockwell Collins, Inc. Cedar Rapids, IA -------------- next part -------------- An HTML attachment was scrubbed... URL: From wade.fs at gmail.com Tue Aug 12 19:15:47 2008 From: wade.fs at gmail.com (wade) Date: Wed, 13 Aug 2008 10:15:47 +0800 Subject: [Live-devel] Does live555MediaServer support IPv6? Message-ID: Hi: Does live555MediaServer support IPv6? I tried it but failed. I think must rewrite some files such as groupsock/GrouosockHelper.cpp, groupsock/include/NetAddress.hh for example. Am I right? -- ?????, ????? Free Spirit, Fantasy Space -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Wed Aug 13 00:28:43 2008 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 13 Aug 2008 00:28:43 -0700 Subject: [Live-devel] Does live555MediaServer support IPv6? In-Reply-To: References: Message-ID: > Does live555MediaServer support IPv6? No, not yet. It's on the 'to do' list, but will be a substantial undertaking (not just modifications to a couple of files). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From Melvin_Raj at satyam.com Wed Aug 13 00:51:43 2008 From: Melvin_Raj at satyam.com (Melvin_Raj) Date: Wed, 13 Aug 2008 15:51:43 +0800 Subject: [Live-devel] help with displaying video content on mpegreceiver In-Reply-To: References: Message-ID: Hello Ross, Is the anyway I can display the received data on the receiving side??for example, if im transmitting a video file from mpegstreamer, can I view it in the mpegreceiver window??? Thank You. DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. From Melvin_Raj at satyam.com Wed Aug 13 05:00:45 2008 From: Melvin_Raj at satyam.com (Melvin_Raj) Date: Wed, 13 Aug 2008 20:00:45 +0800 Subject: [Live-devel] Playback using live55 In-Reply-To: References: Message-ID: Is there any example codes of live555 that is able to playback video? (maybe a pre-recorded video file?) Thank You.. -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson Sent: Wednesday, August 13, 2008 3:29 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Does live555MediaServer support IPv6? > Does live555MediaServer support IPv6? No, not yet. It's on the 'to do' list, but will be a substantial undertaking (not just modifications to a couple of files). -- 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 DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. From slaine at slaine.org Wed Aug 13 06:22:51 2008 From: slaine at slaine.org (Glen Gray) Date: Wed, 13 Aug 2008 14:22:51 +0100 Subject: [Live-devel] Playback using live55 In-Reply-To: References: Message-ID: VLC uses live555 for playback of rtsp streams. Check out the code for that in modules/demux/live555.cpp -- Glen Gray slaine at slaine.org On 13 Aug 2008, at 13:00, Melvin_Raj wrote: > Is there any example codes of live555 that is able to playback > video? (maybe a pre-recorded video file?) > > Thank You.. > > -----Original Message----- > From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com > ] On Behalf Of Ross Finlayson > Sent: Wednesday, August 13, 2008 3:29 PM > To: LIVE555 Streaming Media - development & use > Subject: Re: [Live-devel] Does live555MediaServer support IPv6? > >> Does live555MediaServer support IPv6? > > No, not yet. It's on the 'to do' list, but will be a substantial > undertaking (not just modifications to a couple of files). > -- > > 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 > > > DISCLAIMER: > This email (including any attachments) is intended for the sole use > of the intended recipient/s and may contain material that is > CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance > by others or copying or distribution or forwarding of any or all of > the contents in this message is STRICTLY PROHIBITED. If you are not > the intended recipient, please contact the sender by email and > delete all copies; your cooperation in this regard is appreciated. > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel From finlayson at live555.com Wed Aug 13 06:07:30 2008 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 13 Aug 2008 06:07:30 -0700 Subject: [Live-devel] Playback using live55 In-Reply-To: References: Message-ID: >Is there any example codes of live555 that is able to playback >video? (maybe a pre-recorded video file?) Look at VLC . Note, in particular, the code that interfaces with the "LIVE555 Streaming Media" libraries: "modules/demux/live555.cpp". -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From DionG at Viewcast.com Wed Aug 13 07:30:29 2008 From: DionG at Viewcast.com (Dion Galbreath) Date: Wed, 13 Aug 2008 09:30:29 -0500 Subject: [Live-devel] RTSP Teardown Message-ID: Does live555 support 'RTSP Teardown' it seems that the call is never being sent to a QuickTime and Darwin server so I was wondering if this is supported in this library. Thanks Dion -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Wed Aug 13 10:38:08 2008 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 13 Aug 2008 10:38:08 -0700 Subject: [Live-devel] RTSP Teardown In-Reply-To: References: Message-ID: >Does live555 support 'RTSP Teardown' Yes. > it seems that the call is never being sent to a QuickTime and Darwin server Your RTSP client needs to explicitly send this request - by calling "RTSPClient::tearDownMediaSession()" or ("RTSPClient::tearDownMediaSubsession()"). Note that "openRTSP" will send this request at the end of time specified by the "-d " option, or at the end of any duration specified by by the server (in it's SDP description returned in response to "DESCRIBE"). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From Melvin_Raj at satyam.com Wed Aug 13 19:37:58 2008 From: Melvin_Raj at satyam.com (Melvin_Raj) Date: Thu, 14 Aug 2008 10:37:58 +0800 Subject: [Live-devel] Playback using live55 In-Reply-To: References: Message-ID: Thank You for your reply..i will work on the vlc code....btw, im wondering if the mpegreceiver is able to receive the audio and video stream at the same time...meaning, I tried sending both audio and video on the same port and its successful...but the receiver is only able to receive either the audio or video stream at any given time....is there any suggestion for me to receive the audio and video on the same file at the receiver's end? Ps: I also tried receiving streams usimg vlc player but it also play only either one.... Thank You.. -----Original Message----- From: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson Sent: Wednesday, August 13, 2008 9:08 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Playback using live55 >Is there any example codes of live555 that is able to playback >video? (maybe a pre-recorded video file?) Look at VLC . Note, in particular, the code that interfaces with the "LIVE555 Streaming Media" libraries: "modules/demux/live555.cpp". -- 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 DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. From finlayson at live555.com Wed Aug 13 22:43:12 2008 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 13 Aug 2008 22:43:12 -0700 Subject: [Live-devel] Playback using live55 In-Reply-To: References: Message-ID: >Thank You for your reply..i will work on the vlc code....btw, im >wondering if the mpegreceiver is able to receive the audio and video >stream at the same time There's no such application as "mpegreceiver". Assuming that you mean "testMPEG1or2VideoReceiver", then the answer is no, because that application was written specifically to receive the video stream only. If you want to receive both the video and audio streams from a MPEG-1 or 2 RTP audio+video session, then either 1/ Run both "testMPEG1or2Receiver" and "testMP3Receiver" (as separate, concurrent, applications), or (better) 2/ Use "openRTSP". >is there any suggestion for me to receive the audio and video on the >same file at the receiver's end? Use "openRTSP" with the "-q" or "-i" option. ***See the openRTSP documentation for details*** -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From mamille1 at rockwellcollins.com Thu Aug 14 13:17:37 2008 From: mamille1 at rockwellcollins.com (mamille1 at rockwellcollins.com) Date: Thu, 14 Aug 2008 15:17:37 -0500 Subject: [Live-devel] RockwellCollins playback problem and controlling bit rates In-Reply-To: Message-ID: All Thanks for the earlier advice. We think that our earlier problem with large frames in a recorded video stream was due to insufficient socket buffer space on the receiving end during a recording. ("Large frames" were 2x-5x the size of a normal I-frame or P-frame, but only a fraction of that was really picture data.) Now that we have improved recordings of our video stream, we are working to make playback of those stream files from our storage media look better. A lot of our playback problem stems from sending data to our DSP decoder slightly faster than it can handle. This leads to stutters in the playback (dropped frames) and a sort of digital splatter (blocky video until the next I-frame). We have tried a couple of ways to reduce the rate at which we send data to the decoder, including slowing our system clock down a bit. (Our CPU has no real time clock, and our DSP is not clocked at the same rate as the CPU.) We are looking at controlling the bit rate used during a recording and playback rather than adjusting system time. In our system, we can set a target bit rate used by the DSP when encoding analog to digital, but that seems to work by dropping frames. This makes our recordings look a little worse than before when played back on a PC, since we're missing some frames. Is there a mechanism in the live555 library that allows us to control the bit rate in a playback from our CPU to our DSP without dropping entire video frames? Thanks! -=- Mike Miller Rockwell Collins, Inc. Cedar Rapids, IA -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Thu Aug 14 22:58:04 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 14 Aug 2008 22:58:04 -0700 Subject: [Live-devel] RockwellCollins playback problem and controlling bit rates In-Reply-To: References: Message-ID: >Is there a mechanism in the live555 library that allows us to >control the bit rate in a playback from our CPU to our DSP without >dropping entire video frames? The best way to control the data rate of a streaming (transmitting) application is through the "fDurationInMicroseconds" field that you set in your source object's "doGetNextFrame()" implementation. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From satheesh at streamprocessors.com Fri Aug 15 06:38:00 2008 From: satheesh at streamprocessors.com (Satheesh Ram) Date: Fri, 15 Aug 2008 19:08:00 +0530 Subject: [Live-devel] green screen at the start of MPEG4 stream reception Message-ID: <48A586B8.7090806@streamprocessors.com> Hi all, While receiving and playing back MPEG4 encoded streams using VLC, the first few frames happen to be green screen. The message window shows the following error "ffmpeg warning: warning: first frame is no keyframe". The playback gets back to normal state on the arrival on next MPEG4 keyframe. I observed this with testMPEG4VideoStreamer.exe. and the issue is consistent with repeated runs. Is it because client didn't get first few frames on MPEG4 stream? where do the frames gets dropped? how to avoid losing those few frames? -- Satheesh Ram Off. +91.80.41630270x25 Mob. +91.9945211181 From finlayson at live555.com Fri Aug 15 07:51:06 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 15 Aug 2008 07:51:06 -0700 Subject: [Live-devel] green screen at the start of MPEG4 stream reception In-Reply-To: <48A586B8.7090806@streamprocessors.com> References: <48A586B8.7090806@streamprocessors.com> Message-ID: >While receiving and playing back MPEG4 encoded streams using VLC, >the first few frames happen to be green screen. The message window >shows the following error >"ffmpeg warning: warning: first frame is no keyframe". The playback >gets back to normal state on the arrival on next MPEG4 keyframe. I >observed this with testMPEG4VideoStreamer.exe. and the issue is >consistent with repeated runs. >Is it because client didn't get first few frames on MPEG4 stream? Yes, probably. > where do the frames gets dropped? Remember that "testMPEG4VideoStreamer" is a *multicast* application - it just sends the data to a multicast group, regardless of how many clients have subscribed to the group. Therefore, any data that it sends before the client runs (and subscribes to the group) will be lost. To avoid this, use unicast streaming, using "testOnDemandRTSPServer", or "live555MediaServer". -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From nshamshiva at gmail.com Fri Aug 15 17:07:25 2008 From: nshamshiva at gmail.com (Shiva Shankar N) Date: Fri, 15 Aug 2008 21:07:25 -0300 Subject: [Live-devel] green screen at the start of MPEG4 stream reception In-Reply-To: References: <48A586B8.7090806@streamprocessors.com> Message-ID: <4f96b010808151707j1a4fd82yd9ac941fb50c01f@mail.gmail.com> Hi Ross, *" To avoid this, use unicast streaming, using "testOnDemandRTSPServer", or "live555MediaServer". *" please can you explain us how different is *"testOnDemandRTSPServer", or "live555MediaServer" *with *testMPEG4VideoStreamer *in using to reduce or avoid green screen. I have given below the reason for this green screen and i think it cannot be avoided. If i am wrong let know. It has so happen that session with server has started after Key frame (I-frame) is sent & he as missed it . When the session starts it will be downloading only p-frame without Key-frames in which case there is No reference frame for these P-frame to do motion compensation and this green screen is nothing but motion estimation. You can reduce this green screen period by reducing the GOV(group of videos) or GOP (group of picture) size at the server side in camera settings. P.S : GOP or GOV - Number of P-frames between 2 consecutive I-frames(Key frames). with regards shiv On Fri, Aug 15, 2008 at 11:51 AM, Ross Finlayson wrote: > While receiving and playing back MPEG4 encoded streams using VLC, the first >> few frames happen to be green screen. The message window shows the following >> error >> "ffmpeg warning: warning: first frame is no keyframe". The playback gets >> back to normal state on the arrival on next MPEG4 keyframe. I observed this >> with testMPEG4VideoStreamer.exe. and the issue is consistent with repeated >> runs. >> Is it because client didn't get first few frames on MPEG4 stream? >> > > Yes, probably. > > where do the frames gets dropped? >> > > Remember that "testMPEG4VideoStreamer" is a *multicast* application - it > just sends the data to a multicast group, regardless of how many clients > have subscribed to the group. Therefore, any data that it sends before the > client runs (and subscribes to the group) will be lost. > > To avoid this, use unicast streaming, using "testOnDemandRTSPServer", or > "live555MediaServer". > -- > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From vivekmrathod at gmail.com Sun Aug 17 10:51:13 2008 From: vivekmrathod at gmail.com (vivek rathod) Date: Sun, 17 Aug 2008 23:21:13 +0530 Subject: [Live-devel] redirecting audio stream to form rtp frames Message-ID: <8be4676d0808171051t667da3b0y599b2b149f1e9b0f@mail.gmail.com> hi , i am doing a project on lan radio.(multicast). i need to produce rtp packets containing the audio stream. can i redirect the audio stream (of the mp3 file being played) from Mplayer and make rtp packets using live555 library. please give me some hint or reference. vivek -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Sun Aug 17 14:04:52 2008 From: finlayson at live555.com (Ross Finlayson) Date: Sun, 17 Aug 2008 14:04:52 -0700 Subject: [Live-devel] redirecting audio stream to form rtp frames In-Reply-To: <8be4676d0808171051t667da3b0y599b2b149f1e9b0f@mail.gmail.com> References: <8be4676d0808171051t667da3b0y599b2b149f1e9b0f@mail.gmail.com> Message-ID: >i am doing a project on lan radio.(multicast). >i need to produce rtp packets containing the audio stream. > >can i redirect the audio stream (of the mp3 file being played) from >Mplayer and make rtp packets using live555 library. > > >please give me some hint or reference. Note the "testMP3Streamer" demo application, and read the FAQ. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From nshamshiva at gmail.com Mon Aug 18 11:51:52 2008 From: nshamshiva at gmail.com (Shiva Shankar N) Date: Mon, 18 Aug 2008 15:51:52 -0300 Subject: [Live-devel] Decoding MPEG4 video via RTSP Message-ID: <4f96b010808181151k7c0b9c72mea73b355cfe09387@mail.gmail.com> Hi all, I have recorded a mpeg4 stream using openRTSP.exe and i can play this only in VLC player. I am developing a streaming player so i need to decode this recorded mpeg-4 file using XVID opensource code . But it never decodes this file and exits because of video_object_type_indication number being wrong . Please can somebody tell me why VLC is able to decode this file and not XVID opensource mpeg4 code. Thanks sham -------------- next part -------------- An HTML attachment was scrubbed... URL: From Melvin_Raj at satyam.com Tue Aug 19 18:49:10 2008 From: Melvin_Raj at satyam.com (Melvin_Raj) Date: Wed, 20 Aug 2008 09:49:10 +0800 Subject: [Live-devel] Receiving streams uosing openRTSP In-Reply-To: <4f96b010808181151k7c0b9c72mea73b355cfe09387@mail.gmail.com> References: <4f96b010808181151k7c0b9c72mea73b355cfe09387@mail.gmail.com> Message-ID: Hello , I managed to compile VLC 6.0 but the exe is not running(it just blinks and closes) :( Im currently trying to use openRTSP to receive both audio and video streams....when I send a file using testmpegstreamer and receive using testmpegreceiver, im only able to receive the video stream and not the audio....This is why I decided to use openRTSP....when I run rtsp.exe -q rtsp://xxx.xxx.xxx.xxx. I get an error saying that is unable to connect to sdp : [cid:image001.png at 01C902A9.FEEE2BC0] Can I know what do I need to do to receive and store the stream as a single file(am I typing the correct thing)??? I also tried doing this: [cid:image002.png at 01C902A9.FEEE2BC0] But it only creates a file stream.m4v but its empty :( Thank You so much for your time :) God bless... ________________________________ DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 8698 bytes Desc: image001.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 8826 bytes Desc: image002.png URL: From warren at etr-usa.com Tue Aug 19 21:59:58 2008 From: warren at etr-usa.com (Warren Young) Date: Tue, 19 Aug 2008 22:59:58 -0600 Subject: [Live-devel] [PATCH] Declare RTSP server's timeout Message-ID: <48ABA4CE.1090402@etr-usa.com> live555MediaServer has a hard-coded 45-second timeout in it for detecting clients that go away without telling the server. The timer gets reset when a client sends another RTSP command, which a conforming client does periodically to convince the server that it is still alive. RFC 2326 (RTSP) says the default timeout should be 60 seconds. (Section 12.37) My patch doesn't change this, but you should consider doing so. It's line 44 in mediaServer/DynamicRTSPServer.cpp. That section of the RFC also says that if the server wants a different timeout, it should tell the client what value it uses in response to the client's SETUP command, so it will know to change how often it sends its keepalive packets. The attached patch does this. live555MediaServer passes this hard-coded timeout value to its RTSPServer instance, which stores it in a member variable called fReclamationTestSeconds. The patch adds a constructor parameter to RTSPServer::RTSPClientSession to accept a similar value, stored in a member variable called fSessionTimeout. (I chose the different name because its meaning changes somewhat in this object.) The session object adds this value to the Session: headers in response to SETUP commands from the client, per the RFC. The patch doesn't handle the case where PLAY and SETUP are combined. I decided not to mess with this for a few reasons. First, I don't have such a client. Second, it's nonstandard behavior, but I presume at least one client that does this has been tested against the server, and it works now, so I guess it doesn't need to be told the timeout. -------------- next part -------------- A non-text attachment was scrubbed... Name: live555-rtsp-server-session-timeout.patch Type: text/x-patch Size: 4705 bytes Desc: not available URL: From finlayson at live555.com Wed Aug 20 00:14:48 2008 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 20 Aug 2008 00:14:48 -0700 Subject: [Live-devel] [PATCH] Declare RTSP server's timeout In-Reply-To: <48ABA4CE.1090402@etr-usa.com> References: <48ABA4CE.1090402@etr-usa.com> Message-ID: Our server impelmentation uses both RTSP commands *and* RTCP ("RR") packets from clients to indicate liveness. Therefore, the "timeout" parameter in a "Session:" header is not needed, because periodic RTSP commands (e.g., "GET_PARAMETER") from the client are not needed in order to tell the server that the session is alive. Instead, the server gets this information from RTCP reports. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Wed Aug 20 00:37:53 2008 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 20 Aug 2008 00:37:53 -0700 Subject: [Live-devel] Receiving streams uosing openRTSP In-Reply-To: References: <4f96b010808181151k7c0b9c72mea73b355cfe09387@mail.gmail.com> Message-ID: >I managed to compile VLC 6.0 but the exe is not >running(it just blinks and closes) If your primary goal is just to run VLC, then you could download one of the pre-built binary versions (which includes RTSP client support, using the "LIVE555 Streaming Media" libraries). > L Im currently trying to use openRTSP to >receive both audio and video streams?.when I >send a file using testmpegstreamer and receive >using testmpegreceiver, im only able to receive >the video stream and not the audio That's because "testMPEG1or2VideoReceiver" (sic) receives the video stream only. If you also want to receive the audio stream, then run "testMP3Receiver". (Didn't I already answer this a few days ago??) >?.This is why I decided to use openRTSP?.when I >run rtsp.exe -q rtsp://xxx.xxx.xxx.xxx. I get an >error saying that is unable to connect to sdp : > > The "Unknown error" message suggests that you may have built the application incorrectly, perhaps using the wrong version of 'winsock'. I'm not sure though... -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001 8.png Type: image/png Size: 8698 bytes Desc: not available URL: From peeyushduttamishra at gmail.com Wed Aug 20 07:02:24 2008 From: peeyushduttamishra at gmail.com (Peeyush Mishra) Date: Wed, 20 Aug 2008 19:32:24 +0530 Subject: [Live-devel] live-devel Digest, Vol 58, Issue 16 In-Reply-To: References: Message-ID: 2008/8/20 > Send live-devel mailing list submissions to > live-devel at lists.live555.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.live555.com/mailman/listinfo/live-devel > or, via email, send a message with subject or body 'help' to > live-devel-request at lists.live555.com > > You can reach the person managing the list at > live-devel-owner at lists.live555.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of live-devel digest..." > > > Today's Topics: > > 1. Re: redirecting audio stream to form rtp frames (Ross Finlayson) > 2. Decoding MPEG4 video via RTSP (Shiva Shankar N) > 3. Receiving streams uosing openRTSP (Melvin_Raj) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 17 Aug 2008 14:04:52 -0700 > From: Ross Finlayson > Subject: Re: [Live-devel] redirecting audio stream to form rtp frames > To: LIVE555 Streaming Media - development & use > > Message-ID: > Content-Type: text/plain; charset="us-ascii" ; format="flowed" > > >i am doing a project on lan radio.(multicast). > >i need to produce rtp packets containing the audio stream. > > > >can i redirect the audio stream (of the mp3 file being played) from > >Mplayer and make rtp packets using live555 library. > > > > > >please give me some hint or reference. > > Note the "testMP3Streamer" demo application, and read the FAQ. > -- > > Ross Finlayson > Live Networks, Inc. > http://www.live555.com/ > > > ------------------------------ > > Message: 2 > Date: Mon, 18 Aug 2008 15:51:52 -0300 > From: "Shiva Shankar N" > Subject: [Live-devel] Decoding MPEG4 video via RTSP > To: live-devel at ns.live555.com > Message-ID: > <4f96b010808181151k7c0b9c72mea73b355cfe09387 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi all, > > I have recorded a mpeg4 stream using openRTSP.exe and i can play this only > in VLC player. I am developing a streaming player so i need to decode this > recorded mpeg-4 file using XVID opensource code . But it never decodes this > file and exits because of video_object_type_indication number being wrong . > Please can somebody tell me why VLC is able to decode this file and not > XVID > opensource mpeg4 code. > > Thanks > sham Hi sham . U want to make streaming client (As I understood), For that first you make sure which MPEG4 (ES or AVI means Divx ) use are trying to decode , because both have some diff marker and places so parsing will be differnt, If you VLC is working in your case then you can use FFMPEG APIs (VLC uses FFMPEG) MPEG 4 decoding because I think XVID (means xvidcore lib) gives MPEG4 ES.....so may be marker parsing cause a problem !!!! Thanks Peeyush Mishra > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.live555.com/pipermail/live-devel/attachments/20080818/27ea08fc/attachment-0001.html > > > > ------------------------------ > > Message: 3 > Date: Wed, 20 Aug 2008 09:49:10 +0800 > From: Melvin_Raj > Subject: [Live-devel] Receiving streams uosing openRTSP > To: LIVE555 Streaming Media - development & use > > Message-ID: > < > EF1B36538C5BD043886BD58DDCB7D0420761C14851 at KLCMBX001.corp.satyam.ad> > Content-Type: text/plain; charset="us-ascii" > > Hello , > > I managed to compile VLC 6.0 but the exe is not running(it just blinks and > closes) :( Im currently trying to use openRTSP to receive both audio and > video streams....when I send a file using testmpegstreamer and receive using > testmpegreceiver, im only able to receive the video stream and not the > audio....This is why I decided to use openRTSP....when I run rtsp.exe -q > rtsp://xxx.xxx.xxx.xxx. I get an error saying that is unable to connect to > sdp : > > [cid:image001.png at 01C902A9.FEEE2BC0] > > Can I know what do I need to do to receive and store the stream as a single > file(am I typing the correct thing)??? I also tried doing this: > [cid:image002.png at 01C902A9.FEEE2BC0] > > But it only creates a file stream.m4v but its empty :( > > Thank You so much for your time :) God bless... > > > ________________________________ > DISCLAIMER: > This email (including any attachments) is intended for the sole use of the > intended recipient/s and may contain material that is CONFIDENTIAL AND > PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or > distribution or forwarding of any or all of the contents in this message is > STRICTLY PROHIBITED. If you are not the intended recipient, please contact > the sender by email and delete all copies; your cooperation in this regard > is appreciated. > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.live555.com/pipermail/live-devel/attachments/20080820/c92d017f/attachment.html > > > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: image001.png > Type: image/png > Size: 8698 bytes > Desc: image001.png > URL: < > http://lists.live555.com/pipermail/live-devel/attachments/20080820/c92d017f/attachment.png > > > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: image002.png > Type: image/png > Size: 8826 bytes > Desc: image002.png > URL: < > http://lists.live555.com/pipermail/live-devel/attachments/20080820/c92d017f/attachment-0001.png > > > > ------------------------------ > > _______________________________________________ > live-devel mailing list > live-devel at lists.live555.com > http://lists.live555.com/mailman/listinfo/live-devel > > > End of live-devel Digest, Vol 58, Issue 16 > ****************************************** > -- Thanks Peeyush Mishra -------------- next part -------------- An HTML attachment was scrubbed... URL: From ken.seo at gmail.com Wed Aug 20 09:40:33 2008 From: ken.seo at gmail.com (Ken Seo) Date: Wed, 20 Aug 2008 12:40:33 -0400 Subject: [Live-devel] NAT and RTCP Message-ID: <5e008ff40808200940n2912d2w99fe6b0f6eca893d@mail.gmail.com> Hi, I'm just curious, if anyone addressed the case I'm having now. I have an RTSP server (Live555) running in my testing lab and recently put a router in front of it. Then I configured port-forwarding options in the router so that RTSP clients can access the RTSP server through the router's public IP address. e.g. router: 74.210.123.123 RTSP server: 192.168.0.100 (Live555) client: 74.210.123.122 (OpenRTSP, VLC, QuickTime) Now the client can make an RTSP request just fine and start playing the stream. However, the RTCP packets from the client to the server get lost and as a result, the server stops the streaming with liveness timeout after 45 seconds. I've captured RTCP packets and found out that the packet's destination is set to "192.168.0.100" which is the server's real IPAddress and also is a private address. My best guess will be that the client gets the address from the RTCP packets sent from the server, then use it as destination. Any idea, comment is greatly appreciated, Regards, Ken Seo -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Wed Aug 20 11:25:56 2008 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 20 Aug 2008 11:25:56 -0700 Subject: [Live-devel] NAT and RTCP In-Reply-To: <5e008ff40808200940n2912d2w99fe6b0f6eca893d@mail.gmail.com> References: <5e008ff40808200940n2912d2w99fe6b0f6eca893d@mail.gmail.com> Message-ID: >Now the client can make an RTSP request just fine and start playing >the stream. However, the RTCP packets from the client to the server >get lost and as a result, the server stops the streaming with >liveness timeout after 45 seconds. I've captured RTCP packets and >found out that the packet's destination is set to >"192.168.0.100" which is the server's real >IPAddress and also is a private address. > >My best guess will be that the client gets the address from the RTCP >packets sent from the server No, the client gets the (RTCP destination) address from the "source=" address that the server puts in its RTSP "SETUP" response. The bottom line is that RTSP and NAT currently do not work well, especially if the RTSP server is behind a NAT. Don't do that :-) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From ken.seo at gmail.com Wed Aug 20 13:18:19 2008 From: ken.seo at gmail.com (Ken Seo) Date: Wed, 20 Aug 2008 16:18:19 -0400 Subject: [Live-devel] NAT and RTCP In-Reply-To: References: <5e008ff40808200940n2912d2w99fe6b0f6eca893d@mail.gmail.com> Message-ID: <5e008ff40808201318n16707570t428237d829a86cb7@mail.gmail.com> Thanks Ross for your quick reply! Regards, Ken 2008/8/20 Ross Finlayson > Now the client can make an RTSP request just fine and start playing the > stream. However, the RTCP packets from the client to the server get lost and > as a result, the server stops the streaming with liveness timeout after 45 > seconds. I've captured RTCP packets and found out that the packet's > destination is set to "192.168.0.100" which is the server's real IPAddress > and also is a private address. > > My best guess will be that the client gets the address from the RTCP > packets sent from the server > > > No, the client gets the (RTCP destination) address from the "source=" > address that the server puts in its RTSP "SETUP" response. > > The bottom line is that RTSP and NAT currently do not work well, especially > if the RTSP server is behind a NAT. Don't do that :-) > > -- > > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jmzorko at mac.com Wed Aug 20 18:24:56 2008 From: jmzorko at mac.com (John Zorko) Date: Wed, 20 Aug 2008 18:24:56 -0700 Subject: [Live-devel] trying to capture an .mp4 with openRTSP Message-ID: <2D104D28-DD16-4EDF-BD6E-8A973C812E7F@mac.com> Hello, all ... I'm trying to capture an MPEG4 RTP stream (sent by QTSS via QuickTime Broadcaster) to a file using openRTSP. The problem i'm seeing is that, when I try to open this file, the system (Mac OSX 10.5.4) tells me that it's not a movie file. My openRTSP command looks like this: pugsleypoo:testProgs jmzorko$ ./openRTSP -4 -w 320 -h 240 -f 30 rtsp://10.0.1.199:554/pugscam.sdp > pugscam.mp4 ... and the QuickTime Broadcaster instance is indeed set to those parameters (width 320, height 240, 30 fps). I've read the FAQ, but still can't get the generated file to open. The SDP description returned by openRTSP follows: Opened URL "rtsp://10.0.1.199:554/pugscam.sdp", returning a SDP description: v=0 o=- 438 2625243995 IN IP4 127.0.0.0 s=QuickTime c=IN IP4 0.0.0.0 t=0 0 a=range:npt=now- a=control:* m=audio 0 RTP/AVP 96 b=AS:16 a=3GPP-Adaptation-Support:1 a=rtpmap:96 mpeg4-generic/11025/1 a=fmtp:96 profile-level-id=15;mode=AAC- hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1508 a=mpeg4-esid:101 a=x-bufferdelay:5.000000 a=control:trackID=1 m=video 0 RTP/AVP 97 b=AS:220 a=3GPP-Adaptation-Support:1 a=rtpmap:97 MP4V-ES/90000 a=fmtp:97 profile-level- id = 1 ;config=000001B0F3000001B50EE040C0CF0000010000000120008440FA285020F0A31F a=mpeg4-esid:201 a=cliprect:0,0,240,320 a=framesize:97 320-240 a=x-bufferdelay:5.000000 a=control:trackID=2 ... what am I doing wrong? Regards, John Falling You - exploring the beauty of voice and sound http://www.fallingyou.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Wed Aug 20 18:42:40 2008 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 20 Aug 2008 18:42:40 -0700 Subject: [Live-devel] trying to capture an .mp4 with openRTSP In-Reply-To: <2D104D28-DD16-4EDF-BD6E-8A973C812E7F@mac.com> References: <2D104D28-DD16-4EDF-BD6E-8A973C812E7F@mac.com> Message-ID: >I'm trying to capture an MPEG4 RTP stream (sent by QTSS via >QuickTime Broadcaster) to a file using openRTSP. The problem i'm >seeing is that, when I try to open this file, the system (Mac OSX >10.5.4) tells me that it's not a movie file. My openRTSP command >looks like this: > >pugsleypoo:testProgs jmzorko$ ./openRTSP -4 -w 320 -h 240 -f 30 >rtsp://10.0.1.199:554/pugscam.sdp > >pugscam.mp4 A couple of quick questions: 1/ Are you able to play the stream rtsp://10.0.1.199:554/pugscam.sdp in a media player (QuickTime Player or VLC)? 2/ Are you making sure that "openRTSP" terminates properly? You can't just -C it, otherwise the resulting output file will be incomplete. You have to terminate it with "kill -HUP", or else specify a specific duration in advance, using the "-d " option. If both of these are true, then I don't know what the problem might be; you should ask a QuickTime mailing list why the file won't play. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From jmzorko at mac.com Wed Aug 20 19:07:31 2008 From: jmzorko at mac.com (John Zorko) Date: Wed, 20 Aug 2008 19:07:31 -0700 Subject: [Live-devel] trying to capture an .mp4 with openRTSP In-Reply-To: References: <2D104D28-DD16-4EDF-BD6E-8A973C812E7F@mac.com> Message-ID: <090FD624-F7F8-49BC-914E-553574216E5A@mac.com> Ross, >> I'm trying to capture an MPEG4 RTP stream (sent by QTSS via >> QuickTime Broadcaster) to a file using openRTSP. The problem i'm >> seeing is that, when I try to open this file, the system (Mac OSX >> 10.5.4) tells me that it's not a movie file. My openRTSP command >> looks like this: >> >> pugsleypoo:testProgs jmzorko$ ./openRTSP -4 -w 320 -h 240 -f 30 rtsp://10.0.1.199:554/pugscam.sdp >> > pugscam.mp4 > > A couple of quick questions: > 1/ Are you able to play the stream rtsp://10.0.1.199:554/pugscam.sdp > in a media player (QuickTime Player or VLC)? > 2/ Are you making sure that "openRTSP" terminates properly? You > can't just -C it, otherwise the resulting output file will > be incomplete. You have to terminate it with "kill -HUP", or else > specify a specific duration in advance, using the "-d " > option. > > If both of these are true, then I don't know what the problem might > be; you should ask a QuickTime mailing list why the file won't play. Thank you very much -- while I was able to play the QTSS stream with the QuickTime Player, the problem with the saved .mp4 file was just as you said (ctrl-c vs. sighup). This leads me to another question, though: I would like to play the pugscam.mp4 file as openRTSP is writing to it. I know this sounds crazy, but i'm experimenting with an idea I have, and the first test of said idea would be to see if I can play an .mp4 file from a server as the server is generating it, and I was thinking of basing my server on the openRTSP code. This question may be off-topic now, but do you know what might cause an application to fail to open an .mp4 file as openRTSP is generating it i.e. is there some bit of info that is written to the header of an .mp4 file that is only known after the file is finished being created (though this would effectively mean you need to seek on stdout, which I didn't think was possible)? Regards, John Falling You - exploring the beauty of voice and sound http://www.fallingyou.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From jmzorko at mac.com Wed Aug 20 19:20:39 2008 From: jmzorko at mac.com (John Zorko) Date: Wed, 20 Aug 2008 19:20:39 -0700 Subject: [Live-devel] trying to capture an .mp4 with openRTSP In-Reply-To: References: <2D104D28-DD16-4EDF-BD6E-8A973C812E7F@mac.com> Message-ID: Ross, I think what I need to do is examine what shutdown() does in playCommon.cpp, so I can see if I can fake out a client into thinking that it can play the file. I know it sounds nutso ... Regards, John Falling You - exploring the beauty of voice and sound http://www.fallingyou.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Wed Aug 20 19:29:30 2008 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 20 Aug 2008 19:29:30 -0700 Subject: [Live-devel] trying to capture an .mp4 with openRTSP In-Reply-To: <090FD624-F7F8-49BC-914E-553574216E5A@mac.com> References: <2D104D28-DD16-4EDF-BD6E-8A973C812E7F@mac.com> <090FD624-F7F8-49BC-914E-553574216E5A@mac.com> Message-ID: >Thank you very much -- while I was able to play the QTSS stream with >the QuickTime Player, the problem with the saved .mp4 file was just >as you said (ctrl-c vs. sighup). This leads me to another question, >though: > >I would like to play the pugscam.mp4 file as openRTSP is writing to it. Because of the details of the '.mp4' (or '.mov') file format, I don't think this is possible. These files require certain meta-data ('atoms') be included in the file, along with the media data itself. You can't play the file without this meta-data, and because the meta-data has to be in a well-known place in the file, you can't constantly write the meta-data to the file while the media data is also being written. What you can do, however (though not with our software, as is) is play the *stream* at the same time that you're writing (a copy of) the media data to a file. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From amit.yedidia at elbitsystems.com Wed Aug 20 23:16:29 2008 From: amit.yedidia at elbitsystems.com (amit.yedidia at elbitsystems.com) Date: Thu, 21 Aug 2008 09:16:29 +0300 Subject: [Live-devel] ANNOUNCE method on Server side Message-ID: Hi, Does Live555 support the server receiving ANNOUNCE request from the client (I found it on the client, but not on the server) If not, what would you suggest to implement a record session initialization where the client wish to publish a presentation to the server? Can I do it using SETUP request which includes sdp from the client to the server? Regards, Amit Yedidia Elbit System Ltd. Email: amit.yedidia at elbitsystems.com Tel: 972-4-8318905 ---------------------------------------------------------- The information in this e-mail transmission contains proprietary and business sensitive information. Unauthorized interception of this e-mail may constitute a violation of law. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. You are also asked to contact the sender by reply email and immediately destroy all copies of the original message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Wed Aug 20 23:28:05 2008 From: finlayson at live555.com (Ross Finlayson) Date: Wed, 20 Aug 2008 23:28:05 -0700 Subject: [Live-devel] ANNOUNCE method on Server side In-Reply-To: References: Message-ID: >Does Live555 support the server receiving ANNOUNCE request from the clien No it doesn't. "ANNOUNCE" is rarely implemented, and in fact was removed from the RTSP 2.0 specification. >If not, what would you suggest to implement a record session >initialization where the client wish to publish a presentation to >the server? I suggest not trying to use RTSP for this. Instead, have the server read the input stream from a TCP socket (perhaps - but not necessarily - using HTTP). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From rishikerala at gmail.com Thu Aug 21 01:27:37 2008 From: rishikerala at gmail.com (Rishi kerala) Date: Thu, 21 Aug 2008 17:27:37 +0900 Subject: [Live-devel] Regarding the live555 code in mplayer Message-ID: <4ba29cc0808210127p5a97b19eha3d45d67e49cdc94@mail.gmail.com> Dear Ross, I am trying to use live555 as a server and client as mplayer. Also tried with some mp2 streams . mplayer I am able to play the streams properly. when streams reaches EOF then mplayer is hanging. I just do some debug and got that the mplayer hangs at demux_rtp.cpp file. It goes to an infinte loop there as waiting until the data available ? Can you suggest some method to avoid this.. // Block ourselves until data becomes available: TRACE; TaskScheduler& scheduler = bufferQueue->readSource()->envir().taskScheduler(); scheduler.doEventLoop(&bufferQueue->blockingFlag); Thanks Rishi -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Thu Aug 21 01:40:23 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 21 Aug 2008 01:40:23 -0700 Subject: [Live-devel] Regarding the live555 code in mplayer In-Reply-To: <4ba29cc0808210127p5a97b19eha3d45d67e49cdc94@mail.gmail.com> References: <4ba29cc0808210127p5a97b19eha3d45d67e49cdc94@mail.gmail.com> Message-ID: > I am trying to use live555 as a server and client as mplayer. >Also tried with some mp2 streams . >mplayer I am able to play the streams properly. when streams reaches >EOF then mplayer is hanging. >I just do some debug and got that the mplayer hangs at demux_rtp.cpp >file. It goes to an infinte loop there as >waiting until the data available ? Can you suggest some method to avoid this.. When the stream ends, the server should be sending a RTCP "BYE" packet, which your client (MPlayer) should be receiving. That should be telling it that the stream has ended. If, however, the server is not sending a RTCP "BYE", and is not closing the RTSP TCP connection, then there's not much you can do - because it's not telling you that the stream has ended. Of course, If you really want to stop at this point, then just quit MPlayer :-) -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From warren at etr-usa.com Thu Aug 21 02:42:48 2008 From: warren at etr-usa.com (Warren Young) Date: Thu, 21 Aug 2008 03:42:48 -0600 Subject: [Live-devel] [PATCH] Declare RTSP server's timeout In-Reply-To: References: <48ABA4CE.1090402@etr-usa.com> Message-ID: <48AD3898.5030009@etr-usa.com> Ross Finlayson wrote: > Our server impelmentation uses both RTSP commands *and* RTCP ("RR") > packets from clients to indicate liveness. Therefore, the "timeout" > parameter in a "Session:" header is not needed, because periodic RTSP > commands (e.g., "GET_PARAMETER") from the client are not needed in order > to tell the server that the session is alive. Instead, the server gets > this information from RTCP reports. I have a client here that doesn't send RTCP RR packets to the server. It does send GET_PARAMETER, every ~60 seconds unless told different. Therefore, the server times out, since its hard-coded timeout is shorter. If the server tells the client that it has a shorter timeout than the RFC recommends as a default, it would send its GET_PARAMETER commands more often, so the connection wouldn't time out. I would think this patch would be a no-brainer. The spec says "do X," and the patch makes the server do X. What's the downside? From iskaz at intracomdefense.com Thu Aug 21 13:43:19 2008 From: iskaz at intracomdefense.com (=?ISO-8859-1?Q?=3F=3F=3F=3F=3F=3F=3F=3F_=3F=3F=3F=3F=3F=3F=3F?=) Date: Thu, 21 Aug 2008 23:43:19 +0300 Subject: [Live-devel] MPEG4VideoStreamParser + marker bit not set!. Message-ID: <48ADD367.9090606@intracomdefense.com> Hi all, I try to program a video streamer for the MPEG4000XLP (a 4-Channel MPEG4 Encoder/Decoder for PC104+, http://ampltd.com/prod/mpeg4kxlp.html ) with help of the live media SDK. For the program base structure I have used the test program "testMPEG4VideoStreamer" of live555 testFiles. The core change that I made is that I set in function void play() videoSource = MPEG4VideoStreamFramer::createNew(*env, _inputSource_); where _inputSource_ defined as _DeviceSourceMPG4K* inputSource_; where _DeviceSourceMPG4K_ a class based on the _DeviceSource_ file which was suggested according to the FAQ: "Alternatively, if your encoder presents you with a sequence of frames, rather than a sequence of bytes, then a more efficient solution would be to write your own "FramedSource " subclass that encapsulates your encoder, and delivers audio or video frames directly to the appropriate "*RTPSink" object. This avoids the need for an intermediate 'framer' filter that parses the input byte stream. (If, however, you are streaming MPEG-4 or MPEG-2 video with "B" frames, then you should include the appropriate "*/Discrete/Framer" filter, in order to generate correct presentation times.) For a model of how to do that, see "liveMedia/DeviceSource.cpp " (and "liveMedia/include/DeviceSource.hh "). You will need to fill in parts of this code to do the actual read from your encoder. " It works, but when I run the program I get the following continuous message : MPEG4VideoStreamParser::parseVideoObjectPlane(): Saw unexpected code 0x1b2 MPEG4VideoStreamParser::parseVideoObjectPlane(): marker bit not set! MPEG4VideoStreamParser::parseVideoObjectPlane(): Saw unexpected code 0x1b2 MPEG4VideoStreamParser::parseVideoObjectPlane(): marker bit not set!. .MPEG4VideoStreamParser::parseVideoObjectPlane(): Saw unexpected code 0x1b2 MPEG4VideoStreamParser::parseVideoObjectPlane(): marker bit not set! .... And by the mplayer I see an almost clear Video with some breaks in its flow. What I am doing wrong? How can I set the marker bit ??? Thank you all in advanced, Ioannis Skazikis. Also the mplayer outputs: MPlayer 1.0-1.rc2.10mdv2008.1-4.2.2 (C) 2000-2007 MPlayer Team CPU: Intel(R) Pentium(R) 4 CPU 2.80GHz (Family: 15, Model: 4, Stepping: 1) CPUflags: MMX: 1 MMX2: 1 3DNow: 0 3DNow2: 0 SSE: 1 SSE2: 1 Compiled with runtime CPU detection. mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing rtsp://172.18.5.175:8554/testStream. Resolving 172.18.5.175 for AF_INET6... Couldn't resolve name for AF_INET6: 172.18.5.175 Connecting to server 172.18.5.175[172.18.5.175]: 8554... rtsp_session: unsupported RTSP server. Server type is 'unknown'. STREAM_LIVE555, URL: rtsp://172.18.5.175:8554/testStream Stream not seekable! file format detected. Initiated "video/MP4V-ES" RTP subsession on port 18888 VIDEO: [mp4v] 0x0 0bpp 0.000 fps 0.0 kbps ( 0.0 kbyte/s) xscreensaver_disable: Could not find XScreenSaver window. No value set for `/apps/gnome-screensaver/idle_activation_enabled' ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) ========================================================================== Audio: no sound FPS forced to be 100.000 (ftime: 0.010). Starting playback... Marker bit missing before time_increment VDec: vo config request - 720 x 576 (preferred colorspace: Planar YV12) VDec: using Planar YV12 as output csp (no 0) Movie-Aspect is 1.25:1 - prescaling to correct movie aspect. VO: [xv] 720x576 => 720x576 Planar YV12 [zoom] [mpeg4 @ 0x8898c90]warning: first frame is no keyframe Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment [mpeg4 @ 0x8898c90]ac-tex damaged at 20 8 [mpeg4 @ 0x8898c90]Error at MB: 388 [mpeg4 @ 0x8898c90]concealing 1295 DC, 1295 AC, 1295 MV errors Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment Marker bit missing before time_increment [mpeg4 @ 0x8898c90]ac-tex damaged at 44 2 [mpeg4 @ 0x8898c90]Error at MB: 136 [mpeg4 @ 0x8898c90]concealing 1535 DC, 1535 AC, 1535 MV errors Marker bit missing before time_increment ..... -------------- next part -------------- An HTML attachment was scrubbed... URL: From rishikerala at gmail.com Thu Aug 21 03:02:40 2008 From: rishikerala at gmail.com (Rishi kerala) Date: Thu, 21 Aug 2008 19:02:40 +0900 Subject: [Live-devel] Regarding the live555 code in mplayer In-Reply-To: References: <4ba29cc0808210127p5a97b19eha3d45d67e49cdc94@mail.gmail.com> Message-ID: <4ba29cc0808210302w460e0dadg4e94194c9babc422@mail.gmail.com> Thank's Ross Actually we have a server listing the media file in the SDcard and send to client, then it invoke the live555mediaserver. Client application invoke the mplayer with the list of the files as playlist. So we have to play one by one. So we tired some way to forcefully kill the application. I am digging the code now... If you have any idea please let me know ... Thanks Rishi On Thu, Aug 21, 2008 at 5:40 PM, Ross Finlayson wrote: > I am trying to use live555 as a server and client as mplayer. Also >> tried with some mp2 streams . >> mplayer I am able to play the streams properly. when streams reaches EOF >> then mplayer is hanging. >> I just do some debug and got that the mplayer hangs at demux_rtp.cpp file. >> It goes to an infinte loop there as >> waiting until the data available ? Can you suggest some method to avoid >> this.. >> > > When the stream ends, the server should be sending a RTCP "BYE" packet, > which your client (MPlayer) should be receiving. That should be telling it > that the stream has ended. If, however, the server is not sending a RTCP > "BYE", and is not closing the RTSP TCP connection, then there's not much you > can do - because it's not telling you that the stream has ended. > > Of course, If you really want to stop at this point, then just quit MPlayer > :-) > -- > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Thu Aug 21 06:23:50 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 21 Aug 2008 06:23:50 -0700 Subject: [Live-devel] [PATCH] Declare RTSP server's timeout In-Reply-To: <48AD3898.5030009@etr-usa.com> References: <48ABA4CE.1090402@etr-usa.com> <48AD3898.5030009@etr-usa.com> Message-ID: >Ross Finlayson wrote: >>Our server impelmentation uses both RTSP commands *and* RTCP ("RR") >>packets from clients to indicate liveness. Therefore, the >>"timeout" parameter in a "Session:" header is not needed, because >>periodic RTSP commands (e.g., "GET_PARAMETER") from the client are >>not needed in order to tell the server that the session is alive. >>Instead, the server gets this information from RTCP reports. > >I have a client here that doesn't send RTCP RR packets to the server. Your client is broken, because RTCP is a required part of the RTP/RTCP standard. It is not optional functionality. > It does send GET_PARAMETER, every ~60 seconds unless told >different. Therefore, the server times out, since its hard-coded >timeout is shorter. Feel free to change the "reclamationTestSeconds" value (in "DynamicRTSPServer.cpp") to 60. > If the server tells the client that it has a shorter timeout than >the RFC recommends as a default, it would send its GET_PARAMETER >commands more often, so the connection wouldn't time out. > >I would think this patch would be a no-brainer. The spec says "do >X," and the patch makes the server do X. What's the downside? The downside would be that the server would be telling the client - incorrectly - that it needs to send RTSP commands at least every N seconds. That would be incorrect, because standard RTCP packets ("RR") could also be used to indicate client liveness. I might end up changing the default "reclamationTestSeconds" value from 45 to 60 (which would also solve your problem), but I won't be adding your patch to add the "timeout=" parameter, because that would be misleading. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Thu Aug 21 06:29:18 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 21 Aug 2008 06:29:18 -0700 Subject: [Live-devel] Regarding the live555 code in mplayer In-Reply-To: <4ba29cc0808210302w460e0dadg4e94194c9babc422@mail.gmail.com> References: <4ba29cc0808210127p5a97b19eha3d45d67e49cdc94@mail.gmail.com> <4ba29cc0808210302w460e0dadg4e94194c9babc422@mail.gmail.com> Message-ID: You could also try using VLC as a client, instead of MPlayer. You might find that that works better for you. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Thu Aug 21 06:33:40 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 21 Aug 2008 06:33:40 -0700 Subject: [Live-devel] MPEG4VideoStreamParser + marker bit not set!. In-Reply-To: <48ADD367.9090606@intracomdefense.com> References: <48ADD367.9090606@intracomdefense.com> Message-ID: >videoSource = MPEG4VideoStreamFramer::createNew(*env, inputSource); Because "inputSource" delivers discrete MPEG-4 video frames, rather than an unstructured byte stream, you should use "MPEG4VideoStreamDiscreteFramer" rather than "MPEG4VideoStreamFramer". -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From jstafford at ampltd.com Thu Aug 21 06:38:14 2008 From: jstafford at ampltd.com (James Stafford) Date: Thu, 21 Aug 2008 14:38:14 +0100 Subject: [Live-devel] MPEG4VideoStreamParser + marker bit not set!. In-Reply-To: <48ADD367.9090606@intracomdefense.com> References: <48ADD367.9090606@intracomdefense.com> Message-ID: <48AD6FC6.1030100@ampltd.com> Hi Ioannis, > > I try to program a video streamer for the MPEG4000XLP (a 4-Channel > MPEG4 Encoder/Decoder for PC104+, > http://ampltd.com/prod/mpeg4kxlp.html ) with help of the live media SDK. > > > It works, but when I run the program I get the following continuous > message : > > MPEG4VideoStreamParser::parseVideoObjectPlane(): Saw unexpected code > 0x1b2 > MPEG4VideoStreamParser::parseVideoObjectPlane(): marker bit not set! > MPEG4VideoStreamParser::parseVideoObjectPlane(): Saw unexpected code > 0x1b2 > MPEG4VideoStreamParser::parseVideoObjectPlane(): marker bit not set!. > .MPEG4VideoStreamParser::parseVideoObjectPlane(): Saw unexpected code > 0x1b2 > MPEG4VideoStreamParser::parseVideoObjectPlane(): marker bit not set! > .... The MPEG4000XLP adds a user data packet to the end of each MPEG4 video frame passed into the video callback chain. Players such as VLC don't tend to like this user frame and so our RTSP streaming SDK (VStream SDK) removes it before passing the data into the Live555 libraries. The user start code (0x000001b2) is followed by an unsigned long that contains an ID (0x414d5050). The next unsigned long contains the size of the user frame (including start code, ID and size). -- James Stafford Advanced Micro Peripherals Ltd Unit 1 Harrier House Sedgeway Business Park Witchford Cambridge CB6 2HY Fax:+44 1353 659 600 From jstafford at ampltd.com Thu Aug 21 07:09:18 2008 From: jstafford at ampltd.com (James Stafford) Date: Thu, 21 Aug 2008 15:09:18 +0100 Subject: [Live-devel] MPEG4VideoStreamParser + marker bit not set!. In-Reply-To: <48AD6FC6.1030100@ampltd.com> References: <48ADD367.9090606@intracomdefense.com> <48AD6FC6.1030100@ampltd.com> Message-ID: <48AD770E.60007@ampltd.com> > > The user start code (0x000001b2) is followed by an unsigned long that > contains an ID (0x414d5050). The next unsigned long contains the size > of the user frame (including start code, ID and size). > Having looked at the code, the user frame size after the ID is actually an unsigned short. -- James Stafford Advanced Micro Peripherals Ltd Unit 1 Harrier House Sedgeway Business Park Witchford Cambridge CB6 2HY Fax:+44 1353 659 600 From warren at etr-usa.com Thu Aug 21 07:50:35 2008 From: warren at etr-usa.com (Warren Young) Date: Thu, 21 Aug 2008 08:50:35 -0600 Subject: [Live-devel] [PATCH] Declare RTSP server's timeout In-Reply-To: References: <48ABA4CE.1090402@etr-usa.com> <48AD3898.5030009@etr-usa.com> Message-ID: <48AD80BB.8050702@etr-usa.com> Ross Finlayson wrote: >> >> I have a client here that doesn't send RTCP RR packets to the server. > > Your client is broken, because RTCP is a required part of the RTP/RTCP > standard. It is not optional functionality. RFC 3550, section 6.0: "[RTCP] SHOULD be used in all environments, but particularly in the IP multicast environment." I'm sure I don't need to tell you that "SHOULD" is standardese for "optional". Plus, I'm doing unicast on relatively small private LANs. RTCP RR solves problems that largely don't exist in my world. Later, in section 6.2: "Turning off RTCP reception reports is NOT RECOMMENDED... However, doing so may be appropriate for systems... that don't require feedback on the quality of reception or liveness of receivers and that have other means to avoid congestion." That definitely applies to my situation. My world is K-12 schools, where managed switches are considered high technology, capable of doing anything such a small LAN needs. That's how you get QoS and congestion control (VLANs) in my world. That just leaves alternate ways to indicate liveness, which is the subject of this patch. > I might end up changing > the default "reclamationTestSeconds" value from 45 to 60 You might go slightly higher, to account for differences in interpretation of time between client and server. Not just clock drift and such, but also, how do you define time t=0? Is it the transmission time of the first RTSP request, or from the time of receiving the first RTP packet from the server, or somewhere between? There's usually a couple of seconds difference between these two times. The first keepalive could come in 62 seconds past the server's t=0. Every one thereafter will be ~60 seconds apart. > I won't be adding your patch to add the > "timeout=" parameter, because that would be misleading. What if the server decided to do this based on the User-Agent value? I can make it check for my client, and send it only when it sees that it's necessary. Understand the tradeoff: we'd be adding this code to save 0.00004 Mbit/s of network bandwidth. (~300 bytes every 60 seconds.) From tony at lava.net Thu Aug 21 11:13:03 2008 From: tony at lava.net (Antonio Querubin) Date: Thu, 21 Aug 2008 08:13:03 -1000 (HST) Subject: [Live-devel] ANNOUNCE method on Server side In-Reply-To: References: Message-ID: On Wed, 20 Aug 2008, Ross Finlayson wrote: >> Does Live555 support the server receiving ANNOUNCE request from the clien > > No it doesn't. "ANNOUNCE" is rarely implemented, and in fact was removed > from the RTSP 2.0 specification. To be more accurate it's been moved out of the 2.0 core spec (a work-in-progress) and some work is on-going to make it an extension: http://www.ietf.org/internet-drafts/draft-stiemerling-rtsp-announce-01.txt Antonio Querubin whois: AQ7-ARIN From finlayson at live555.com Thu Aug 21 11:47:10 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 21 Aug 2008 11:47:10 -0700 Subject: [Live-devel] [PATCH] Declare RTSP server's timeout In-Reply-To: <48AD80BB.8050702@etr-usa.com> References: <48ABA4CE.1090402@etr-usa.com> <48AD3898.5030009@etr-usa.com> <48AD80BB.8050702@etr-usa.com> Message-ID: >Later, in section 6.2: "Turning off RTCP reception reports is NOT >RECOMMENDED... However, doing so may be appropriate for systems... >that don't require feedback on the quality of reception or liveness >of receivers and that have other means to avoid congestion." > >That definitely applies to my situation. My world is K-12 schools, >where managed switches are considered high technology, capable of >doing anything such a small LAN needs. That's how you get QoS and >congestion control (VLANs) in my world. I don't understand/believe this. Most people, when they try to explain why they don't implement RTCP, give all sorts of excuses, but deep down, the reason they don't implement RTCP is because they think it's too 'complicated'. But if you're using our software to develop your client, then there's no complexity - you already get a RTCP implementation that you enable with just one line of code. Anyway, your client already needs to *receive* RTCP "SR" packets, in order to do A/V sync, unless you are sending audio-only, video-only, or a Transport Stream only. Also, if your client doesn't send RTCP "RR" packets, then you will miss out on enhanced server features - such as QOS statistics - that we may implement in the future. >>I might end up changing the default "reclamationTestSeconds" value >>from 45 to 60 > >You might go slightly higher, to account for differences in >interpretation of time between client and server. OK, I might make it 65 seconds, by default. But there are no current plans to have our server send the "timeout=" parameter, for the reasons I've given. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From warren at etr-usa.com Fri Aug 22 06:16:28 2008 From: warren at etr-usa.com (Warren Young) Date: Fri, 22 Aug 2008 07:16:28 -0600 Subject: [Live-devel] [PATCH] Declare RTSP server's timeout In-Reply-To: References: <48ABA4CE.1090402@etr-usa.com> <48AD3898.5030009@etr-usa.com> <48AD80BB.8050702@etr-usa.com> Message-ID: <48AEBC2C.9090905@etr-usa.com> Ross Finlayson wrote: > > if you're using our software to develop your client, I can see how you might have gotten the impression that I was writing the client myself. When I said I had a client here, I meant it was sitting on the table over there to my right. :) It's a set-top box, much like an Amino, but far more powerful. I'm using your server to feed streams to it, for testing. I'm being circumspect about the brand name and model because I don't imagine you have one, and it's not necessary that you go get one. This isn't about the quirks of Box X, because Box X is obeying the letter of the RFC, if perhaps not the spirit. I've already asked the makers of Box X to make it send RTCP RR, but in the meantime, I have an fix. They know this, and that may prevent them from putting resources into this soon, or ever. I can maintain a private branch of the software with this fix in it, but of course that means an integration step every time you release a new version that has something I want. Besides, there may be other clients out there that would benefit from this patch. The coming IPTV revolution is going to result in a whole lot of random STBs you've never heard of from companies you've never heard of coming on the market. The nature of things is that they'll have all sorts of odd behaviors, so even if this is the only box today that does this, it's probably not going to be the last. > your client already needs to *receive* RTCP "SR" packets, I presume it does, since I don't see ICMP port unreachable coming back whenever I see an SR go by. I guess it could just be eating them. I'm sending MPEG-2 transport streams, so I don't see that SR is necessary for A/V sync. The sync information is in the transport stream headers. From finlayson at live555.com Fri Aug 22 07:55:51 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 22 Aug 2008 07:55:51 -0700 Subject: [Live-devel] [PATCH] Declare RTSP server's timeout In-Reply-To: <48AEBC2C.9090905@etr-usa.com> References: <48ABA4CE.1090402@etr-usa.com> <48AD3898.5030009@etr-usa.com> <48AD80BB.8050702@etr-usa.com> <48AEBC2C.9090905@etr-usa.com> Message-ID: >Besides, there may be other clients out there that would benefit >from this patch. For the last time: I won't be adding a patch to have the server set the "timeout=" parameter, because this would - incorrectly - be telling cliente that they need to send periodic RTSP commands to indicate 'liveness'. What I *will* do is change the default server "reclamationTestSeconds" value to 65 seconds. That should solve the problem you're having with your client. > The coming IPTV revolution is going to result in a whole lot of >random STBs you've never heard of from companies you've never heard >of coming on the market. Any of these STB manufacturers are welcome to contact me directly (or use this mailing list) if they require assistance making their products work with our software (or make them standards compliant in general). But I'm not going to change the server software just to help some random company I've "never heard of" that happens to be too lazy (or clueless) to follow protocol standards. I'm getting really tired of the (apparently common) attitude that the software that's embedded in a special-purpose physical device (no matter how small, obscure and transient its manufacturer) is somehow 'sacrosanct', and that it should always be *our* software that has to adapt to work with them, even if they're flouting protocol standards. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From marcel at marcelgagne.com Sat Aug 23 06:51:04 2008 From: marcel at marcelgagne.com (Marcel Gagne) Date: Sat, 23 Aug 2008 09:51:04 -0400 Subject: [Live-devel] Audio format of captured file Message-ID: <200808230951.05660.marcel@marcelgagne.com> Hello everyone, I am now happily capturing twin treams from my Panasonic BL-C131A network camera using RTSP. The resulting files are audio-G726-32-1 and video-MP4V-ES-2 and both look to be about the size I'd expect for the time captured. In fact, I can use ffmpeg to convert to video file to whatever I want and it plays beautifully. The audio, on the other hand, is a problem. I'm guessing that it's a g726 encoded file, but every attempt I have made to convert it to something else (MP3, OGG, wav, etc) gets me nowhere. audio-G726-32-1: could not find codec parameters Now, I have tried telling ffmpeg that it's a g726 encoded file, but it still fails. Is there something special about this file? Should I be using a particular ffmpeg codec? Which one? The whole point of this exercise is to take the captured video and audio streams, convert them to a single flash file (or whatever), and make the video available to other systems on my local network. Any help or guidance would be appreciated. -- Marcel (Writer and Free Thinker at Large) Gagn? Note: This massagee wos nat speel or gramer-checkered. Main Websites: www.marcelgagne.com AND www.cookingwithlinux.com Author of the "Moving to Linux" series of books Follow me : http://twitter.com/wftl Join the WFTL-LUG : http://www.wftl-lug.org -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. From mamille1 at rockwellcollins.com Sat Aug 23 14:22:20 2008 From: mamille1 at rockwellcollins.com (mamille1 at rockwellcollins.com) Date: Sat, 23 Aug 2008 16:22:20 -0500 Subject: [Live-devel] RockwellCollins playback problem Message-ID: Ross and all Thanks for the earlier advice. We've been able to make improvements. It turns out that our OS was misconfigured, causing it run 2% faster than we should. We've taken these steps: - corrected OS timing rate - increased socket send and receive buffer sizes on our end, following the advice in the FAQ We have the option in our Linux-based system to adjust the system time tick to 4 ms or 1 ms. Within the liveMedia library, the resolution of time is in microseconds. Is the resolution of our system's time -- 1,000 or 4,000 microseconds -- small enough for decent playback? Thanks! -=- Mike Miller SW Engineer, Govt. Systems Rockwell Collins, Inc. Cedar Rapids, IA -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Sat Aug 23 15:08:41 2008 From: finlayson at live555.com (Ross Finlayson) Date: Sat, 23 Aug 2008 15:08:41 -0700 Subject: [Live-devel] Audio format of captured file In-Reply-To: <200808230951.05660.marcel@marcelgagne.com> References: <200808230951.05660.marcel@marcelgagne.com> Message-ID: > audio-G726-32-1: could not find codec parameters > >Now, I have tried telling ffmpeg that it's a g726 encoded file, but it still >fails. Is there something special about this file Not really. The audio stream uses the RTP payload format "audio/G726-32", which is G.726 audio encoded at a sampling frequency of 8000 Hz, with 4 bits-per-sample. Note that these 4-bit samples are packed into bytes in 'little endian' order. See RFC 3551, section 4.5.4. The resulting file contains (exactly) the contents of the RTP stream, and so is in the same format. >? Should I be using a >particular ffmpeg codec? Which one? I'm sorry, I can't help you here. I'm not an expert on "ffmpeg" (or any other decoder software). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Sat Aug 23 15:46:11 2008 From: finlayson at live555.com (Ross Finlayson) Date: Sat, 23 Aug 2008 15:46:11 -0700 Subject: [Live-devel] RockwellCollins playback problem In-Reply-To: References: Message-ID: >We have the option in our Linux-based system to adjust the system >time tick to 4 ms or 1 ms. Within the liveMedia library, the >resolution of time is in microseconds. Is the resolution of our >system's time -- 1,000 or 4,000 microseconds -- small enough for >decent playback? Yes, that should be OK. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From soumya.patra at lge.com Mon Aug 25 23:52:22 2008 From: soumya.patra at lge.com (soumya patra) Date: Tue, 26 Aug 2008 12:22:22 +0530 Subject: [Live-devel] RTSP Authentication Message-ID: <20080826065222.3226B55800E@LGEMRELSE7Q.lge.com> Hi Ross, We have Streaming server which streams H264 perfectly (both unicast & multicast). We have created a dynamic RTSP server for streaming with different media codecs. Now we want to use digest authentication in our RTSP Server for some specific Client. Can you explain how to use Authenticator class. Waiting for your response. Regards Soumya -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Tue Aug 26 00:28:40 2008 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Aug 2008 00:28:40 -0700 Subject: [Live-devel] RTSP Authentication In-Reply-To: <20080826065222.3226B55800E@LGEMRELSE7Q.lge.com> References: <20080826065222.3226B55800E@LGEMRELSE7Q.lge.com> Message-ID: >Hi Ross, > We have Streaming server which streams H264 perfectly (both >unicast & multicast). > We have created a dynamic RTSP server for streaming with >different media codecs. > Now we want to use digest authentication in our RTSP Server for >some specific Client. >Can you explain how to use Authenticator class. Look at the code (in "live555MediaServer.cpp") that's bracketed with #ifdef ACCESS_CONTROL #endif Enable this code, and read the comments in it. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From amit.yedidia at elbitsystems.com Tue Aug 26 03:10:03 2008 From: amit.yedidia at elbitsystems.com (amit.yedidia at elbitsystems.com) Date: Tue, 26 Aug 2008 13:10:03 +0300 Subject: [Live-devel] RFC2326bis Message-ID: Does LIVE555 comply (now or in the future) to RFC 2326bis? Regards, Amit Yedidia Elbit System Ltd. Email: amit.yedidia at elbitsystems.com Tel: 972-4-8318905 ---------------------------------------------------------- The information in this e-mail transmission contains proprietary and business sensitive information. Unauthorized interception of this e-mail may constitute a violation of law. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. You are also asked to contact the sender by reply email and immediately destroy all copies of the original message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Tue Aug 26 04:37:05 2008 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Aug 2008 04:37:05 -0700 Subject: [Live-devel] RFC2326bis In-Reply-To: References: Message-ID: >Does LIVE555 comply (now or in the future) to RFC 2326bis? Yes, sometime in the future, after the 'RFC 2326bis' Internet Draft (RTSP 2.0) becomes a standard RFC. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From Jerry.Johns at nuvation.com Tue Aug 26 09:07:10 2008 From: Jerry.Johns at nuvation.com (Jerry Johns) Date: Tue, 26 Aug 2008 09:07:10 -0700 Subject: [Live-devel] NAT and RTCP Message-ID: <274F7B50569A1B4C9D7BCAB17A9C7BE101111C01@mailguy3.skynet.nuvation.com> Hey Ross, If I would like to go ahead and use a NAT with UDP streaming mode, how would I facilitate this? I have a similar behaviour where after 45 seconds, I get a liveness timeout due to the inability of RTCP packets (i.e receiver reports) to get to the streaming server from the client. Can I statically forward the RTCP ports ahead of time in the router? My hunch tells me the RTCP ports are dynamically set by the stack and so cannot be forwarded easily - the streaming over TCP option works well in VLC, and no timeouts do occur (but that involves using only VLC). Quicktime unfortunately does not support transmission over TCP (they do RTSP over HTTP), and it's absolutely imperative for our client that the stream be present in Quicktime. Are there any solutions you can recommend for streaming over UDP over the internet, unicast? I saw an RFC spec where you can do "a=rtcp:53020 IN IP4 126.16.64.4" to mitigate this dynamicness - (RFC 3605). A simple grep in the liveMedia source code shows that we don't support this currently Thank you, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Tue Aug 26 17:04:03 2008 From: finlayson at live555.com (Ross Finlayson) Date: Tue, 26 Aug 2008 17:04:03 -0700 Subject: [Live-devel] NAT and RTCP In-Reply-To: <274F7B50569A1B4C9D7BCAB17A9C7BE101111C01@mailguy3.skynet.nuvation.com> References: <274F7B50569A1B4C9D7BCAB17A9C7BE101111C01@mailguy3.skynet.nuvation.com> Message-ID: >My hunch tells me the RTCP ports are dynamically set by the stack Yes, however, the RTP and RTCP ports start at 6970 by default (see "liveMedia/include/OnDemandServerMediaSubsession.hh"). (I chose that number because other servers - e.g., Apple's - use the same number.) RTP ports are always even (so, 6970, 6972, 6974, ...), and RTCP ports are always odd (so, 6971, 6973, 6975, ...). If you have a limit on how many concurrent streams you may be serving at once, you may be able to arrange port forwarding just for a small rance of ports, starting at 6970. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From ken.seo at gmail.com Thu Aug 28 12:08:58 2008 From: ken.seo at gmail.com (Ken Seo) Date: Thu, 28 Aug 2008 15:08:58 -0400 Subject: [Live-devel] RTSP-over-HTTP on the server side. Message-ID: <5e008ff40808281208h44321497y621064770eb6ac92@mail.gmail.com> Hi Ross, I'm just wondering what is the current status of RTSP-over-HTTP implementation on the Live555 Streaming server. The only information I see, is the following comment found in "live555MediaServer.cpp" #if 0 // RTSP-over-HTTP tunneling is not yet working // 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 number (8000). Thanks for your help as always! Ken Seo -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at subfocal.net Thu Aug 28 13:31:12 2008 From: mike at subfocal.net (Mike Mueller) Date: Thu, 28 Aug 2008 16:31:12 -0400 Subject: [Live-devel] openRTSP producing bad video files Message-ID: <20080828203112.GM20854@samus.subfocal.net> Hi All, I am having a problem with openRTSP recording video from a few Axis video cameras. I'm basically running openRTSP like this: openRTSP -q -b 80000 -w 704 -h 480 -f 30 rtsp://url > foo.mov When I play foo.mov in mplayer, it's jerky and plays too fast, where 10 seconds of wall time will go by in 4 to 6 seconds. Despite the playback issue, the video looks fine. If I try to play the mov file in Quicktime (on a Windows PC), Quicktime refuses and says it doesn't understand the file format. I was under the impression that the command I'm using should produce a valid .mov file. There are a lot of variables on the camera for controlling the MP4 stream it produces, so perhaps the answer lies in tweaking there. Unfortunately, so far playing with them has not helped. My version of openRTSP was built from sources on Fedora 6, using live555-latest.tar.gz on July 30, 2008. Any ideas? Thanks, Mike -- Mike Mueller mike at subfocal.net From finlayson at live555.com Thu Aug 28 15:51:29 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 28 Aug 2008 15:51:29 -0700 Subject: [Live-devel] openRTSP producing bad video files In-Reply-To: <20080828203112.GM20854@samus.subfocal.net> References: <20080828203112.GM20854@samus.subfocal.net> Message-ID: >I am having a problem with openRTSP recording video from a few Axis >video cameras. I'm basically running openRTSP like this: > > openRTSP -q -b 80000 -w 704 -h 480 -f 30 rtsp://url > foo.mov > >When I play foo.mov in mplayer, it's jerky and plays too fast, where 10 >seconds of wall time will go by in 4 to 6 seconds. It sounds like your "-f" (frame rate) parameter is wrong. Perhaps it should be 15, or 12 frames-per-second? You need to get that correct. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Thu Aug 28 15:52:09 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 28 Aug 2008 15:52:09 -0700 Subject: [Live-devel] RTSP-over-HTTP on the server side. In-Reply-To: <5e008ff40808281208h44321497y621064770eb6ac92@mail.gmail.com> References: <5e008ff40808281208h44321497y621064770eb6ac92@mail.gmail.com> Message-ID: >I'm just wondering what is the current status of RTSP-over-HTTP >implementation on the Live555 Streaming server. It's partially implemented, but not all there yet. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From mike at subfocal.net Thu Aug 28 16:24:35 2008 From: mike at subfocal.net (Mike Mueller) Date: Thu, 28 Aug 2008 19:24:35 -0400 Subject: [Live-devel] openRTSP producing bad video files In-Reply-To: References: <20080828203112.GM20854@samus.subfocal.net> Message-ID: <20080828232435.GO20854@samus.subfocal.net> On Thu, Aug 28, 2008 at 03:51:29PM -0700, Ross Finlayson wrote: >> I am having a problem with openRTSP recording video from a few Axis >> video cameras. I'm basically running openRTSP like this: >> >> openRTSP -q -b 80000 -w 704 -h 480 -f 30 rtsp://url > foo.mov >> >> When I play foo.mov in mplayer, it's jerky and plays too fast, where 10 >> seconds of wall time will go by in 4 to 6 seconds. > > It sounds like your "-f" (frame rate) parameter is wrong. Perhaps it > should be 15, or 12 frames-per-second? You need to get that correct. I did try 15 with similar results. I can try other values tomorrow I suppose. Is it possible that the mpeg4 stream actually has a variable framerate? -- Mike Mueller mike at subfocal.net From finlayson at live555.com Thu Aug 28 18:57:28 2008 From: finlayson at live555.com (Ross Finlayson) Date: Thu, 28 Aug 2008 18:57:28 -0700 Subject: [Live-devel] openRTSP producing bad video files In-Reply-To: <20080828232435.GO20854@samus.subfocal.net> References: <20080828203112.GM20854@samus.subfocal.net> <20080828232435.GO20854@samus.subfocal.net> Message-ID: >On Thu, Aug 28, 2008 at 03:51:29PM -0700, Ross Finlayson wrote: >>> I am having a problem with openRTSP recording video from a few Axis >>> video cameras. I'm basically running openRTSP like this: >>> >>> openRTSP -q -b 80000 -w 704 -h 480 -f 30 rtsp://url > foo.mov >>> >>> When I play foo.mov in mplayer, it's jerky and plays too fast, where 10 >>> seconds of wall time will go by in 4 to 6 seconds. >> >> It sounds like your "-f" (frame rate) parameter is wrong. Perhaps it >> should be 15, or 12 frames-per-second? You need to get that correct. > >I did try 15 with similar results. I can try other values tomorrow I >suppose. Is it possible that the mpeg4 stream actually has a variable >framerate? I don't think so. You can find the actual frame rate quite easily: Just play the stream using QuickTime Player, and look at the 'stream information' (there's a menu item for this). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From rmcouat at smartt.com Fri Aug 29 00:24:25 2008 From: rmcouat at smartt.com (Ron McOuat) Date: Fri, 29 Aug 2008 00:24:25 -0700 Subject: [Live-devel] openRTSP producing bad video files In-Reply-To: <20080828232435.GO20854@samus.subfocal.net> References: <20080828203112.GM20854@samus.subfocal.net> <20080828232435.GO20854@samus.subfocal.net> Message-ID: <48B7A429.1080007@smartt.com> On the Axis cameras if you go to Setup then pick Video & Image on the left menu and sub item Image you should find a setting for the max framerate. If it is set to unlimited it will go at a best effort rate which varies by model depending on processor speed and resolution requested. I would set an actual value in here that the camera is capable of sustaining, check the datasheet at www.axis.com for your camera model. As a test I would choose CIF or 320x240 resolution depending on camera model and maybe 10 fps and see if that works for recording and playback in mplayer. The settings you put into the Image form define the MPEG4 stream. My 207MW is set to unlimited and QuickTime is playing the stream from the camera using rtsp://... but QuickTime windows Movie Inspector and Movie Properties do not reveal a frame rate. Changed the setting to limit it to 5 fps still reveals no frame rate value that I could find, testing on version 7.5 on Mac OS X Leopard 10.5.4. Using openRTSP to record and playing the .mov file back Movie Inspector shows the frame rate I used on the openRTSP command line. The SDP description from the camera did have 5.0 fps in the result which matches. Setting the camera back to unlimited, the SDP says 30.0 fps as listed by openRTSP debug output. Playing the .mov file in QuickTime with camera and openRTSP set to 30 fps and w 1280, h 720 plays the movie back several times too fast - I have the time display from the camera turned on so the time on the video is known. The QuickTime time scrub bar is advancing at normal time but the movie is over showing the last frame while the time bar is still advancing showing it is playing back. I tried the same in VLC and 20 sec of video plays in 2 sec watching the video content window but the scrub bar plays at the correct speed for 20 sec. Changed the camera to 320 x 240 at 5 fps, recorded again using openRTSP with correct w, h and f parameters and the video plays back at 2x speed but the scrub bar moves at the correct speed for the movie duration in both VLC and QuickTime. This is with the 2008.07.25 version of live555. I found some movies I recorded from Axis cameras using openRTSP a year ago and they play back correctly, the scrub bar and the frames are in sync and visually the time recorded in the movie advances at the correct time scale in QuickTime and VLC. Something has changed. I have saved most of the versions of live555 between those 2 times so it would be possible for me to do a binary search on the revisions to see at what version the behavior of recorded .mov files changed. Not tonight, need some ZZZs, it is late here, if anyone wants the .mov recorded tonight that plays back too fast let me know. Ron Mike Mueller wrote: > On Thu, Aug 28, 2008 at 03:51:29PM -0700, Ross Finlayson wrote: > >>> I am having a problem with openRTSP recording video from a few Axis >>> video cameras. I'm basically running openRTSP like this: >>> >>> openRTSP -q -b 80000 -w 704 -h 480 -f 30 rtsp://url > foo.mov >>> >>> When I play foo.mov in mplayer, it's jerky and plays too fast, where 10 >>> seconds of wall time will go by in 4 to 6 seconds. >>> >> It sounds like your "-f" (frame rate) parameter is wrong. Perhaps it >> should be 15, or 12 frames-per-second? You need to get that correct. >> > > I did try 15 with similar results. I can try other values tomorrow I > suppose. Is it possible that the mpeg4 stream actually has a variable > framerate? > > From harislye at gmail.com Fri Aug 29 03:01:24 2008 From: harislye at gmail.com (M Haris Lye) Date: Fri, 29 Aug 2008 18:01:24 +0800 Subject: [Live-devel] H264 RTP streaming tutorial Message-ID: Hello, May I inquire if anybody has the H264 RTP streaming tutorial as mentioned in the previous posting http://www.mail-archive.com/live-devel at lists.live555.com/msg00238.html I hope it can be shared with live555 newbie like me. I need to refer to it to create a customized streaming server for a research project. Thank you so much. It is greatly appreciated. Regards Haris -------------- next part -------------- An HTML attachment was scrubbed... URL: From linux_is_next at hotmail.com Fri Aug 29 07:35:18 2008 From: linux_is_next at hotmail.com (bos marcel) Date: Fri, 29 Aug 2008 16:35:18 +0200 Subject: [Live-devel] SDP and 3GP and Mpeg4 In-Reply-To: <20080826065222.3226B55800E@LGEMRELSE7Q.lge.com> References: <20080826065222.3226B55800E@LGEMRELSE7Q.lge.com> Message-ID: Any one knows how to configure this? is there a howto file? --------------------------------------------http://culture.zapto.org allt du vill veta From: soumya.patra at lge.comTo: live-devel at ns.live555.comDate: Tue, 26 Aug 2008 12:22:22 +0530Subject: [Live-devel] RTSP Authentication Hi Ross, We have Streaming server which streams H264 perfectly (both unicast & multicast). We have created a dynamic RTSP server for streaming with different media codecs. Now we want to use digest authentication in our RTSP Server for some specific Client. Can you explain how to use Authenticator class. Waiting for your response. Regards Soumya _________________________________________________________________ Skaffa Messenger i mobilen! http://windowslivemobile.msn.com/Homepage.aspx?lang=se-se -------------- next part -------------- An HTML attachment was scrubbed... URL: From finlayson at live555.com Fri Aug 29 13:13:49 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 29 Aug 2008 13:13:49 -0700 Subject: [Live-devel] SDP and 3GP and Mpeg4 In-Reply-To: References: <20080826065222.3226B55800E@LGEMRELSE7Q.lge.com> Message-ID: > >Any one knows how to configure this? is there a howto file? What a ridiculously vague question. You're going to have to be a *lot* more specific if you have any hope of being understood. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ From finlayson at live555.com Fri Aug 29 13:16:28 2008 From: finlayson at live555.com (Ross Finlayson) Date: Fri, 29 Aug 2008 13:16:28 -0700 Subject: [Live-devel] openRTSP producing bad video files In-Reply-To: <48B7A429.1080007@smartt.com> References: <20080828203112.GM20854@samus.subfocal.net> <20080828232435.GO20854@samus.subfocal.net> <48B7A429.1080007@smartt.com> Message-ID: >Something has changed. The code that writes '.mov' or '.mp4' files is "QuickTimeFileSink.cpp". In the past year or so, there have been no significant changes to this file, apart from adding support for H.264 video (which is not relevant for your stream). -- Ross Finlayson Live Networks, Inc. http://www.live555.com/