From silvije.milisic at carnet.hr Thu Jun 1 00:42:58 2006
From: silvije.milisic at carnet.hr (Silvije)
Date: Thu, 01 Jun 2006 09:42:58 +0200
Subject: [Live-devel] testRelay
In-Reply-To: <7.0.1.0.1.20060531094407.01ef9c90@live555.com>
References: <447D5897.70805@carnet.hr>
<7.0.1.0.1.20060531094407.01ef9c90@live555.com>
Message-ID: <447E9A82.3010101@carnet.hr>
Ross Finlayson wrote:
>> I am using testRelay to relay multicast from one network to another.
>> I have PC with two network cards, each in different network.
>>
>> Multicast source is relayed but there is problem: Total multicast
>> bitrate of source is about 20 Mbps and when it is relayed it grows up to
>> 100 Mbps..
>>
>> This is very bad.. is there some kind of a bug?
>
> No, but have you modfied the "testRelay.cpp" code?
yes, i have modified source address and port and also
destination address and port
my source is 239.255.255.1:12345
and destination 239.255.255.10:12345
I suspect that
> you have changed the destination multicast address to be the same as
> the incoming multicast address. Don't do this - it will cause
> outgoing multicast packets to get 'looped back' and re-transmitted
> (ad infinitum).
I did not such thing. And output to destination on different network
interfce is done by applying static route into routing table.
Thanx,
Silvije
>
> "testRelay" was not intended to be used to relay traffic between two
> different networks using the same multicast address. (For that, you
> need proper multicast routing - i.e., enabled in the OS.)
>
>
> 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
From finlayson at live555.com Thu Jun 1 01:00:10 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 01 Jun 2006 01:00:10 -0700
Subject: [Live-devel] testRelay
In-Reply-To: <447E9A82.3010101@carnet.hr>
References: <447D5897.70805@carnet.hr>
<7.0.1.0.1.20060531094407.01ef9c90@live555.com>
<447E9A82.3010101@carnet.hr>
Message-ID: <7.0.1.0.1.20060601005848.01f8b5c8@live555.com>
>yes, i have modified source address and port and also
>destination address and port
>
>my source is 239.255.255.1:12345
>and destination 239.255.255.10:12345
Try using a different port (as well as a different multicast address)
for the destination. Some operating systems don't properly handle
different multicast addresses using the same port number.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From lucabe72 at email.it Thu Jun 1 00:24:38 2006
From: lucabe72 at email.it (Luca Abeni)
Date: Thu, 01 Jun 2006 09:24:38 +0200
Subject: [Live-devel] testRelay
In-Reply-To: <447E9A82.3010101@carnet.hr>
References: <447D5897.70805@carnet.hr><7.0.1.0.1.20060531094407.01ef9c90@live555.com><447E9A82.3010101@carnet.hr>
Message-ID: <1149146678.1402.5.camel@labeni.mm.mbigroup.it>
Hi Silvije,
On Thu, 2006-06-01 at 09:42 +0200, Silvije wrote:
[...]
> > No, but have you modfied the "testRelay.cpp" code?
>
> yes, i have modified source address and port and also
> destination address and port
>
> my source is 239.255.255.1:12345
> and destination 239.255.255.10:12345
I did not look much at the source, but I see you are using the same port
for input and output. If the bind() on the input socket is performed on
INADDR_ANY, I suspect you will receive the traffic that you are
forwarding...
I think you have to set ReceivingInterfaceAddr and SendingInterfaceAddr
properly, or to use different ports for input and output.
Of course, I am just speculating ;-)
Luca
--
Proud to be "coglione"
From roy.subrato at gmail.com Thu Jun 1 03:41:25 2006
From: roy.subrato at gmail.com (subrato roy)
Date: Thu, 1 Jun 2006 13:41:25 +0300
Subject: [Live-devel] FramedSource::getNextFrame()
Message-ID: <64849090606010341q102ab7bfj14a06c5cec2b70b7@mail.gmail.com>
hello ,
I am tryting to open 2 input stream(mpeg2-ts) sources simultaneously
in 2 threads...
while one source opens up and works well , the second one complains of
"being read more than once at a time".
can someone please tell me the procedure to handle getNextFrame() to
open more than one source. And also i want to stop the flow to sink
(after videoSource::startPlaying()) for as long as i wish and then
start again. is this possible ?
thank you for all your help
subrato
From finlayson at live555.com Thu Jun 1 06:03:09 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 01 Jun 2006 06:03:09 -0700
Subject: [Live-devel] FramedSource::getNextFrame()
In-Reply-To: <64849090606010341q102ab7bfj14a06c5cec2b70b7@mail.gmail.com
>
References: <64849090606010341q102ab7bfj14a06c5cec2b70b7@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060601060133.01f88228@live555.com>
>I am tryting to open 2 input stream(mpeg2-ts) sources simultaneously
>in 2 threads...
Please, everybody, read the FAQ - and especially
http://www.live555.com/liveMedia/faq.html#threads
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From roy.subrato at gmail.com Thu Jun 1 06:13:32 2006
From: roy.subrato at gmail.com (subrato roy)
Date: Thu, 1 Jun 2006 16:13:32 +0300
Subject: [Live-devel] FramedSource::getNextFrame()
In-Reply-To: <7.0.1.0.1.20060601060133.01f88228@live555.com>
References: <64849090606010341q102ab7bfj14a06c5cec2b70b7@mail.gmail.com>
<7.0.1.0.1.20060601060133.01f88228@live555.com>
Message-ID: <64849090606010613s3c8bc29eq83aa9336e55c4636@mail.gmail.com>
Ok ,
but can i do a sequential 'pause and play' with more than one source
with getNextFrame() ? if yes please tell me how to do it.
On 6/1/06, Ross Finlayson wrote:
>
> >I am tryting to open 2 input stream(mpeg2-ts) sources simultaneously
> >in 2 threads...
>
> Please, everybody, read the FAQ - and especially
> http://www.live555.com/liveMedia/faq.html#threads
>
>
> 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
>
From finlayson at live555.com Thu Jun 1 12:25:26 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 01 Jun 2006 12:25:26 -0700
Subject: [Live-devel] FramedSource::getNextFrame()
In-Reply-To: <64849090606010613s3c8bc29eq83aa9336e55c4636@mail.gmail.com
>
References: <64849090606010341q102ab7bfj14a06c5cec2b70b7@mail.gmail.com>
<7.0.1.0.1.20060601060133.01f88228@live555.com>
<64849090606010613s3c8bc29eq83aa9336e55c4636@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060601121813.01fdfd28@live555.com>
At 06:13 AM 6/1/2006, you wrote:
>Ok ,
>but can i do a sequential 'pause and play' with more than one source
>with getNextFrame() ?
I don't really understand your question. "getNextFrame()" can be
called on many different objects concurrently. (However, once you've
called "getNextFrame()" on an object, you can't call it again on the
*same* object until the first call completes (with delivery to the
downstream client).)
If you want to 'pause' media flow, then call
"MediaSink::stopPlaying()" on your 'sink' object. Then, to restart
it, call "MediaSink::startPlaying()" on your 'sink' object once again.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From bidibulle at operamail.com Thu Jun 1 12:50:38 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Thu, 01 Jun 2006 20:50:38 +0100
Subject: [Live-devel] FramedSource::getNextFrame()
Message-ID: <20060601195038.A406D43CC1@ws5-1.us4.outblaze.com>
Roy,
If I understand correctly what you mean I would say :
PAUSE : simply calls sink.stopPlaying()
PLAY (AGAIN) : simply calls sink.startPlaying(source,...)
Hope this helps,
David
> ----- Original Message -----
> From: "Ross Finlayson"
> To: "LIVE555 Streaming Media - development & use"
> Subject: Re: [Live-devel] FramedSource::getNextFrame()
> Date: Thu, 01 Jun 2006 12:25:26 -0700
>
>
> At 06:13 AM 6/1/2006, you wrote:
> > Ok ,
> > but can i do a sequential 'pause and play' with more than one source
> > with getNextFrame() ?
>
> I don't really understand your question. "getNextFrame()" can be
> called on many different objects concurrently. (However, once you've
> called "getNextFrame()" on an object, you can't call it again on the
> *same* object until the first call completes (with delivery to the
> downstream client).)
>
> If you want to 'pause' media flow, then call
> "MediaSink::stopPlaying()" on your 'sink' object. Then, to restart
> it, call "MediaSink::startPlaying()" on your 'sink' object once again.
>
>
> 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
>
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 8 at http://www.opera.com
Powered by Outblaze
From elios.flht at thales-bm.com Fri Jun 2 01:15:22 2006
From: elios.flht at thales-bm.com (Fabien LAI-HONG-TING)
Date: Fri, 02 Jun 2006 10:15:22 +0200
Subject: [Live-devel] How to receive mp4 video data
Message-ID: <447FF39A.6000109@thales-bm.com>
Hi,
This is certainly a basic question, but i'm a programmer and english
learner .. i need some help.
My goal is to write a basic RTP mp4 video player for windows using
live55 and directshow mp4 decoder.
For the moment i'm trying to receive video data to send them to the mp4
decoder, (i'll deal with the audio track later)
This is my flow control:
RTSP: description parsing, setup video only, play,
while (1)
{
getNextFrame(buffer, max_size_of_Buffer, afterReadingFunc, void *,
onSourceClose, void * )
doEventLoop( flag)
}
in my afterReadingFunc i'm doing: flag = ~0;
I believed that on each getNextFrame call "Buffer" should be filled
with all the data of one frame, with the size in byte of the frame
thanks to the second parameter of afterGettingFunc.
And i thought that i would give each frame one after another to the decoder.
But when i launch the programm, the unsigned frameSize parameter doesn't
correspond to the amount of data written in buffer.
Sometimes the length is taller than the data vailable in the buffer,
sometimes the buffer is empty and not the length,
And i can't make find any similarities between the data received from
getNextFrame and the ethereal capture.
There is a lot of things that i 'm misunderstanding if someone could
give me some hints, that would be very kind!
Thanks
From richard.monette at gmail.com Fri Jun 2 07:47:58 2006
From: richard.monette at gmail.com (Richard Monette)
Date: Fri, 2 Jun 2006 10:47:58 -0400
Subject: [Live-devel] Unresolved External Symbols
Message-ID:
I have successfully built and tested the examples using Visual Studio 2003
on Windows XP.
I am attempting to create my own new project using the libraries. I have
linked all the libraries ( that I could find at least ) and included all the
header files ( that I could find at least ) but I am still getting
unresolved externals. In order to get rid of a bunch of "_printf etc, etc"
errors I have switched to compile in multithreaded debug.
Any ideas?
Thanks very much,
Richard Monette
The errors;
------ Build started: Project: NewLiveProject, Configuration: Debug Win32
------
Linking...
LINK : warning LNK4098: defaultlib 'MSVCRT' conflicts with use of other
libs; use /NODEFAULTLIB:library
libgroupsock.lib(NetAddress.obj) : error LNK2001: unresolved external symbol
__imp__ntohs at 4
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved external
symbol __imp__ntohs at 4
libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external symbol
__imp__ntohs at 4 referenced in function "public: char * __thiscall
RTSPServer::rtspURL(class ServerMediaSession const *)const "
(?rtspURL at RTSPServer@@QBEPADPBVServerMediaSession@@@Z)
libLiveMedia.lib(OnDemandServerMediaSubsession.obj) : error LNK2001:
unresolved external symbol __imp__ntohs at 4
libLiveMedia.lib(RTCP.obj) : error LNK2001: unresolved external symbol
__imp__ntohs at 4
libLiveMedia.lib(RTPInterface.obj) : error LNK2001: unresolved external
symbol __imp__ntohs at 4
libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external symbol
__imp__closesocket at 4 referenced in function "protected: static int __cdecl
RTSPServer::setUpOurSocket(class UsageEnvironment &,class Port &)"
(?setUpOurSocket at RTSPServer@@KAHAAVUsageEnvironment@@AAVPort@@@Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved external
symbol __imp__closesocket at 4
libgroupsock.lib(NetInterface.obj) : error LNK2001: unresolved external
symbol __imp__closesocket at 4
libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external symbol
__imp__listen at 8 referenced in function "protected: static int __cdecl
RTSPServer::setUpOurSocket(class UsageEnvironment &,class Port &)"
(?setUpOurSocket at RTSPServer@@KAHAAVUsageEnvironment@@AAVPort@@@Z)
libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external symbol
__imp__send at 16 referenced in function "private: void __thiscall
RTSPServer::RTSPClientSession::incomingRequestHandler1(void)"
(?incomingRequestHandler1 at RTSPClientSession@RTSPServer@@AAEXXZ)
libLiveMedia.lib(RTPInterface.obj) : error LNK2001: unresolved external
symbol __imp__send at 16
libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external symbol
__imp__accept at 12 referenced in function "private: void __thiscall
RTSPServer::incomingConnectionHandler1(void)"
(?incomingConnectionHandler1 at RTSPServer@@AAEXXZ)
libLiveMedia.lib(MediaSink.obj) : error LNK2019: unresolved external symbol
__imp__htonl at 4 referenced in function "public: void __thiscall
OutPacketBuffer::enqueueWord(unsigned int)" (?enqueueWord at OutPacketBuffer@@
QAEXI at Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__htonl at 4 referenced in function "unsigned int __cdecl
getBufferSize(class UsageEnvironment &,int,int)"
(?getBufferSize@@YAIAAVUsageEnvironment@@HH at Z)
libLiveMedia.lib(MediaSink.obj) : error LNK2019: unresolved external symbol
__imp__ntohl at 4 referenced in function "public: unsigned int __thiscall
OutPacketBuffer::extractWord(unsigned int)" (?extractWord at OutPacketBuffer@@
QAEII at Z)
libLiveMedia.lib(RTCP.obj) : error LNK2019: unresolved external symbol
__imp__ntohl at 4 referenced in function "protected: virtual __thiscall
RTCPInstance::~RTCPInstance(void)" (??1RTCPInstance@@MAE at XZ)
libgroupsock.lib(NetAddress.obj) : error LNK2001: unresolved external symbol
__imp__ntohl at 4
libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved external
symbol __imp__ntohl at 4
libLiveMedia.lib(OnDemandServerMediaSubsession.obj) : error LNK2019:
unresolved external symbol __imp__gethostname at 8 referenced in function
"protected: __thiscall
OnDemandServerMediaSubsession::OnDemandServerMediaSubsession(class
UsageEnvironment &,unsigned int)" (??0OnDemandServerMediaSubsession@@
IAE at AAVUsageEnvironment@@I at Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__gethostname at 8 referenced in function "int __cdecl
setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
(?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@I at Z)
libgroupsock.lib(NetAddress.obj) : error LNK2019: unresolved external symbol
__imp__htons at 4 referenced in function "public: __thiscall
Port::Port(unsigned short)" (??0Port@@QAE at G@Z)
libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
__imp__inet_addr at 4 referenced in function _our_inet_addr
libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
__imp__inet_ntoa at 4 referenced in function _our_inet_ntoa
libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
__imp__WSACleanup at 0 referenced in function _initializeWinsockIfNecessary
libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
__imp__WSAStartup at 8 referenced in function _initializeWinsockIfNecessary
libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
__imp__gethostbyname at 4 referenced in function _our_gethostbyname
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__gethostbyname at 4 referenced in function "int __cdecl
setupStreamSocket(class UsageEnvironment &,class Port,unsigned int)"
(?setupStreamSocket@@YAHAAVUsageEnvironment@@VPort@@I at Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__bind at 12 referenced in function "int __cdecl
setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
(?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@I at Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__setsockopt at 20 referenced in function "int __cdecl
setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
(?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@I at Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__socket at 12 referenced in function "int __cdecl
setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
(?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@I at Z)
libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2019:
unresolved external symbol __imp__socket at 12 referenced in function _$E8
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__ioctlsocket at 12 referenced in function "int __cdecl
setupStreamSocket(class UsageEnvironment &,class Port,unsigned int)"
(?setupStreamSocket@@YAHAAVUsageEnvironment@@VPort@@I at Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__recvfrom at 24 referenced in function "int __cdecl
readSocket(class UsageEnvironment &,int,unsigned char *,unsigned int,struct
sockaddr_in &,struct timeval *)"
(?readSocket@@YAHAAVUsageEnvironment@@HPAEIAAUsockaddr_in@@PAUtimeval@@@Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol ___WSAFDIsSet at 8 referenced in function "int __cdecl
blockUntilReadable(class UsageEnvironment &,int,struct timeval *)"
(?blockUntilReadable@@YAHAAVUsageEnvironment@@HPAUtimeval@@@Z)
libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2019:
unresolved external symbol ___WSAFDIsSet at 8 referenced in function "public:
virtual void * __thiscall BasicTaskScheduler::`scalar deleting
destructor'(unsigned int)" (??_GBasicTaskScheduler@@UAEPAXI at Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__select at 20 referenced in function "int __cdecl
blockUntilReadable(class UsageEnvironment &,int,struct timeval *)"
(?blockUntilReadable@@YAHAAVUsageEnvironment@@HPAUtimeval@@@Z)
libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2001:
unresolved external symbol __imp__select at 20
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__sendto at 24 referenced in function "unsigned int __cdecl
writeSocket(class UsageEnvironment &,int,struct in_addr,class Port,unsigned
char,unsigned char *,unsigned int)"
(?writeSocket@@YAIAAVUsageEnvironment@@HUin_addr@@VPort@@EPAEI at Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__getsockopt at 20 referenced in function "unsigned int __cdecl
getBufferSize(class UsageEnvironment &,int,int)"
(?getBufferSize@@YAIAAVUsageEnvironment@@HH at Z)
libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
symbol __imp__getsockname at 12 referenced in function "unsigned int __cdecl
getSourcePort0(int,unsigned short &)" (?getSourcePort0@@YAIHAAG at Z)
libBasicUsageEnvironment.lib(BasicUsageEnvironment.obj) : error LNK2019:
unresolved external symbol __imp__WSAGetLastError at 0 referenced in function
"public: virtual int __thiscall BasicUsageEnvironment::getErrno(void)const "
(?getErrno at BasicUsageEnvironment@@UBEHXZ)
libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2019:
unresolved external symbol __imp__WSAGetLastError at 0 referenced in function
"protected: virtual void __thiscall BasicTaskScheduler::SingleStep(unsigned
int)" (?SingleStep at BasicTaskScheduler@@MAEXI at Z)
Debug/NewLiveProject.exe : fatal error LNK1120: 25 unresolved externals
Build log was saved at "file://c:\Documents and
Settings\support\Desktop\Richards
Live\live\NewLiveProject\NewLiveProject\Debug\BuildLog.htm"
NewLiveProject - 44 error(s), 1 warning(s)
---------------------- Done ----------------------
Build: 0 succeeded, 1 failed, 0 skipped
-------------- next part --------------
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From richard.monette at gmail.com Fri Jun 2 07:50:40 2006
From: richard.monette at gmail.com (Richard Monette)
Date: Fri, 2 Jun 2006 10:50:40 -0400
Subject: [Live-devel] Unresolved External Symbols
In-Reply-To:
References:
Message-ID:
Solved my own problem. The problem is solved by linking to ws2_32.lib.
Found the answer here:
http://www.openldap.org/lists/openldap-software/200105/msg00552.html
Cheers,
Rick
On 6/2/06, Richard Monette wrote:
>
> I have successfully built and tested the examples using Visual Studio 2003
> on Windows XP.
>
> I am attempting to create my own new project using the libraries. I have
> linked all the libraries ( that I could find at least ) and included all the
> header files ( that I could find at least ) but I am still getting
> unresolved externals. In order to get rid of a bunch of "_printf etc, etc"
> errors I have switched to compile in multithreaded debug.
>
> Any ideas?
>
> Thanks very much,
>
> Richard Monette
>
> The errors;
>
> ------ Build started: Project: NewLiveProject, Configuration: Debug Win32
> ------
>
> Linking...
> LINK : warning LNK4098: defaultlib 'MSVCRT' conflicts with use of other
> libs; use /NODEFAULTLIB:library
> libgroupsock.lib(NetAddress.obj) : error LNK2001: unresolved external
> symbol __imp__ntohs at 4
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved external
> symbol __imp__ntohs at 4
> libLiveMedia.lib(RTSPServer.obj ) : error LNK2019: unresolved external
> symbol __imp__ntohs at 4 referenced in function "public: char * __thiscall
> RTSPServer::rtspURL(class ServerMediaSession const *)const "
> (?rtspURL at RTSPServer@@QBEPADPBVServerMediaSession@@@Z)
> libLiveMedia.lib(OnDemandServerMediaSubsession.obj) : error LNK2001:
> unresolved external symbol __imp__ntohs at 4
> libLiveMedia.lib(RTCP.obj) : error LNK2001: unresolved external symbol
> __imp__ntohs at 4
> libLiveMedia.lib (RTPInterface.obj) : error LNK2001: unresolved external
> symbol __imp__ntohs at 4
> libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external
> symbol __imp__closesocket at 4 referenced in function "protected: static int
> __cdecl RTSPServer::setUpOurSocket(class UsageEnvironment &,class Port &)"
> (?setUpOurSocket at RTSPServer@@KAHAAVUsageEnvironment@@AAVPort@@@Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved external
> symbol __imp__closesocket at 4
> libgroupsock.lib(NetInterface.obj) : error LNK2001: unresolved external
> symbol __imp__closesocket at 4
> libLiveMedia.lib (RTSPServer.obj) : error LNK2019: unresolved external
> symbol __imp__listen at 8 referenced in function "protected: static int __cdecl
> RTSPServer::setUpOurSocket(class UsageEnvironment &,class Port &)"
> (?setUpOurSocket at RTSPServer@@KAHAAVUsageEnvironment@@AAVPort@@@Z)
> libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external
> symbol __imp__send at 16 referenced in function "private: void __thiscall
> RTSPServer::RTSPClientSession::incomingRequestHandler1(void)"
> (?incomingRequestHandler1@ RTSPClientSession at RTSPServer@@AAEXXZ)
> libLiveMedia.lib(RTPInterface.obj) : error LNK2001: unresolved external
> symbol __imp__send at 16
> libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external
> symbol __imp__accept at 12 referenced in function "private: void __thiscall
> RTSPServer::incomingConnectionHandler1(void)"
> (?incomingConnectionHandler1 at RTSPServer@@AAEXXZ)
> libLiveMedia.lib(MediaSink.obj) : error LNK2019: unresolved external
> symbol __imp__htonl at 4 referenced in function "public: void __thiscall
> OutPacketBuffer::enqueueWord(unsigned int)" (?enqueueWord at OutPacketBuffer@@
> QAEXI at Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__htonl at 4 referenced in function "unsigned int __cdecl
> getBufferSize(class UsageEnvironment &,int,int)"
> (?getBufferSize@@YAIAAVUsageEnvironment@@ HH at Z)
> libLiveMedia.lib(MediaSink.obj) : error LNK2019: unresolved external
> symbol __imp__ntohl at 4 referenced in function "public: unsigned int
> __thiscall OutPacketBuffer::extractWord(unsigned int)"
> (?extractWord at OutPacketBuffer@@ QAEII at Z)
> libLiveMedia.lib(RTCP.obj) : error LNK2019: unresolved external symbol
> __imp__ntohl at 4 referenced in function "protected: virtual __thiscall
> RTCPInstance::~RTCPInstance(void)" (??1RTCPInstance@@MAE at XZ )
> libgroupsock.lib(NetAddress.obj) : error LNK2001: unresolved external
> symbol __imp__ntohl at 4
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved external
> symbol __imp__ntohl at 4
> libLiveMedia.lib(OnDemandServerMediaSubsession.obj ) : error LNK2019:
> unresolved external symbol __imp__gethostname at 8 referenced in function
> "protected: __thiscall
> OnDemandServerMediaSubsession::OnDemandServerMediaSubsession(class
> UsageEnvironment &,unsigned int)" (??0OnDemandServerMediaSubsession@@
> IAE at AAVUsageEnvironment@@I at Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__gethostname at 8 referenced in function "int __cdecl
> setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
> (?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> libgroupsock.lib(NetAddress.obj) : error LNK2019: unresolved external
> symbol __imp__htons at 4 referenced in function "public: __thiscall
> Port::Port(unsigned short)" (??0Port@@QAE at G@Z)
> libgroupsock.lib(inet.obj ) : error LNK2019: unresolved external symbol
> __imp__inet_addr at 4 referenced in function _our_inet_addr
> libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
> __imp__inet_ntoa at 4 referenced in function _our_inet_ntoa
> libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
> __imp__WSACleanup at 0 referenced in function _initializeWinsockIfNecessary
> libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
> __imp__WSAStartup at 8 referenced in function _initializeWinsockIfNecessary
> libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
> __imp__gethostbyname at 4 referenced in function _our_gethostbyname
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__gethostbyname at 4 referenced in function "int __cdecl
> setupStreamSocket(class UsageEnvironment &,class Port,unsigned int)"
> (?setupStreamSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__bind at 12 referenced in function "int __cdecl
> setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
> (?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__setsockopt at 20 referenced in function "int __cdecl
> setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
> (?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__socket at 12 referenced in function "int __cdecl
> setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
> (?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2019:
> unresolved external symbol __imp__socket at 12 referenced in function _$E8
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__ioctlsocket at 12 referenced in function "int __cdecl
> setupStreamSocket(class UsageEnvironment &,class Port,unsigned int)"
> (?setupStreamSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__recvfrom at 24 referenced in function "int __cdecl
> readSocket(class UsageEnvironment &,int,unsigned char *,unsigned int,struct
> sockaddr_in &,struct timeval *)"
> (?readSocket@@YAHAAVUsageEnvironment@@HPAEIAAUsockaddr_in@@PAUtimeval@@@Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol ___WSAFDIsSet at 8 referenced in function "int __cdecl
> blockUntilReadable(class UsageEnvironment &,int,struct timeval *)"
> (?blockUntilReadable@@YAHAAVUsageEnvironment@@HPAUtimeval@@@Z)
> libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2019:
> unresolved external symbol ___WSAFDIsSet at 8 referenced in function "public:
> virtual void * __thiscall BasicTaskScheduler::`scalar deleting
> destructor'(unsigned int)" (??_GBasicTaskScheduler@@ UAEPAXI at Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__select at 20 referenced in function "int __cdecl
> blockUntilReadable(class UsageEnvironment &,int,struct timeval *)"
> (?blockUntilReadable@@YAHAAVUsageEnvironment@@HPAUtimeval@@@Z)
> libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2001:
> unresolved external symbol __imp__select at 20
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__sendto at 24 referenced in function "unsigned int __cdecl
> writeSocket(class UsageEnvironment &,int,struct in_addr,class Port,unsigned
> char,unsigned char *,unsigned int)"
> (?writeSocket@@YAIAAVUsageEnvironment@@HUin_addr@@VPort@@ EPAEI at Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__getsockopt at 20 referenced in function "unsigned int __cdecl
> getBufferSize(class UsageEnvironment &,int,int)"
> (?getBufferSize@@YAIAAVUsageEnvironment@@ HH at Z)
> libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> symbol __imp__getsockname at 12 referenced in function "unsigned int __cdecl
> getSourcePort0(int,unsigned short &)" (?getSourcePort0@@ YAIHAAG at Z)
> libBasicUsageEnvironment.lib(BasicUsageEnvironment.obj) : error LNK2019:
> unresolved external symbol __imp__WSAGetLastError at 0 referenced in function
> "public: virtual int __thiscall BasicUsageEnvironment::getErrno(void)const "
> (?getErrno at BasicUsageEnvironment@@UBEHXZ)
> libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2019:
> unresolved external symbol __imp__WSAGetLastError at 0 referenced in function
> "protected: virtual void __thiscall BasicTaskScheduler::SingleStep(unsigned
> int)" (?SingleStep at BasicTaskScheduler@@ MAEXI at Z)
> Debug/NewLiveProject.exe : fatal error LNK1120: 25 unresolved externals
>
> Build log was saved at "file://c:\Documents and
> Settings\support\Desktop\Richards
> Live\live\NewLiveProject\NewLiveProject\Debug\BuildLog.htm"
> NewLiveProject - 44 error(s), 1 warning(s)
>
>
> ---------------------- Done ----------------------
>
> Build: 0 succeeded, 1 failed, 0 skipped
>
>
>
>
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From yetanotherid at gmail.com Fri Jun 2 12:54:17 2006
From: yetanotherid at gmail.com (S Mathur)
Date: Fri, 2 Jun 2006 15:54:17 -0400
Subject: [Live-devel] RTSPclient and DarwinInjector in the same application
Message-ID:
Hello,
I have modified the openRTSP/playCommon.cpp code such that it
intercepts the M4A-LATM subsession in my streaming session and feeds
it to a decoder filter. The decoder filter feeds data to an encoder
filter which in turns feeds an MPEG4LATMRTPSink.
The subsession related code is as follows...
subsession->sink = latmSink;
subsession->sink->startPlaying(*aacEncFilter, subsessionAfterPlaying,
subsession);
....
My latmSink is also associated with a DarwinInjector:
injector->addStream(latmSink, audioRTCP);
injector->setDestination(..)
My problem: A get a select() failed error in
BasicTaskScheduler::SingleStep() because RTCPInstance's destructor
gets invoked prematurely. The ANNOUNCE (of the modified stream named
liveST.sdp) to the Darwin server does goes through successfully.
Here is the transcript, error is at the bottom:
Received ANNOUNCE response: RTSP/1.0 200 OK
Server: DSS/5.6.0 (Build/496; Platform/Win32; Release/Development;
BuildNote/LP64 Support)
Cseq: 4
RTCPInstance[0x86cd5b8]::RTCPInstance()
schedule(1.786040->1149276500.435798)
Sending request: SETUP rtsp://myDSS:554/liveST.sdp/trackID=1 RTSP/1.0
CSeq: 5
Transport: RTP/AVP/TCP;unicast;mode=receive;interleaved=0-1
User-Agent: Injector (LIVE555 Streaming Media v2006.03.16)
Received SETUP response: RTSP/1.0 200 OK
Server: DSS/5.6.0 (Build/496; Platform/Win32; Release/Development;
BuildNote/LP64 Support)
Cseq: 5
Cache-Control: no-cache
Session: 14018773254377
Date: Fri, 02 Jun 2006 19:28:29 GMT
Expires: Fri, 02 Jun 2006 19:28:29 GMT
Transport: RTP/AVP/TCP;unicast;mode=receive;interleaved=0-1
Sending request: PLAY rtsp://myDSS:554/liveST.sdp RTSP/1.0
CSeq: 6
Session: 14018773254377
Range: npt=0.000-
User-Agent: Injector (LIVE555 Streaming Media v2006.03.16)
Received PLAY response: RTSP/1.0 200 OK
Server: DSS/5.6.0 (Build/496; Platform/Win32; Release/Development;
BuildNote/LP64 Support)
Cseq: 6
Session: 14018773254377
RTP-Info: url=rtsp://myDSS:554/liveST.sdp/trackID=1
RTCPInstance[0x86cd5b8]::~RTCPInstance()
sending BYE
sending RTCP packet
80c90001 c923bac9 81cb0001 c923bac9
sendRTPOverTCP: 16 bytes over channel 1 (socket 8)
sendRTPOverTCP: completed
Play this stream (from the Darwin Streaming Server) using the URL:
rtsp://myDSS/liveST.sdp
[startPlaying(latmSink) gets called here. Greenday64kbpsPlus.3gp is
the input stream to be transcoded]
Sending request: PLAY rtsp://myDSS:554/Greenday64kbpsPlus.3gp RTSP/1.0
CSeq: 7
Session: 47446503751336
Range: npt=0.000-
User-Agent: ./testProgram (LIVE555 Streaming Media v2006.03.16)
Received PLAY response: RTSP/1.0 200 OK
Server: DSS/5.6.0 (Build/496; Platform/Win32; Release/Development;
BuildNote/LP64 Support)
Cseq: 7
Session: 47446503751336
Range: npt=0.00000-285.83667
RTP-Info: url=rtsp://myDSS:554/Greenday64kbpsPlus.3gp/trackID=65536;seq=15921;rtptime=30605
Started playing session
Receiving streamed data (for up to 290.836670 seconds)...
BasicTaskScheduler::SingleStep(): select() fails: Bad file descriptor
Any suggestions on what I am doing wrong?
Thanks,
Mathur
From kushal.dalal at einfochips.com Sun Jun 4 21:43:56 2006
From: kushal.dalal at einfochips.com (Kushal Dalal)
Date: Mon, 5 Jun 2006 10:13:56 +0530
Subject: [Live-devel] Global Macro
Message-ID: <200606050444.k554i31S002769@ns.live555.com>
Hi All
I want to define a global macro which can be used in all files of live.
Where (in which header file/files) should I define it?
Thanks in advance.
Kushal
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From finlayson at live555.com Sun Jun 4 22:34:52 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 04 Jun 2006 22:34:52 -0700
Subject: [Live-devel] Global Macro
In-Reply-To: <200606050444.k554i31S002769@ns.live555.com>
References: <200606050444.k554i31S002769@ns.live555.com>
Message-ID: <7.0.1.0.1.20060604223319.0212e4e0@live555.com>
>I want to define a global macro which can be used in all files of live.
>Where (in which header file/files) should I define it?
I suggest "groupsock/include/NetCommon.h". That file is included by
most of the others.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From vlcplayer at gmail.com Mon Jun 5 04:29:40 2006
From: vlcplayer at gmail.com (Lukasz Kondrad)
Date: Mon, 5 Jun 2006 14:29:40 +0300
Subject: [Live-devel] [live-devel] Diffrent input
Message-ID:
Hello,
Currently I am working on a player which will be able to decode television
in the DVB-H standard. I think that your library would be suitable for my
solution. I have only one problem. I need that the source of RTP frames
won't be from the network layer of OS but from my own stream. I demux
transport stream and select out the mpe section in which is IP/UDP/RTP
datagrams.
May you give me some hint, or it is possible to do that by use your library.
Which functions shall I concentrate on ??
best regards
Lukasz
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From vlcplayer at gmail.com Mon Jun 5 05:20:29 2006
From: vlcplayer at gmail.com (Lukasz Kondrad)
Date: Mon, 5 Jun 2006 15:20:29 +0300
Subject: [Live-devel] [live-devel] Read from diffrent input
Message-ID:
Hello,
Currently I am working on a player which will be able to decode television
in the DVB-H standard. I think that your library would be suitable for my
solution. I have only one problem. I need that the source of RTP frames
won't be from the network layer of OS but from my own stream. I demux
transport stream and select out the mpe section in which is IP/UDP/RTP
datagrams.
May you give me some hint, or it is possible to do that by use your library.
Which functions shall I concentrate on ??
best regards
Lukasz
-------------- next part --------------
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From finlayson at live555.com Mon Jun 5 11:14:58 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 05 Jun 2006 11:14:58 -0700
Subject: [Live-devel] [live-devel] Read from diffrent input
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060605111216.01f57148@live555.com>
>Currently I am working on a player which will be able to decode
>television in the DVB-H standard. I think that your library would be
>suitable for my solution. I have only one problem. I need that the
>source of RTP frames won't be from the network layer of OS but from
>my own stream. I demux transport stream and select out the mpe
>section in which is IP/UDP/RTP datagrams.
>May you give me some hint, or it is possible to do that by use your
>library. Which functions shall I concentrate on ??
Unfortunately the "LIVE555 Streaming Media" libraries are not well
set up to handle input from 'synthetic' RTP sources (rather than real
network interfaces). (This is due to a design flaw in the code; it
will be fixed someday.)
In the meantime, if you really want to try to do this with the
existing code, I suggest looking at modifying the implementation of
the "RTPInterface" class (in partiocular, in the way that it asks
for, and handles, new incoming RTP packets).
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From yetanotherid at gmail.com Mon Jun 5 12:14:16 2006
From: yetanotherid at gmail.com (S Mathur)
Date: Mon, 5 Jun 2006 15:14:16 -0400
Subject: [Live-devel] RTSPclient and DarwinInjector in the same
application
Message-ID:
Hello again,
I created and started an additional taskSchedular for my Darwin
Injector, and I can now recieve as well as transmit RTSP/RTP streams
within the same application.
Thanks,
Mathur
On 6/5/06, live-devel-request at ns.live555.com
wrote:
> 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: Unresolved External Symbols (Richard Monette)
> 2. RTSPclient and DarwinInjector in the same application (S Mathur)
> 3. Global Macro (Kushal Dalal)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 2 Jun 2006 10:50:40 -0400
> From: "Richard Monette"
> Subject: Re: [Live-devel] Unresolved External Symbols
> To: "LIVE555 Streaming Media - development & use"
>
> Message-ID:
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Solved my own problem. The problem is solved by linking to ws2_32.lib.
>
> Found the answer here:
> http://www.openldap.org/lists/openldap-software/200105/msg00552.html
>
> Cheers,
>
> Rick
>
> On 6/2/06, Richard Monette wrote:
> >
> > I have successfully built and tested the examples using Visual Studio 2003
> > on Windows XP.
> >
> > I am attempting to create my own new project using the libraries. I have
> > linked all the libraries ( that I could find at least ) and included all the
> > header files ( that I could find at least ) but I am still getting
> > unresolved externals. In order to get rid of a bunch of "_printf etc, etc"
> > errors I have switched to compile in multithreaded debug.
> >
> > Any ideas?
> >
> > Thanks very much,
> >
> > Richard Monette
> >
> > The errors;
> >
> > ------ Build started: Project: NewLiveProject, Configuration: Debug Win32
> > ------
> >
> > Linking...
> > LINK : warning LNK4098: defaultlib 'MSVCRT' conflicts with use of other
> > libs; use /NODEFAULTLIB:library
> > libgroupsock.lib(NetAddress.obj) : error LNK2001: unresolved external
> > symbol __imp__ntohs at 4
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved external
> > symbol __imp__ntohs at 4
> > libLiveMedia.lib(RTSPServer.obj ) : error LNK2019: unresolved external
> > symbol __imp__ntohs at 4 referenced in function "public: char * __thiscall
> > RTSPServer::rtspURL(class ServerMediaSession const *)const "
> > (?rtspURL at RTSPServer@@QBEPADPBVServerMediaSession@@@Z)
> > libLiveMedia.lib(OnDemandServerMediaSubsession.obj) : error LNK2001:
> > unresolved external symbol __imp__ntohs at 4
> > libLiveMedia.lib(RTCP.obj) : error LNK2001: unresolved external symbol
> > __imp__ntohs at 4
> > libLiveMedia.lib (RTPInterface.obj) : error LNK2001: unresolved external
> > symbol __imp__ntohs at 4
> > libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external
> > symbol __imp__closesocket at 4 referenced in function "protected: static int
> > __cdecl RTSPServer::setUpOurSocket(class UsageEnvironment &,class Port &)"
> > (?setUpOurSocket at RTSPServer@@KAHAAVUsageEnvironment@@AAVPort@@@Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved external
> > symbol __imp__closesocket at 4
> > libgroupsock.lib(NetInterface.obj) : error LNK2001: unresolved external
> > symbol __imp__closesocket at 4
> > libLiveMedia.lib (RTSPServer.obj) : error LNK2019: unresolved external
> > symbol __imp__listen at 8 referenced in function "protected: static int __cdecl
> > RTSPServer::setUpOurSocket(class UsageEnvironment &,class Port &)"
> > (?setUpOurSocket at RTSPServer@@KAHAAVUsageEnvironment@@AAVPort@@@Z)
> > libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external
> > symbol __imp__send at 16 referenced in function "private: void __thiscall
> > RTSPServer::RTSPClientSession::incomingRequestHandler1(void)"
> > (?incomingRequestHandler1@ RTSPClientSession at RTSPServer@@AAEXXZ)
> > libLiveMedia.lib(RTPInterface.obj) : error LNK2001: unresolved external
> > symbol __imp__send at 16
> > libLiveMedia.lib(RTSPServer.obj) : error LNK2019: unresolved external
> > symbol __imp__accept at 12 referenced in function "private: void __thiscall
> > RTSPServer::incomingConnectionHandler1(void)"
> > (?incomingConnectionHandler1 at RTSPServer@@AAEXXZ)
> > libLiveMedia.lib(MediaSink.obj) : error LNK2019: unresolved external
> > symbol __imp__htonl at 4 referenced in function "public: void __thiscall
> > OutPacketBuffer::enqueueWord(unsigned int)" (?enqueueWord at OutPacketBuffer@@
> > QAEXI at Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__htonl at 4 referenced in function "unsigned int __cdecl
> > getBufferSize(class UsageEnvironment &,int,int)"
> > (?getBufferSize@@YAIAAVUsageEnvironment@@ HH at Z)
> > libLiveMedia.lib(MediaSink.obj) : error LNK2019: unresolved external
> > symbol __imp__ntohl at 4 referenced in function "public: unsigned int
> > __thiscall OutPacketBuffer::extractWord(unsigned int)"
> > (?extractWord at OutPacketBuffer@@ QAEII at Z)
> > libLiveMedia.lib(RTCP.obj) : error LNK2019: unresolved external symbol
> > __imp__ntohl at 4 referenced in function "protected: virtual __thiscall
> > RTCPInstance::~RTCPInstance(void)" (??1RTCPInstance@@MAE at XZ )
> > libgroupsock.lib(NetAddress.obj) : error LNK2001: unresolved external
> > symbol __imp__ntohl at 4
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2001: unresolved external
> > symbol __imp__ntohl at 4
> > libLiveMedia.lib(OnDemandServerMediaSubsession.obj ) : error LNK2019:
> > unresolved external symbol __imp__gethostname at 8 referenced in function
> > "protected: __thiscall
> > OnDemandServerMediaSubsession::OnDemandServerMediaSubsession(class
> > UsageEnvironment &,unsigned int)" (??0OnDemandServerMediaSubsession@@
> > IAE at AAVUsageEnvironment@@I at Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__gethostname at 8 referenced in function "int __cdecl
> > setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
> > (?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> > libgroupsock.lib(NetAddress.obj) : error LNK2019: unresolved external
> > symbol __imp__htons at 4 referenced in function "public: __thiscall
> > Port::Port(unsigned short)" (??0Port@@QAE at G@Z)
> > libgroupsock.lib(inet.obj ) : error LNK2019: unresolved external symbol
> > __imp__inet_addr at 4 referenced in function _our_inet_addr
> > libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
> > __imp__inet_ntoa at 4 referenced in function _our_inet_ntoa
> > libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
> > __imp__WSACleanup at 0 referenced in function _initializeWinsockIfNecessary
> > libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
> > __imp__WSAStartup at 8 referenced in function _initializeWinsockIfNecessary
> > libgroupsock.lib(inet.obj) : error LNK2019: unresolved external symbol
> > __imp__gethostbyname at 4 referenced in function _our_gethostbyname
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__gethostbyname at 4 referenced in function "int __cdecl
> > setupStreamSocket(class UsageEnvironment &,class Port,unsigned int)"
> > (?setupStreamSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__bind at 12 referenced in function "int __cdecl
> > setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
> > (?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__setsockopt at 20 referenced in function "int __cdecl
> > setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
> > (?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__socket at 12 referenced in function "int __cdecl
> > setupDatagramSocket(class UsageEnvironment &,class Port,unsigned int)"
> > (?setupDatagramSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> > libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2019:
> > unresolved external symbol __imp__socket at 12 referenced in function _$E8
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__ioctlsocket at 12 referenced in function "int __cdecl
> > setupStreamSocket(class UsageEnvironment &,class Port,unsigned int)"
> > (?setupStreamSocket@@YAHAAVUsageEnvironment@@VPort@@ I at Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__recvfrom at 24 referenced in function "int __cdecl
> > readSocket(class UsageEnvironment &,int,unsigned char *,unsigned int,struct
> > sockaddr_in &,struct timeval *)"
> > (?readSocket@@YAHAAVUsageEnvironment@@HPAEIAAUsockaddr_in@@PAUtimeval@@@Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol ___WSAFDIsSet at 8 referenced in function "int __cdecl
> > blockUntilReadable(class UsageEnvironment &,int,struct timeval *)"
> > (?blockUntilReadable@@YAHAAVUsageEnvironment@@HPAUtimeval@@@Z)
> > libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2019:
> > unresolved external symbol ___WSAFDIsSet at 8 referenced in function "public:
> > virtual void * __thiscall BasicTaskScheduler::`scalar deleting
> > destructor'(unsigned int)" (??_GBasicTaskScheduler@@ UAEPAXI at Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__select at 20 referenced in function "int __cdecl
> > blockUntilReadable(class UsageEnvironment &,int,struct timeval *)"
> > (?blockUntilReadable@@YAHAAVUsageEnvironment@@HPAUtimeval@@@Z)
> > libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2001:
> > unresolved external symbol __imp__select at 20
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__sendto at 24 referenced in function "unsigned int __cdecl
> > writeSocket(class UsageEnvironment &,int,struct in_addr,class Port,unsigned
> > char,unsigned char *,unsigned int)"
> > (?writeSocket@@YAIAAVUsageEnvironment@@HUin_addr@@VPort@@ EPAEI at Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__getsockopt at 20 referenced in function "unsigned int __cdecl
> > getBufferSize(class UsageEnvironment &,int,int)"
> > (?getBufferSize@@YAIAAVUsageEnvironment@@ HH at Z)
> > libgroupsock.lib(GroupsockHelper.obj) : error LNK2019: unresolved external
> > symbol __imp__getsockname at 12 referenced in function "unsigned int __cdecl
> > getSourcePort0(int,unsigned short &)" (?getSourcePort0@@ YAIHAAG at Z)
> > libBasicUsageEnvironment.lib(BasicUsageEnvironment.obj) : error LNK2019:
> > unresolved external symbol __imp__WSAGetLastError at 0 referenced in function
> > "public: virtual int __thiscall BasicUsageEnvironment::getErrno(void)const "
> > (?getErrno at BasicUsageEnvironment@@UBEHXZ)
> > libBasicUsageEnvironment.lib(BasicTaskScheduler.obj) : error LNK2019:
> > unresolved external symbol __imp__WSAGetLastError at 0 referenced in function
> > "protected: virtual void __thiscall BasicTaskScheduler::SingleStep(unsigned
> > int)" (?SingleStep at BasicTaskScheduler@@ MAEXI at Z)
> > Debug/NewLiveProject.exe : fatal error LNK1120: 25 unresolved externals
> >
> > Build log was saved at "file://c:\Documents and
> > Settings\support\Desktop\Richards
> > Live\live\NewLiveProject\NewLiveProject\Debug\BuildLog.htm"
> > NewLiveProject - 44 error(s), 1 warning(s)
> >
> >
> > ---------------------- Done ----------------------
> >
> > Build: 0 succeeded, 1 failed, 0 skipped
> >
> >
> >
> >
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: http://lists.live555.com/pipermail/live-devel/attachments/20060602/9ecd22ee/attachment-0001.html
>
> ------------------------------
>
> Message: 2
> Date: Fri, 2 Jun 2006 15:54:17 -0400
> From: "S Mathur"
> Subject: [Live-devel] RTSPclient and DarwinInjector in the same
> application
> To: live-devel at ns.live555.com
> Message-ID:
>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hello,
>
> I have modified the openRTSP/playCommon.cpp code such that it
> intercepts the M4A-LATM subsession in my streaming session and feeds
> it to a decoder filter. The decoder filter feeds data to an encoder
> filter which in turns feeds an MPEG4LATMRTPSink.
>
> The subsession related code is as follows...
> subsession->sink = latmSink;
> subsession->sink->startPlaying(*aacEncFilter, subsessionAfterPlaying,
> subsession);
> ....
>
> My latmSink is also associated with a DarwinInjector:
> injector->addStream(latmSink, audioRTCP);
> injector->setDestination(..)
>
>
> My problem: A get a select() failed error in
> BasicTaskScheduler::SingleStep() because RTCPInstance's destructor
> gets invoked prematurely. The ANNOUNCE (of the modified stream named
> liveST.sdp) to the Darwin server does goes through successfully.
>
> Here is the transcript, error is at the bottom:
>
> Received ANNOUNCE response: RTSP/1.0 200 OK
> Server: DSS/5.6.0 (Build/496; Platform/Win32; Release/Development;
> BuildNote/LP64 Support)
> Cseq: 4
>
> RTCPInstance[0x86cd5b8]::RTCPInstance()
> schedule(1.786040->1149276500.435798)
> Sending request: SETUP rtsp://myDSS:554/liveST.sdp/trackID=1 RTSP/1.0
> CSeq: 5
> Transport: RTP/AVP/TCP;unicast;mode=receive;interleaved=0-1
> User-Agent: Injector (LIVE555 Streaming Media v2006.03.16)
>
>
> Received SETUP response: RTSP/1.0 200 OK
> Server: DSS/5.6.0 (Build/496; Platform/Win32; Release/Development;
> BuildNote/LP64 Support)
> Cseq: 5
> Cache-Control: no-cache
> Session: 14018773254377
> Date: Fri, 02 Jun 2006 19:28:29 GMT
> Expires: Fri, 02 Jun 2006 19:28:29 GMT
> Transport: RTP/AVP/TCP;unicast;mode=receive;interleaved=0-1
>
>
> Sending request: PLAY rtsp://myDSS:554/liveST.sdp RTSP/1.0
> CSeq: 6
> Session: 14018773254377
> Range: npt=0.000-
> User-Agent: Injector (LIVE555 Streaming Media v2006.03.16)
>
>
> Received PLAY response: RTSP/1.0 200 OK
> Server: DSS/5.6.0 (Build/496; Platform/Win32; Release/Development;
> BuildNote/LP64 Support)
> Cseq: 6
> Session: 14018773254377
> RTP-Info: url=rtsp://myDSS:554/liveST.sdp/trackID=1
>
>
> RTCPInstance[0x86cd5b8]::~RTCPInstance()
> sending BYE
> sending RTCP packet
> 80c90001 c923bac9 81cb0001 c923bac9
> sendRTPOverTCP: 16 bytes over channel 1 (socket 8)
> sendRTPOverTCP: completed
> Play this stream (from the Darwin Streaming Server) using the URL:
> rtsp://myDSS/liveST.sdp
>
> [startPlaying(latmSink) gets called here. Greenday64kbpsPlus.3gp is
> the input stream to be transcoded]
> Sending request: PLAY rtsp://myDSS:554/Greenday64kbpsPlus.3gp RTSP/1.0
> CSeq: 7
> Session: 47446503751336
> Range: npt=0.000-
> User-Agent: ./testProgram (LIVE555 Streaming Media v2006.03.16)
>
>
> Received PLAY response: RTSP/1.0 200 OK
> Server: DSS/5.6.0 (Build/496; Platform/Win32; Release/Development;
> BuildNote/LP64 Support)
> Cseq: 7
> Session: 47446503751336
> Range: npt=0.00000-285.83667
> RTP-Info: url=rtsp://myDSS:554/Greenday64kbpsPlus.3gp/trackID=65536;seq=15921;rtptime=30605
>
> Started playing session
> Receiving streamed data (for up to 290.836670 seconds)...
> BasicTaskScheduler::SingleStep(): select() fails: Bad file descriptor
>
>
> Any suggestions on what I am doing wrong?
> Thanks,
> Mathur
>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 5 Jun 2006 10:13:56 +0530
> From: "Kushal Dalal"
> Subject: [Live-devel] Global Macro
> To:
> Message-ID: <200606050444.k554i31S002769 at ns.live555.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi All
>
>
>
> I want to define a global macro which can be used in all files of live.
>
> Where (in which header file/files) should I define it?
>
>
>
> Thanks in advance.
>
> Kushal
>
> -------------- next part --------------
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>
> ------------------------------
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
> End of live-devel Digest, Vol 32, Issue 3
> *****************************************
>
From barounis at ceid.upatras.gr Tue Jun 6 07:52:10 2006
From: barounis at ceid.upatras.gr (barounis at ceid.upatras.gr)
Date: Tue, 6 Jun 2006 17:52:10 +0300
Subject: [Live-devel] stopPlaying()
In-Reply-To: <7.0.1.0.1.20060524195000.01ec82d8@live555.com>
References: <1143586422.4429be76673a7@my.ceid.upatras.gr>
<7.0.1.0.1.20060328150729.01f93568@live555.com>
<1144792970.443c278a9bc41@my.ceid.upatras.gr>
<7.0.1.0.1.20060411151024.01d638a0@live555.com>
<1146055961.444f6d19519e3@my.ceid.upatras.gr>
<1146132663.445098b73a8d8@my.ceid.upatras.gr>
<7.0.1.0.1.20060427053054.01f51830@live555.com>
<1146146329.4450ce19cb6d9@my.ceid.upatras.gr>
<7.0.1.0.1.20060427073327.01f7ed50@live555.com>
<1146923099.445ca85b050bc@my.ceid.upatras.gr>
<7.0.1.0.1.20060506224156.01f868d8@live555.com>
<1147875340.446b300cb6f38@my.ceid.upatras.gr>
<1148308140.4471caacbc138@my.ceid.upatras.gr>
<1148508296.4474d888245e1@my.ceid.upatras.gr>
<7.0.1.0.1.20060524195000.01ec82d8@live555.com>
Message-ID: <1149605530.4485969a331ed@my.ceid.upatras.gr>
?????? ?????? ??? Ross Finlayson :
>
> >I would like to have the ability to stop and then play another file
> >during the
> >same session RTSPServer from the RTSP server side. This means that I have
> to
> >call the MediaSink::stopPlaying or the destructor ~Mediasink().
>
> Just the former. Because you want to keep the RTP session intact,
> you should *not* close the (RTP)Sink object.
>
> To change the input source for a running stream, you should do the
> following, in order:
>
> sink->stopPlaying();
> Medium::close(oldSource);
> create newSource
> sink->startPlaying(newSource, ...);
Hello,
I am working on the RTPTransmissionStats::noteIncomingRR() method and after
having implemented the "sink->stopPlaying()" I am looking for a way to implement
"Medium::close(oldSource)".
Having problem with the "Medium* medium" pointer that close() takes as argument
can you please tell me the name of the pointer that I should use? I have also
noticed that the close() function is called many times during the streaming and
I don't know the proper pointer for the object that must be close()ed.
Thank you very much
Kostas
----------------------------------------------------
This mail was sent through http://my.ceid.upatras.gr
From p.pradeep09 at gmail.com Tue Jun 6 23:44:34 2006
From: p.pradeep09 at gmail.com (Pradeep Kumar PALAPARTHY)
Date: Wed, 7 Jun 2006 12:14:34 +0530
Subject: [Live-devel] Error while streaming using live 555 test programs
Message-ID: <8daef9c70606062344k5f9a2ef1paf3add96a2e2cd42@mail.gmail.com>
Dear all,
Im running onDemandRTSPServer for rtsp streaming.
I have written some sample programs to connect the rtsp server using
mp4player rtsp liraries.
When i send two SETUP(rstp command) requests for both audio & video, the
session is getting increased in response to the second setup request.
Following are my requests * responses
*1Request*
SETUP rtsp://192.168.1.196:8554/mpeg1or2AudioVideoTest/track1 RTSP/1.0
CSeq: 1
Transport: RTP/AVP;unicast;client_port:1024-1025
User-Agent: STL Version 1.0
*Response is*
RTSP 200 OK
CSeq: 1
Transport: RTP/AVP;unicast;destination:192.168.1.196
;client_port:1024-1025;server_port:32780-32780
Session: *32*
User-Agent: STL Version 1.0
*2Request*
**
SETUP rtsp://192.168.1.196:8554/mpeg1or2AudioVideoTest/track2 RTSP/1.0
CSeq: 1
Session: *32*
Transport: RTP/AVP;unicast;client_port:1026-1027
User-Agent: STL Version 1.0
Response is
RTSP 200 OK
CSeq: 1
Transport: RTP/AVP;unicast;destination:192.168.1.196
;client_port:1024-1025;server_port:32780-32780
Session: *33*
User-Agent: STL Version 1.0
The problem is that is is treating the second request as new session...i
suppose.
Please any can help me in this regard.
Thanks in Advance
Regards,
Pradeep
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From tchristensen at nordija.com Wed Jun 7 00:04:41 2006
From: tchristensen at nordija.com (Thomas Christensen)
Date: Wed, 7 Jun 2006 09:04:41 +0200
Subject: [Live-devel] Error while streaming using live 555 test programs
In-Reply-To: <8daef9c70606062344k5f9a2ef1paf3add96a2e2cd42@mail.gmail.com>
References: <8daef9c70606062344k5f9a2ef1paf3add96a2e2cd42@mail.gmail.com>
Message-ID: <01C8F661-4FDA-4A74-8F4E-CD1103795F7E@nordija.com>
Hi Pradeep
Right off the bat: Shouldn't your CSeq be increasing? Haven't checked
with the source code, but it could be that the RTSP server figures
that CSeqs should be increasing if the same session should be reused.
Ie: Second request should have CSeq:2
Cheers
Thomas
On 07/06/2006, at 8.44, Pradeep Kumar PALAPARTHY wrote:
> Dear all,
>
> Im running onDemandRTSPServer for rtsp streaming.
> I have written some sample programs to connect the rtsp server
> using mp4player rtsp liraries.
> When i send two SETUP(rstp command) requests for both audio &
> video, the session is getting increased in response to the second
> setup request.
>
> Following are my requests * responses
>
> 1Request
>
> SETUP rtsp://192.168.1.196:8554/mpeg1or2AudioVideoTest/track1 RTSP/1.0
> CSeq: 1
> Transport: RTP/AVP;unicast;client_port:1024-1025
> User-Agent: STL Version 1.0
>
> Response is
>
> RTSP 200 OK
> CSeq: 1
> Transport: RTP/AVP;unicast;destination:192.168.1.196;client_port:
> 1024-1025;server_port:32780-32780
> Session: 32
> User-Agent: STL Version 1.0
>
> 2Request
>
> SETUP rtsp://192.168.1.196:8554/mpeg1or2AudioVideoTest/track2 RTSP/1.0
> CSeq: 1
> Session: 32
> Transport: RTP/AVP;unicast;client_port:1026-1027
> User-Agent: STL Version 1.0
>
> Response is
>
> RTSP 200 OK
> CSeq: 1
> Transport: RTP/AVP;unicast;destination:192.168.1.196;client_port:
> 1024-1025;server_port:32780-32780
> Session: 33
> User-Agent: STL Version 1.0
>
>
> The problem is that is is treating the second request as new
> session...i suppose.
> Please any can help me in this regard.
>
> Thanks in Advance
>
> Regards,
> Pradeep
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
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From finlayson at live555.com Wed Jun 7 00:31:59 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 07 Jun 2006 00:31:59 -0700
Subject: [Live-devel] Error while streaming using live 555 test programs
In-Reply-To: <8daef9c70606062344k5f9a2ef1paf3add96a2e2cd42@mail.gmail.co
m>
References: <8daef9c70606062344k5f9a2ef1paf3add96a2e2cd42@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060607002012.01fd6b10@live555.com>
>Im running onDemandRTSPServer for rtsp streaming.
>I have written some sample programs to connect the rtsp server using
>mp4player rtsp liraries.
>When i send two SETUP(rstp command) requests for both audio & video,
>the session is getting increased in response to the second setup request.
>
>Following are my requests * responses
>
>1Request
>
>SETUP rtsp://192.168.1.196:8554/mpeg1or2AudioVideoTest/track1 RTSP/1.0
>CSeq: 1
>Transport: RTP/AVP;unicast;client_port:1024-1025
>User-Agent: STL Version 1.0
>
>Response is
>
>RTSP 200 OK
>CSeq: 1
>Transport:
>RTP/AVP;unicast;destination:192.168.1.196;client_port:1024-1025;server_port:32780-32780
>Session: 32
>User-Agent: STL Version 1.0
Bzzt! Wrong! Our RTSP server implementation does not return a
"User-Agent" string in its responses. You've been busted :-) -
either you didn't record the protocol exchange properly, or else you
modified our RTSP server implementation (in which case you can't
expect much support).
If you want to record the real RTSP protocol exchange by our RTSP
server implementation, then add
#define DEBUG 1
to near the start of the file "RTSPServer.cpp".
And if you want to see an example of a RTSP client that correctly
plays the "mpeg1or2AudioVideoTest" stream, then run "openRTSP", with
the "-V" option (see ).
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From p.pradeep09 at gmail.com Wed Jun 7 02:59:37 2006
From: p.pradeep09 at gmail.com (Pradeep Kumar PALAPARTHY)
Date: Wed, 7 Jun 2006 15:29:37 +0530
Subject: [Live-devel] Error while streaming using live 555 test programs
In-Reply-To: <01C8F661-4FDA-4A74-8F4E-CD1103795F7E@nordija.com>
References: <8daef9c70606062344k5f9a2ef1paf3add96a2e2cd42@mail.gmail.com>
<01C8F661-4FDA-4A74-8F4E-CD1103795F7E@nordija.com>
Message-ID: <8daef9c70606070259p76d6d041t5a19e4bc9bbf9c7b@mail.gmail.com>
Hi ,
Actually its a typo error sicne i have copied ....Im sending incremented
CSeq.
Regards,
Pradeep
On 6/7/06, Thomas Christensen wrote:
>
> Hi Pradeep
>
> Right off the bat: Shouldn't your CSeq be increasing? Haven't checked with
> the source code, but it could be that the RTSP server figures that CSeqs
> should be increasing if the same session should be reused.
>
>
> Ie: Second request should have *CSeq:2*
> *
> *
> Cheers
>
>
> Thomas
>
> On 07/06/2006, at 8.44, Pradeep Kumar PALAPARTHY wrote:
>
> Dear all,
>
> Im running onDemandRTSPServer for rtsp streaming.
> I have written some sample programs to connect the rtsp server using
> mp4player rtsp liraries.
> When i send two SETUP(rstp command) requests for both audio & video, the
> session is getting increased in response to the second setup request.
>
> Following are my requests * responses
>
> *1Request*
>
> SETUP rtsp://192.168.1.196:8554/mpeg1or2AudioVideoTest/track1 RTSP/1.0
> CSeq: 1
> Transport: RTP/AVP;unicast;client_port:1024-1025
> User-Agent: STL Version 1.0
>
> *Response is*
>
> RTSP 200 OK
> CSeq: 1
> Transport: RTP/AVP;unicast;destination:192.168.1.196
> ;client_port:1024-1025;server_port:32780-32780
> Session: *32*
> User-Agent: STL Version 1.0
>
> *2Request*
> **
> SETUP rtsp://192.168.1.196:8554/mpeg1or2AudioVideoTest/track2 RTSP/1.0
> CSeq: 1
> Session: *32*
> Transport: RTP/AVP;unicast;client_port:1026-1027
> User-Agent: STL Version 1.0
>
> Response is
>
> RTSP 200 OK
> CSeq: 1
> Transport: RTP/AVP;unicast;destination:192.168.1.196
> ;client_port:1024-1025;server_port:32780-32780
> Session: *33*
> User-Agent: STL Version 1.0
>
>
> The problem is that is is treating the second request as new session...i
> suppose.
> Please any can help me in this regard.
>
> Thanks in Advance
>
> Regards,
> Pradeep
>
> _______________________________________________
> 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
>
>
>
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From richard.monette at gmail.com Wed Jun 7 11:00:57 2006
From: richard.monette at gmail.com (Richard Monette)
Date: Wed, 7 Jun 2006 14:00:57 -0400
Subject: [Live-devel] Network Problem
Message-ID:
I can run the testOnDemandRTSPServer fine and I am am able to receive the
video on the same machine as the server but I cannot get the video on any
other machines on the network.
I do not beleive I have any firewall issues.
Any ideas? Are there any tests to check if its a network issue or something
to do with the server?
Rick
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From finlayson at live555.com Wed Jun 7 12:03:06 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 07 Jun 2006 12:03:06 -0700
Subject: [Live-devel] Network Problem
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060607120131.01fd6b10@live555.com>
At 11:00 AM 6/7/2006, you wrote:
>I can run the testOnDemandRTSPServer fine and I am am able to
>receive the video on the same machine as the server but I cannot get
>the video on any other machines on the network.
>
>I do not beleive I have any firewall issues.
>
>Any ideas?
No, not really.
> Are there any tests to check if its a network issue or something
> to do with the server?
Try running "openRTSP -V" on the
other machines. This may tell you whether (and if not, why not) RTP
data packets are arriving there from the server.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From richard.monette at gmail.com Wed Jun 7 12:17:35 2006
From: richard.monette at gmail.com (Richard Monette)
Date: Wed, 7 Jun 2006 15:17:35 -0400
Subject: [Live-devel] Network Problem
In-Reply-To: <7.0.1.0.1.20060607120131.01fd6b10@live555.com>
References:
<7.0.1.0.1.20060607120131.01fd6b10@live555.com>
Message-ID:
I ran the openRTSP example using -V and it works fine. Strange, must be
something to do with the QuickTime player I am using. Another quick question
- I see in the openRTSP example it is possible to get the QoS metrics
reported. Is it possible to turn these on within the testOnDemandRTSPServer?
On a completely different tact, I have extensive experience working with PHP
and MySQL . Would it be possible to work together on making the archives
searchable?
Thanks much.
Rick
On 6/7/06, Ross Finlayson wrote:
>
> At 11:00 AM 6/7/2006, you wrote:
> >I can run the testOnDemandRTSPServer fine and I am am able to
> >receive the video on the same machine as the server but I cannot get
> >the video on any other machines on the network.
> >
> >I do not beleive I have any firewall issues.
> >
> >Any ideas?
>
> No, not really.
>
> > Are there any tests to check if its a network issue or something
> > to do with the server?
>
> Try running "openRTSP -V" on the
> other machines. This may tell you whether (and if not, why not) RTP
> data packets are arriving there from the server.
>
>
> 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
>
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From lorenooliveira at gmail.com Wed Jun 7 12:19:49 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Wed, 7 Jun 2006 16:19:49 -0300
Subject: [Live-devel] sample applications
Message-ID:
Where do I download the examples without links (e.g. *
testMPEG1or2VideoStreamer*, *testMPEG1or2VideoReceiver*, and *
testMPEG4VideoStreamer*) refered in the live site?
Loreno
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From finlayson at live555.com Wed Jun 7 12:20:47 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 07 Jun 2006 12:20:47 -0700
Subject: [Live-devel] Network Problem
Message-ID: <7.0.1.0.1.20060607122013.01fc4ae0@live555.com>
At 11:00 AM 6/7/2006, you wrote:
>I can run the testOnDemandRTSPServer fine and I am am able to
>receive the video on the same machine as the server but I cannot get
>the video on any other machines on the network.
>
>I do not beleive I have any firewall issues.
>
>Any ideas?
Are these client machines running Windows XP? If so, then perhaps
their firewall is blocking RTP?
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From lorenooliveira at gmail.com Wed Jun 7 12:21:51 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Wed, 7 Jun 2006 16:21:51 -0300
Subject: [Live-devel] sample applications
In-Reply-To:
References:
Message-ID:
OPS, I?ve just found it :-)
On 6/7/06, Loreno Oliveira wrote:
>
> Where do I download the examples without links (e.g. *
> testMPEG1or2VideoStreamer*, *testMPEG1or2VideoReceiver*, and *testMPEG4VideoStreamer
> *) refered in the live site?
>
> Loreno
>
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From finlayson at live555.com Wed Jun 7 12:37:02 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 07 Jun 2006 12:37:02 -0700
Subject: [Live-devel] Network Problem
In-Reply-To:
References:
<7.0.1.0.1.20060607120131.01fd6b10@live555.com>
Message-ID: <7.0.1.0.1.20060607123359.020147e8@live555.com>
At 12:17 PM 6/7/2006, you wrote:
>I ran the openRTSP example using -V and it works fine. Strange, must
>be something to do with the QuickTime player I am using. Another
>quick question - I see in the openRTSP example it is possible to get
>the QoS metrics reported. Is it possible to turn these on within the
>testOnDemandRTSPServer?
These are just reported by the client, and don't need any special
server support. You'll get them automatically, provided that you are
receiving RTCP.
>On a completely different tact, I have extensive experience working
>with PHP and MySQL . Would it be possible to work together on making
>the archives searchable?
Possibly, although note that the archives are generated by (and
accessed through) "mailman", so you should look at whatever
approaches other mailman sites have used to search their archives.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Wed Jun 7 12:38:28 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 07 Jun 2006 12:38:28 -0700
Subject: [Live-devel] sample applications
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060607123719.01fd6b10@live555.com>
At 12:19 PM 6/7/2006, you wrote:
>Where do I download the examples without links (e.g.
>testMPEG1or2VideoStreamer, testMPEG1or2VideoReceiver, and
>testMPEG4VideoStreamer) refered in the live site?
I'm not sure I understand your question. Those applications are demo
applications, and are built automatically - in the
"testProgs" directory - when you build the "LIVE555 Streaming Media" code.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From sdhays.neon.com.tw at gmail.com Wed Jun 7 19:10:26 2006
From: sdhays.neon.com.tw at gmail.com (Scott Hays)
Date: Thu, 8 Jun 2006 10:10:26 +0800
Subject: [Live-devel] Extraneous object links on Linux
Message-ID: <9866ce4f0606071910l211be02cq10fa9b1878b8b8a@mail.gmail.com>
I've just discovered that the size of the test programs on Linux is much
larger than the size of the same test programs compiled on Windows. For
example, my Windows compiled version of testMPEG4VideoStreamer is
Furthermore, the source of the size difference seems to be that the Linux
(GNU) linker is pulling a lot of useless object files from the Live555
libraries. For example, doing `nm -C testMPEG4VideoStreamer | grep MP3`
results in a screenfull of symbols for MP3ADURTPSink, MP3FileSource, etc.
Has anyone else noticed this, and, if so, how did you work around it? I'm
developing for embedded systems and this is adding over 200K to the size of
my program (very valuable space). Someone who's helping me reduced the size
of the program by manually commenting out large portions of the Makefiles
and then recompiling the libraries, but since I try to keep up-to-date with
the Live555 libraries during my development, this seems tedious and
error-prone.
Scott
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From finlayson at live555.com Wed Jun 7 23:11:45 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 07 Jun 2006 23:11:45 -0700
Subject: [Live-devel] Extraneous object links on Linux
In-Reply-To: <9866ce4f0606071910l211be02cq10fa9b1878b8b8a@mail.gmail.com
>
References: <9866ce4f0606071910l211be02cq10fa9b1878b8b8a@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060607230908.01fa5968@live555.com>
At 07:10 PM 6/7/2006, you wrote:
>I've just discovered that the size of the test programs on Linux is
>much larger than the size of the same test programs compiled on
>Windows. For example, my Windows compiled version of
>testMPEG4VideoStreamer is Furthermore, the source of the size
>difference seems to be that the Linux (GNU) linker is pulling a lot
>of useless object files from the Live555 libraries. For example,
>doing `nm -C testMPEG4VideoStreamer | grep MP3` results in a
>screenfull of symbols for MP3ADURTPSink, MP3FileSource, etc.
>
>Has anyone else noticed this, and, if so, how did you work around it?
As you've noted, this is an issue with the GNU linker. Perhaps newer
versions of the linker do a better job; are you using an up-to-date
version? Alternatively, perhaps there's some flag you can give the
linker... (I dunno - I'm not an expert on the GNU toolset.)
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From richard.monette at gmail.com Thu Jun 8 06:28:51 2006
From: richard.monette at gmail.com (Richard Monette)
Date: Thu, 8 Jun 2006 09:28:51 -0400
Subject: [Live-devel] Network Problem
In-Reply-To: <7.0.1.0.1.20060607123359.020147e8@live555.com>
References:
<7.0.1.0.1.20060607120131.01fd6b10@live555.com>
<7.0.1.0.1.20060607123359.020147e8@live555.com>
Message-ID:
On 6/7/06, Ross Finlayson wrote:
>
> At 12:17 PM 6/7/2006, you wrote:
> >I ran the openRTSP example using -V and it works fine. Strange, must
> >be something to do with the QuickTime player I am using. Another
> >quick question - I see in the openRTSP example it is possible to get
> >the QoS metrics reported. Is it possible to turn these on within the
> >testOnDemandRTSPServer?
>
> These are just reported by the client, and don't need any special
> server support. You'll get them automatically, provided that you are
> receiving RTCP.
>
> >On a completely different tact, I have extensive experience working
> >with PHP and MySQL . Would it be possible to work together on making
> >the archives searchable?
>
> Possibly, although note that the archives are generated by (and
> accessed through) "mailman", so you should look at whatever
> approaches other mailman sites have used to search their archives.
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
Seems like the easiest way to search the archives is just to use Google and
define the site you want to search within. For example,
your keywords site:http://lists.live555.com/pipermail/live-devel/
Seems to work alright. You could probably prevent a lot of 're-posting' by
letting people know such a search is possible.
I am using XP, Ill take a look at the software firewall settings today.
Richard E. Monette
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From barounis at ceid.upatras.gr Thu Jun 8 07:36:52 2006
From: barounis at ceid.upatras.gr (barounis at ceid.upatras.gr)
Date: Thu, 8 Jun 2006 17:36:52 +0300
Subject: [Live-devel] Network Problem
In-Reply-To:
References:
<7.0.1.0.1.20060607120131.01fd6b10@live555.com>
<7.0.1.0.1.20060607123359.020147e8@live555.com>
Message-ID: <1149777412.4488360445b67@my.ceid.upatras.gr>
> Seems like the easiest way to search the archives is just to use Google and
> define the site you want to search within. For example,
>
> your keywords site:http://lists.live555.com/pipermail/live-devel/
>
> Seems to work alright. You could probably prevent a lot of 're-posting' by
> letting people know such a search is possible.
Hello,
If you are talking about ways for searching the list, I have found this site
http://dir.gmane.org/gmane.comp.video.livedotcom.devel
The archive for this list can be read the following ways:
* On the web, using frames and threads.
* On the web, using a blog-like, flat interface.
You can also search the list by filling the form at the bottom of the page.
The only error according to this site is that the list's url is "www.live.com"
although it should be live555.
Hope I helped
Kostas
----------------------------------------------------
This mail was sent through http://my.ceid.upatras.gr
From braymond at echostorm.net Thu Jun 8 16:45:40 2006
From: braymond at echostorm.net (Brian Raymond)
Date: Thu, 8 Jun 2006 19:45:40 -0400
Subject: [Live-devel] MPEG2-TS demux
Message-ID: <141a20cd955a6f19d00b2ad7d198f34f@echostorm.net>
I have been playing around with Live555 as an RTSP server and it's
worked out good however I'm having some trouble understanding how I go
about receiving a MPEG2-TS over UDP and demux the ES for processing. I
want to dump the ES to different fifos for external processing (one
video, one metadata) however I can't get a handle on how I might go
about it. Can someone provide me a pointer regarding where I might
look?
Thanks.
From finlayson at live555.com Thu Jun 8 17:02:38 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 08 Jun 2006 17:02:38 -0700
Subject: [Live-devel] MPEG2-TS demux
In-Reply-To: <141a20cd955a6f19d00b2ad7d198f34f@echostorm.net>
References: <141a20cd955a6f19d00b2ad7d198f34f@echostorm.net>
Message-ID: <7.0.1.0.1.20060608165641.01fa3a70@live555.com>
At 04:45 PM 6/8/2006, you wrote:
>I have been playing around with Live555 as an RTSP server and it's
>worked out good however I'm having some trouble understanding how I go
>about receiving a MPEG2-TS over UDP and demux the ES for processing.
I'm a bit puzzled by your question - it's not clear to me exactly
what you want to do.
Do you want to
1/ write a new RTSP client that receives MPEG Transport Stream data
from an existing LIVE555 (or other) RTSP server, or:
2/ write a RTSP server that will serve MPEG Transport Stream data to
some existing RTSP client that happens to ask for raw UDP (rather
than the standard RTP/RTCP)?
If you want to do 1/, then you can just run "openRTSP -v": see
. This will output, to stdout, the
received MPEG Transport Stream data. You can then pipe this to some
separate application that demultiplexes the Transport Stream
data. (Note that the "LIVE555 Streaming Media" libraries currently
don't have any direct support for demultiplexiing Transport Stream data.)
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From braymond at echostorm.net Thu Jun 8 17:13:22 2006
From: braymond at echostorm.net (Brian Raymond)
Date: Thu, 8 Jun 2006 20:13:22 -0400
Subject: [Live-devel] MPEG2-TS demux
In-Reply-To: <7.0.1.0.1.20060608165641.01fa3a70@live555.com>
References: <141a20cd955a6f19d00b2ad7d198f34f@echostorm.net>
<7.0.1.0.1.20060608165641.01fa3a70@live555.com>
Message-ID: <0837870fa2cf793a58b7b097cc820521@echostorm.net>
I was actually looking for something that will demux the transport
stream and stream one pid and drop the other on a pipe however based on
your comment I see that it isn't supported. I saw some mention of
transport streams in the context of live555 so I was wondering if that
was possible.
Thanks for the information.
On Jun 8, 2006, at 8:02 PM, Ross Finlayson wrote:
> At 04:45 PM 6/8/2006, you wrote:
>> I have been playing around with Live555 as an RTSP server and it's
>> worked out good however I'm having some trouble understanding how I go
>> about receiving a MPEG2-TS over UDP and demux the ES for processing.
>
> I'm a bit puzzled by your question - it's not clear to me exactly
> what you want to do.
>
> Do you want to
> 1/ write a new RTSP client that receives MPEG Transport Stream data
> from an existing LIVE555 (or other) RTSP server, or:
> 2/ write a RTSP server that will serve MPEG Transport Stream data to
> some existing RTSP client that happens to ask for raw UDP (rather
> than the standard RTP/RTCP)?
>
> If you want to do 1/, then you can just run "openRTSP -v": see
> . This will output, to stdout, the
> received MPEG Transport Stream data. You can then pipe this to some
> separate application that demultiplexes the Transport Stream
> data. (Note that the "LIVE555 Streaming Media" libraries currently
> don't have any direct support for demultiplexiing Transport Stream
> data.)
>
>
> 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
>
>
From sanandchoubey at yahoo.com Fri Jun 9 03:04:33 2006
From: sanandchoubey at yahoo.com (Anand Choubey)
Date: Fri, 9 Jun 2006 03:04:33 -0700 (PDT)
Subject: [Live-devel] WIndows Medai Player Problem
Message-ID: <20060609100433.80521.qmail@web50812.mail.yahoo.com>
Hi
I am new in streaming world. I downloaded live source
code and built binaries.
I run code with testWAVAudioStreamer.cpp test program.
RTSP cliens are MPlayer, VLC and windows media
problem.
In MPlayer and VLC play destroyed voice.
But Windows media player is not able to locate live
rtsp server.
Thanks
Anand Choubey
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
From vlcplayer at gmail.com Fri Jun 9 03:51:55 2006
From: vlcplayer at gmail.com (Lukasz Kondrad)
Date: Fri, 9 Jun 2006 13:51:55 +0300
Subject: [Live-devel] [live-devel] Two network adapters
Message-ID:
Hello,
When I have two network adapters with two different addresses, how does
live555 choose from which read rtp ? Can I somehow specify that?
BR
Lukasz
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From finlayson at live555.com Fri Jun 9 07:57:19 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 09 Jun 2006 07:57:19 -0700
Subject: [Live-devel] WIndows Medai Player Problem
In-Reply-To: <20060609100433.80521.qmail@web50812.mail.yahoo.com>
References: <20060609100433.80521.qmail@web50812.mail.yahoo.com>
Message-ID: <7.0.1.0.1.20060609075202.01feef18@live555.com>
>I am new in streaming world. I downloaded live source
>code and built binaries.
>I run code with testWAVAudioStreamer.cpp test program.
>RTSP cliens are MPlayer, VLC and windows media
>problem.
>In MPlayer and VLC play destroyed voice.
>But Windows media player is not able to locate live
>rtsp server.
Windows Media Player is not standards-compliant; you cannot (in
general) expect it to play any standard RTSP/RTP stream.
MPlayer and (especially) VLC should work, however, so I'm surprised
to hear you say that it plays "destroyed voice".
Could you please also test using "testOnDemandRTSPServer" as your
server, and VLC as your client? If that also fails, then please post
a link to your "test.wav" file (or post the file itself if it's <
1MByte in size), and I'll take a look at it.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From sarpotdaramogh at yahoo.com Mon Jun 12 03:11:56 2006
From: sarpotdaramogh at yahoo.com (Amogh Sarpotdar)
Date: Mon, 12 Jun 2006 03:11:56 -0700 (PDT)
Subject: [Live-devel] Debugging Live555 library.
Message-ID: <20060612101156.83447.qmail@web55507.mail.re4.yahoo.com>
Hi,
I am trying to debug the Live555 library, but failing despite of taking all the steps mentioned in documentation.
I generated the make files with #nodebug comment removed. I am using Microsoft Visual C++ 6.0. I have Studio 7 (.net) present on the same machine but I have ensured that no studio 7 libraries are being included by accident.
I dont find the .pdb or .dbg file after building goupsock,livemedia etc.
Can anybody give me guidelines on this?
Thanks
Regards
Amogh S
__________________________________________________
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From kushal.dalal at einfochips.com Mon Jun 12 21:40:52 2006
From: kushal.dalal at einfochips.com (Kushal Dalal)
Date: Tue, 13 Jun 2006 10:10:52 +0530
Subject: [Live-devel] streaming method
Message-ID: <200606130441.k5D4fNcX071479@ns.live555.com>
Hi All
I am new to streaming. I have some basic questions.
I want to develop network camera.
My requirement is something like this.
I will capture live video and compress it using encoder.
That compressed live video should be streamed over network.
Client connected to camera should view live video using media player plugins
(VLC, Real, WMP) of web browser.
Which streaming method (RTP/RTSP, HTTP, or .) should be used for that?
Can I do that using Live?
Thanks in advance.
Kushal
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From sarpotdaramogh at yahoo.com Tue Jun 13 01:59:20 2006
From: sarpotdaramogh at yahoo.com (Amogh Sarpotdar)
Date: Tue, 13 Jun 2006 01:59:20 -0700 (PDT)
Subject: [Live-devel] RTSP Server code - need some help
In-Reply-To: <7.0.1.0.1.20060613014812.01f99c68@live555.com>
Message-ID: <20060613085920.26473.qmail@web55503.mail.re4.yahoo.com>
Hi
First of all, let me take this opportunity to thank Ross and
congratulate for this work. This library is really amazing.
Here is my query : The RTSP server in your code is listening on
8554 and serving requests on open connection. This server works on a
public IP. I am trying to serve media from desktops (behind firewall) to
other desktops (through a central computer - kind of p2p work). For this,
I will have to change the communication architecture. I will have to
make this RTSP server connect to central computer (my server where all
desktops register themselves) and send the data to it, from where it will
find its destination.
In short, I need to provide RTSP server functionality with the
client-oriented connection. Please guide me how can I do this. Please
tell me about the files/code that I should concentrate on to change.
Thanks in advance,
Regards
Amogh Sarpotdar
Telesoft, India
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From barounis at ceid.upatras.gr Tue Jun 13 03:30:52 2006
From: barounis at ceid.upatras.gr (barounis at ceid.upatras.gr)
Date: Tue, 13 Jun 2006 13:30:52 +0300
Subject: [Live-devel] stopPlaying()
In-Reply-To: <7.0.1.0.1.20060524195000.01ec82d8@live555.com>
References: <1143586422.4429be76673a7@my.ceid.upatras.gr>
<7.0.1.0.1.20060328150729.01f93568@live555.com>
<1144792970.443c278a9bc41@my.ceid.upatras.gr>
<7.0.1.0.1.20060411151024.01d638a0@live555.com>
<1146055961.444f6d19519e3@my.ceid.upatras.gr>
<1146132663.445098b73a8d8@my.ceid.upatras.gr>
<7.0.1.0.1.20060427053054.01f51830@live555.com>
<1146146329.4450ce19cb6d9@my.ceid.upatras.gr>
<7.0.1.0.1.20060427073327.01f7ed50@live555.com>
<1146923099.445ca85b050bc@my.ceid.upatras.gr>
<7.0.1.0.1.20060506224156.01f868d8@live555.com>
<1147875340.446b300cb6f38@my.ceid.upatras.gr>
<1148308140.4471caacbc138@my.ceid.upatras.gr>
<1148508296.4474d888245e1@my.ceid.upatras.gr>
<7.0.1.0.1.20060524195000.01ec82d8@live555.com>
Message-ID: <1150194652.448e93dc35886@my.ceid.upatras.gr>
>
> >I would like to have the ability to stop and then play another file
> >during the
> >same session RTSPServer from the RTSP server side. This means that I have
> to
> >call the MediaSink::stopPlaying or the destructor ~Mediasink().
>
> Just the former. Because you want to keep the RTP session intact,
> you should *not* close the (RTP)Sink object.
>
> To change the input source for a running stream, you should do the
> following, in order:
>
> sink->stopPlaying();
> Medium::close(oldSource);
> create newSource
> sink->startPlaying(newSource, ...);
Hello,
There is a basic question I would like to ask. Concerning the above and the
ability to stop and play another file, which is the best class (file of the
source code) where I should implement this code ?
Currently I am trying implementing this code in the
RTPTransmissionStats::noteIncomingRR() examining the RTCP report packets sent by
the client, but I'am afraid of having problems concerning the scope and the
ability of communication between classes and objects.
Would it be easy to try to implement the above code on the
RTPTransmissionStats::noteIncomingRR() function or this must be done somewhere
else ?
Thank you very much
Best regards
----------------------------------------------------
This mail was sent through http://my.ceid.upatras.gr
From finlayson at live555.com Tue Jun 13 12:26:09 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 13 Jun 2006 12:26:09 -0700
Subject: [Live-devel] stopPlaying()
In-Reply-To: <1150194652.448e93dc35886@my.ceid.upatras.gr>
References: <1143586422.4429be76673a7@my.ceid.upatras.gr>
<7.0.1.0.1.20060328150729.01f93568@live555.com>
<1144792970.443c278a9bc41@my.ceid.upatras.gr>
<7.0.1.0.1.20060411151024.01d638a0@live555.com>
<1146055961.444f6d19519e3@my.ceid.upatras.gr>
<1146132663.445098b73a8d8@my.ceid.upatras.gr>
<7.0.1.0.1.20060427053054.01f51830@live555.com>
<1146146329.4450ce19cb6d9@my.ceid.upatras.gr>
<7.0.1.0.1.20060427073327.01f7ed50@live555.com>
<1146923099.445ca85b050bc@my.ceid.upatras.gr>
<7.0.1.0.1.20060506224156.01f868d8@live555.com>
<1147875340.446b300cb6f38@my.ceid.upatras.gr>
<1148308140.4471caacbc138@my.ceid.upatras.gr>
<1148508296.4474d888245e1@my.ceid.upatras.gr>
<7.0.1.0.1.20060524195000.01ec82d8@live555.com>
<1150194652.448e93dc35886@my.ceid.upatras.gr>
Message-ID: <7.0.1.0.1.20060613122349.01f4ddd8@live555.com>
>There is a basic question I would like to ask. Concerning the above and the
>ability to stop and play another file, which is the best class (file of the
>source code) where I should implement this code ?
>
>Currently I am trying implementing this code in the
>RTPTransmissionStats::noteIncomingRR() examining the RTCP report
>packets sent by
>the client, but I'am afraid of having problems concerning the scope and the
>ability of communication between classes and objects.
>
>Would it be easy to try to implement the above code on the
>RTPTransmissionStats::noteIncomingRR() function or this must be done somewhere
>else ?
I don't recommend modifying the supplied library code, if you don't
have to (because that makes it difficult to upgrade to newer versions
of the library code).
In your case, I suggest using a separate "RTCP RR handler function"
that gets called each time a new RTCP "RR" packet arrives, and using
the "RTCPInstance::setRRHandler()" function to register it.
(See "liveMedia/include/RTCP.hh")
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From roy.subrato at gmail.com Wed Jun 14 04:32:43 2006
From: roy.subrato at gmail.com (subrato roy)
Date: Wed, 14 Jun 2006 14:32:43 +0300
Subject: [Live-devel] increase of buffer size
Message-ID: <64849090606140432n6a987b6fhba48f02b45a26708@mail.gmail.com>
Hello ,
I am working with the framedSource::getNextFrame(). there is a need
for me to get as large a read as possible and fill it up in the fTo
bufer. Can someone tell me how to increase the buffer size for this.
Also i would like to know how large a buffer can be done in this way .
thanks for all the help.
subrato
From kushal.dalal at einfochips.com Wed Jun 14 04:41:20 2006
From: kushal.dalal at einfochips.com (Kushal Dalal)
Date: Wed, 14 Jun 2006 17:11:20 +0530
Subject: [Live-devel] increase of buffer size
In-Reply-To: <64849090606140432n6a987b6fhba48f02b45a26708@mail.gmail.com>
Message-ID: <200606141141.k5EBfhcM089280@ns.live555.com>
Look at OutPacketBuffer class implementation (specially constructor) to
change buffer size.
-----Original Message-----
From: live-devel-bounces at ns.live555.com
[mailto:live-devel-bounces at ns.live555.com] On Behalf Of subrato roy
Sent: Wednesday, June 14, 2006 5:03 PM
To: live-devel at ns.live555.com
Subject: [Live-devel] increase of buffer size
Hello ,
I am working with the framedSource::getNextFrame(). there is a need
for me to get as large a read as possible and fill it up in the fTo
bufer. Can someone tell me how to increase the buffer size for this.
Also i would like to know how large a buffer can be done in this way .
thanks for all the help.
subrato
eInfochips Business Disclaimer:
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From bidibulle at operamail.com Wed Jun 14 08:50:39 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Wed, 14 Jun 2006 16:50:39 +0100
Subject: [Live-devel] Groupsock handling in OnDemandServerMediaSubsession.cpp
Message-ID: <20060614155039.E1D8343CBF@ws5-1.us4.outblaze.com>
Hi Ross,
I'm facing a problem with groupsock creation in a class derived from OnDemandServerMediaSubsession. I know you normally don't give support on modified code but as I guess the issue also resides in the original code too, I will try to catch your interest anyway :-)
Those problems mainly occur under heavy load when a RTSP server receives lots of unicast requests from clients.
Basically, we are in the getStreamParameters method and fReuseFirstSource is false so for each new request we are supposed to create a new groupsock for the RTP destination and a new groupsock for the RTCP destination. The RTP port must have an even number and the RTCP port is supposed to be RTP port + 1. If the RTP port is not even a new trial is done and the old port number is for sure not used because kept for a while in rtpGroupsock_old
Extract from the code :
[...]
while (1) {
rtpGroupsock = new Groupsock(envir(), dummyAddr, 0, 255);
if (!getSourcePort(envir(), rtpGroupsock->socketNum(), serverRTPPort)) break;
serverRTPPortNum = ntohs(serverRTPPort.num());
// If the port number's even, we're done:
if ((serverRTPPortNum&1) == 0) break;
// Try again (while keeping the old 'groupsock' around, so that we get
// a different socket number next time):
delete rtpGroupsock_old;
rtpGroupsock_old = rtpGroupsock;
}
delete rtpGroupsock_old;
[...]
My problem was that if the RTP port is not even for two consecutive times, the first failed port number is not kept in rtpGroupsock_old anymore and will be used again. So, a new failed port and the previous one (also failed) available again --> loop of failed trials.
So I tried to optimize this algorithm. In case of failed port number, I call the Groupsock constructor with a "preferred" (even !) port number, which value is previous_trial_port + 1. But the problem is that liveMedia code was designed to support multicast, the socket options are such that you can bind twice on the same port number (SO_REUSEADDR and SO_REUSEPOR are True, see this in GroupSockHelper::setupDatagramSocket()). Therefore I can have my RTP ports shared for several sessions which is something I certainly don't want.
I tried to modified the socket options in GroupSockHelper and I got what I wanted. So the question is, do you follow my analysis and agree with the issue ? If yes, how can we get a satisfying result ? If the solution is to modify the socket options this is currently not possible in runtime as the SO_REUSEPORT and SO_REUSEADDR are always true because the reuseFlag is hardcoded to value 1. What about giving this reuseFlag in Groupsock constructor ?
Thanks in advance for your feedback,
David
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 8 at http://www.opera.com
Powered by Outblaze
From finlayson at live555.com Wed Jun 14 13:53:43 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 14 Jun 2006 13:53:43 -0700
Subject: [Live-devel] increase of buffer size
In-Reply-To: <64849090606140432n6a987b6fhba48f02b45a26708@mail.gmail.com
>
References: <64849090606140432n6a987b6fhba48f02b45a26708@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060614135055.02021468@live555.com>
At 04:32 AM 6/14/2006, you wrote:
>Hello ,
>
>I am working with the framedSource::getNextFrame(). there is a need
>for me to get as large a read as possible and fill it up in the fTo
>bufer. Can someone tell me how to increase the buffer size for this.
The buffer (and thus buffer size) is specified by the 'sink' object
that's at the end of the chain - i.e., the object that you call
"startPlaying()" on.
For example, for "FileSink", there is a buffer size parameter.
If your sink is a "RTPSink" subclass - i.e., for sending RTP packets
- then you specify the buffer size by setting
"OutPacketBuffer::maxSize" *before* you create the "RTPSink" object.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Wed Jun 14 14:08:27 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 14 Jun 2006 14:08:27 -0700
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
In-Reply-To: <20060614155039.E1D8343CBF@ws5-1.us4.outblaze.com>
References: <20060614155039.E1D8343CBF@ws5-1.us4.outblaze.com>
Message-ID: <7.0.1.0.1.20060614135757.01fb9870@live555.com>
>So the question is, do you follow my analysis and agree with the issue ?
Yes, I agree that there's an issue - however, I don't agree with the
proposed solution of not setting the SO_REUSE* options when creating
sockets. IMHO, all sockets should have those options
set. (Monitoring/debugging of network applications by other
applications running on the same computer is easier if you know that
all sockets will be readable.)
The solution to the problem you saw is to modify the code (that you
quoted) to keep around 'groupsocks' for all trials, not just the most
recent one - i.e., to change "rtpGroupsock_old" to be a list of
pointers, rather than just a single pointer. I don't have time right
now to make this change, so feel free to propose a modification and
send a patch to the list.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From kushal.dalal at einfochips.com Wed Jun 14 22:16:55 2006
From: kushal.dalal at einfochips.com (Kushal Dalal)
Date: Thu, 15 Jun 2006 10:46:55 +0530
Subject: [Live-devel] streaming method
In-Reply-To: <200606130441.k5D4fNcX071479@ns.live555.com>
Message-ID: <200606150517.k5F5HIgH088199@ns.live555.com>
Hi
Please attend to this mail.
I am in real confusion.
Help me.
Thanks
Kushal
_____
From: live-devel-bounces at ns.live555.com
[mailto:live-devel-bounces at ns.live555.com] On Behalf Of Kushal Dalal
Sent: Tuesday, June 13, 2006 10:11 AM
To: live-devel at ns.live555.com
Subject: [Live-devel] streaming method
Hi All
I am new to streaming. I have some basic questions.
I want to develop network camera.
My requirement is something like this.
I will capture live video and compress it using encoder.
That compressed live video should be streamed over network.
Client connected to camera should view live video using media player plugins
(VLC, Real, WMP) of web browser.
Which streaming method (RTP/RTSP, HTTP, or .) should be used for that?
Can I do that using Live?
Thanks in advance.
Kushal
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From sarpotdaramogh at yahoo.com Thu Jun 15 03:13:00 2006
From: sarpotdaramogh at yahoo.com (Amogh Sarpotdar)
Date: Thu, 15 Jun 2006 03:13:00 -0700 (PDT)
Subject: [Live-devel] taskScheduler.doEventLoop : RTSPServer does not return
when in loop
Message-ID: <20060615101300.48363.qmail@web55503.mail.re4.yahoo.com>
Hi all,
I have created a dialog based test application using live library which serves incoming RTSP audio/video requests.
Similar to test application I have called the taskScheduler().doEventLoop() function as shown below -
env->taskScheduler().doEventLoop(); // does not return
nReturnResult = 0;
Problem is that once test application starts, it dosent respond to any of GUI actions (e.g. attempting to close the application). Any idea how to interrupt this loop?
Thanks in advance,
With Best Regards
Amogh Sarpotdar
Telesoft, India
__________________________________________________
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From bidibulle at operamail.com Thu Jun 15 03:36:16 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Thu, 15 Jun 2006 11:36:16 +0100
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
Message-ID: <20060615103616.9356E43CBF@ws5-1.us4.outblaze.com>
Ross,
I see your point. Anyway, let me another chance to convince you :
- Monitoring/debugging of LiveMedia based applications will only be possible on UDP ports and on non-Windows environments (windows doesn't support SO_REUSEPORT option)
- In the current code, you create the RTCP groupsock by specifying the desired port number (RTP+1). If this port is already used by another process, your RTCP handler could receive any type of undesired traffic
- I found a compromise that should make everybody happy : The set of "setsockoption" could be done after the socket binding. It would avoid LiveMedia to open a socket which is already in use and enable debugging applicaction to inspect the LiveMedia's UDP sockets.
- If you agree the RTCP groupsock creation in OnDemandServerMediaSubsession.cpp will have to be followed by an error handling in case the bind() fails (getSourcePort() returns False in this case). The rest of my proposed algorithm to optimize trials could remain in my own code...
What's your feeling about that ?
David
const int reuseFlag = 1;
if (setsockopt(newSocket, SOL_SOCKET, SO_REUSEADDR,
(const char*)&reuseFlag, sizeof reuseFlag) < 0) {
socketErr(env, "setsockopt(SO_REUSEADDR) error: ");
closeSocket(newSocket);
return -1;
}
#if defined(__WIN32__) || defined(_WIN32)
// Windoze doesn't handle SO_REUSEPORT or IP_MULTICAST_LOOP
#else
#ifdef SO_REUSEPORT
if (setsockopt(newSocket, SOL_SOCKET, SO_REUSEPORT,
(const char*)&reuseFlag, sizeof reuseFlag) < 0) {
socketErr(env, "setsockopt(SO_REUSEPORT) error: ");
closeSocket(newSocket);
return -1;
}
#endif
#ifdef IP_MULTICAST_LOOP
const u_int8_t loop = (u_int8_t)setLoopback;
if (setsockopt(newSocket, IPPROTO_IP, IP_MULTICAST_LOOP,
(const char*)&loop, sizeof loop) < 0) {
socketErr(env, "setsockopt(IP_MULTICAST_LOOP) error: ");
closeSocket(newSocket);
return -1;
}
#endif
#endif
> ----- Original Message -----
> From: "Ross Finlayson"
> To: "LIVE555 Streaming Media - development & use"
> Subject: Re: [Live-devel] Groupsock handling in OnDemandServerMediaSubsession.cpp
> Date: Wed, 14 Jun 2006 14:08:27 -0700
>
>
>
> > So the question is, do you follow my analysis and agree with the issue ?
>
> Yes, I agree that there's an issue - however, I don't agree with the
> proposed solution of not setting the SO_REUSE* options when creating
> sockets. IMHO, all sockets should have those options
> set. (Monitoring/debugging of network applications by other
> applications running on the same computer is easier if you know that
> all sockets will be readable.)
>
> The solution to the problem you saw is to modify the code (that you
> quoted) to keep around 'groupsocks' for all trials, not just the most
> recent one - i.e., to change "rtpGroupsock_old" to be a list of
> pointers, rather than just a single pointer. I don't have time right
> now to make this change, so feel free to propose a modification and
> send a patch to the list.
>
>
> 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
>
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 8 at http://www.opera.com
Powered by Outblaze
From bidibulle at operamail.com Thu Jun 15 03:41:32 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Thu, 15 Jun 2006 11:41:32 +0100
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
Message-ID: <20060615104132.A830E43CC2@ws5-1.us4.outblaze.com>
In my previous email, the quoted code at the end of the message would be the code to move after the bind(). Sorry for forgetting this in my previous message.
David
> ----- Original Message -----
> From: "Ross Finlayson"
> To: "LIVE555 Streaming Media - development & use"
> Subject: Re: [Live-devel] Groupsock handling in OnDemandServerMediaSubsession.cpp
> Date: Wed, 14 Jun 2006 14:08:27 -0700
>
>
>
> > So the question is, do you follow my analysis and agree with the issue ?
>
> Yes, I agree that there's an issue - however, I don't agree with the
> proposed solution of not setting the SO_REUSE* options when creating
> sockets. IMHO, all sockets should have those options
> set. (Monitoring/debugging of network applications by other
> applications running on the same computer is easier if you know that
> all sockets will be readable.)
>
> The solution to the problem you saw is to modify the code (that you
> quoted) to keep around 'groupsocks' for all trials, not just the most
> recent one - i.e., to change "rtpGroupsock_old" to be a list of
> pointers, rather than just a single pointer. I don't have time right
> now to make this change, so feel free to propose a modification and
> send a patch to the list.
>
>
> 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
>
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 8 at http://www.opera.com
Powered by Outblaze
From roy.subrato at gmail.com Thu Jun 15 04:20:49 2006
From: roy.subrato at gmail.com (subrato roy)
Date: Thu, 15 Jun 2006 14:20:49 +0300
Subject: [Live-devel] increase of buffer size
In-Reply-To: <7.0.1.0.1.20060614135055.02021468@live555.com>
References: <64849090606140432n6a987b6fhba48f02b45a26708@mail.gmail.com>
<7.0.1.0.1.20060614135055.02021468@live555.com>
Message-ID: <64849090606150420t4bc08111jca3b4326824922b9@mail.gmail.com>
On 6/14/06, Ross Finlayson wrote:
> At 04:32 AM 6/14/2006, you wrote:
> >Hello ,
> >
> >I am working with the framedSource::getNextFrame(). there is a need
> >for me to get as large a read as possible and fill it up in the fTo
> >bufer. Can someone tell me how to increase the buffer size for this.
>
> The buffer (and thus buffer size) is specified by the 'sink' object
> that's at the end of the chain - i.e., the object that you call
> "startPlaying()" on.
>
> For example, for "FileSink", there is a buffer size parameter.
>
> If your sink is a "RTPSink" subclass - i.e., for sending RTP packets
> - then you specify the buffer size by setting
> "OutPacketBuffer::maxSize" *before* you create the "RTPSink" object.
>
>
> Ross Finlayson
Hello All,
As instructed by you i changed the buffer size for the sink. But still
the buffer gets filled with 7 TS packets . looking at the code i think
i need to change the frame size to get more TS packets to fill this
bigger buffer completely. this is how i calculate number of TS packets
:
fFrameSize += frameSize;
unsigned const numTSPackets = fFrameSize/TRANSPORT_PACKET_SIZE;
fFrameSize = numTSPackets*TRANSPORT_PACKET_SIZE;
in the function
afterGettingFrame(void* clientData, unsigned frameSize, ....)
to summarize my problem again :
1) i need this clientData (my buffer ) to be bigger and i think
i achieved this.
2) i need frameSize increased to actually allow me to have more
TS packets in this Bigger buffer.
Thanks for all your help
subrato
From vlcplayer at gmail.com Thu Jun 15 06:04:42 2006
From: vlcplayer at gmail.com (Lukasz Kondrad)
Date: Thu, 15 Jun 2006 16:04:42 +0300
Subject: [Live-devel] [live-devel] Specify network adapter
Message-ID:
Hi there.
Since my last mail got no answer, i will try a new one, cleaner.
I have two network adapters, one eth0 (ethernet) and the second dvb0_0
(created on dvb card).
I try to decode a dvb-h signal and to do that I am using Vlc player. When I
transmit a multicast, ts over udp/ip, I can specify the interface's address
(network adapter's address) in vlc so it takes the stream from dvb0_0 and
the multicast is decoded. But when I transmit mpeg4 video over rtp, vlc uses
livedotcom module which is based on live555 library. The live555 doesn't
take into account the interface address which I specified and it tried to
take the stream from eth0.
My question: how can I specify on which network addapter shall live555 look
on.
BR
Lukasz
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From finlayson at live555.com Thu Jun 15 07:39:06 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 15 Jun 2006 07:39:06 -0700
Subject: [Live-devel] taskScheduler.doEventLoop : RTSPServer does not
return when in loop
In-Reply-To: <20060615101300.48363.qmail@web55503.mail.re4.yahoo.com>
References: <20060615101300.48363.qmail@web55503.mail.re4.yahoo.com>
Message-ID: <7.0.1.0.1.20060615073848.01b89270@live555.com>
>Problem is that once test application starts, it dosent respond to
>any of GUI actions (e.g. attempting to close the application). Any
>idea how to interrupt this loop?
Yes. Read the FAQ.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Thu Jun 15 07:55:30 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 15 Jun 2006 07:55:30 -0700
Subject: [Live-devel] [live-devel] Specify network adapter
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060615074545.01bb1300@live555.com>
>My question: how can I specify on which network addapter shall
>live555 look on.
You can do this by setting the global variables "SenderInterfaceAddr"
and/or "ReceivingInterfaceAddr", *before* creating network objects
(e.g., groupsocks, RTPSources or RTPSinks).
See the "case 'I':" code in "testProgs/playCommon.cpp" for an example
of how "ReceivingInterfaceAddr" can be set. (Alternatively, if you
have just an IP address, rather than a domain name, you could call
"our_inet_addr()" to set it.)
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Thu Jun 15 08:03:45 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 15 Jun 2006 08:03:45 -0700
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
In-Reply-To: <20060615103616.9356E43CBF@ws5-1.us4.outblaze.com>
References: <20060615103616.9356E43CBF@ws5-1.us4.outblaze.com>
Message-ID: <7.0.1.0.1.20060615075741.01c217c8@live555.com>
>- I found a compromise that should make everybody happy : The set
>of "setsockoption" could be done after the socket binding. It would
>avoid LiveMedia to open a socket which is already in use and enable
>debugging applicaction to inspect the LiveMedia's UDP sockets.
>
>- If you agree the RTCP groupsock creation in
>OnDemandServerMediaSubsession.cpp will have to be followed by an
>error handling in case the bind() fails (getSourcePort() returns
>False in this case). The rest of my proposed algorithm to optimize
>trials could remain in my own code...
>
>What's your feeling about that ?
Please send a complete set of proposed patches to the current source
code, and I'll take a look at it.
(I'm a bit worried that changing the order that
"setsockopt(SO_REUSE...)" and "bind()" is done might prevent pairs of
applications like "testMP3Streamer" and "testMP3Receiver" from
working together on the same (Unix) host - if started in either
order. So please test for that also.)
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From vlcplayer at gmail.com Fri Jun 16 03:24:25 2006
From: vlcplayer at gmail.com (Lukasz Kondrad)
Date: Fri, 16 Jun 2006 13:24:25 +0300
Subject: [Live-devel] [live-devel] Specify network adapter
In-Reply-To: <7.0.1.0.1.20060615074545.01bb1300@live555.com>
References:
<7.0.1.0.1.20060615074545.01bb1300@live555.com>
Message-ID:
On 6/15/06, Ross Finlayson wrote:
>
>
> >My question: how can I specify on which network addapter shall
> >live555 look on.
>
> You can do this by setting the global variables "SenderInterfaceAddr"
> and/or "ReceivingInterfaceAddr", *before* creating network objects
> (e.g., groupsocks, RTPSources or RTPSinks).
>
> See the "case 'I':" code in "testProgs/playCommon.cpp" for an example
> of how "ReceivingInterfaceAddr" can be set. (Alternatively, if you
> have just an IP address, rather than a domain name, you could call
> "our_inet_addr()" to set it.)
Hello,
Thanks for your answer,
I did as you said I set the ReceivingInterfaceAddr
ReceivingInterfaceAddr = our_inet_addr("130.230.12.126");
but some problem appers after execution:
Unable to determine our source address: no address
What could be wrong ??
Best regards
Lukasz
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From vlcplayer at gmail.com Fri Jun 16 05:10:47 2006
From: vlcplayer at gmail.com (Lukasz Kondrad)
Date: Fri, 16 Jun 2006 15:10:47 +0300
Subject: [Live-devel] [live-devel] Specify network adapter
In-Reply-To:
References:
<7.0.1.0.1.20060615074545.01bb1300@live555.com>
Message-ID:
Hello Ross,
I am sorry I was hurry with my question. I insted of ReceivingInterfaceAddr
I set SendingInterfaceAddr = our_inet_addr("130.230.50.50") and now it
works. I receive the ip flow from dvb0_0 card so I can play the dvb-h
stream.
Thanks for your help.
BR
Lukasz
On 6/16/06, Lukasz Kondrad wrote:
>
>
>
> On 6/15/06, Ross Finlayson wrote:
> >
> >
> > >My question: how can I specify on which network addapter shall
> > >live555 look on.
> >
> > You can do this by setting the global variables "SenderInterfaceAddr"
> > and/or "ReceivingInterfaceAddr", *before* creating network objects
> > (e.g., groupsocks, RTPSources or RTPSinks).
> >
> > See the "case 'I':" code in "testProgs/playCommon.cpp" for an example
> > of how "ReceivingInterfaceAddr" can be set. (Alternatively, if you
> > have just an IP address, rather than a domain name, you could call
> > "our_inet_addr()" to set it.)
>
>
>
> Hello,
> Thanks for your answer,
>
> I did as you said I set the ReceivingInterfaceAddr
> ReceivingInterfaceAddr = our_inet_addr("130.230.12.126");
>
> but some problem appers after execution:
>
> Unable to determine our source address: no address
>
> What could be wrong ??
>
> Best regards
> Lukasz
>
>
>
>
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From Christian.Lees at dsto.defence.gov.au Sun Jun 18 20:39:58 2006
From: Christian.Lees at dsto.defence.gov.au (Lees, Christian)
Date: Mon, 19 Jun 2006 13:39:58 +1000
Subject: [Live-devel] use with uclibc
Message-ID:
Has anyone gotten the LIVE libraries to work under a system using
uClibc. I have managed to get the libraries and test programs to
compile, but when I go to run the test programs I get the warnings
# ./testMPEG4VideoStreamer
18:34:17 Groupsock(4: 232.81.158.102, 18888, 255): failed to join group:
setsockopt(IP_ADD_MEMBERSHIP) error: No such device
Unable to determine our source address: This computer has an invalid IP
address: 0x0
18:34:17 Groupsock(5: 232.81.158.102, 18889, 255): failed to join group:
setsockopt(IP_ADD_MEMBERSHIP) error: No such device
Unable to determine our source address: This computer has an invalid IP
address: 0x0
Play this stream using the URL "rtsp://0.0.0.0:8554/testStream"
Beginning streaming...
Beginning to read from file...
I can connect from another machine, it gets the SDP information right,
but no RTP packets arrive. Multicast is enabled in the kernel. Has
this got something to do with uClibc's incomplete libresolv?
Any Suggestions? Other then moving to glibc. The machine is a Via C3-2,
kernel 2.6.10 with the adeos extensions.
TIA
Christian
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From finlayson at live555.com Sun Jun 18 21:58:49 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 18 Jun 2006 21:58:49 -0700
Subject: [Live-devel] use with uclibc
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060618214948.01fa6ea8@live555.com>
> Multicast is enabled in the kernel.
Are you sure (see below)?
> Has this got something to do with uClibc's incomplete libresolv?
Possibly. Your immediate problem is that the function
"ourSourceAddressForMulticast()" is failing. I suggest tracing
through the execution of that code (in
"groupsock/GroupsockHelper.cpp") and figuring out why it fails for you.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From bidibulle at operamail.com Mon Jun 19 05:09:02 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Mon, 19 Jun 2006 13:09:02 +0100
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
Message-ID: <20060619120902.A52ECCA0A6@ws5-11.us4.outblaze.com>
Ross,
> (I'm a bit worried that changing the order that
> "setsockopt(SO_REUSE...)" and "bind()" is done might prevent pairs of
> applications like "testMP3Streamer" and "testMP3Receiver" from
> working together on the same (Unix) host - if started in either
> order. So please test for that also.)
Actually each time a Groupsock is created with a preferred port in
constructor, I should add an error handling mechanism in case the
port is already used. So the task will be difficult and quite
risky. So, as I really need to have the possibility to use free
ports only and I see there is no big interest for this for the
existing library, I made a small patch to GroupsockHelper.ccp that
test a compilation flag called DONOTREUSEPORT before seting
SO_REUSE_ADDR and SO_REUSE_PORT. So, the whole livemedia code won't
be impacted at all (default is false of course) and the feature can
be simply activated if required by adding this flag in the
Makefile(s).
David
>
>
> 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
>
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 8 at http://www.opera.com
Powered by Outblaze
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From finlayson at live555.com Mon Jun 19 07:46:20 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 19 Jun 2006 07:46:20 -0700
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
In-Reply-To: <20060619120902.A52ECCA0A6@ws5-11.us4.outblaze.com>
References: <20060619120902.A52ECCA0A6@ws5-11.us4.outblaze.com>
Message-ID: <7.0.1.0.1.20060619072230.01f7b8c8@live555.com>
>So, as I really need to have the possibility to use free
>ports only and I see there is no big interest for this for the
>existing library
Now I'm confused. I thought you had identified an issue (in the code
in "OnDemandServerMediaSubsession.cpp" where we check that a
newly-created 'groupsock' - with no explicitly-specified port number
- ends up getting an even-numbered port) that might affect *every*
user of the library (if they're running a unicast RTSP/RTP
server). If that's the case, then we can't make the use of the
SO_REUSE_* flags a compile-time option (because, in general, for
other uses of 'groupsock's, we want the SO_REUSE_* flags set).
If, however, this is something that - for whatever reason - affects
your particular application only, then I'm not going to make a change
to the released code just for this.
So the real question is: Is there, or is there not, a real problem
with the way that the code in "OnDemandServerMediaSubsession.cpp"
currently works. If there is (although I, personally, have never
seen the current code fail), then perhaps the best solution would be
to have a special new Groupsock constructor just for this case: I.e.,
for creating one that ends up using an arbitrary, but even-numbered
port. This special new constructor could then omit setting the
SO_REUSE_* flags, if that's necessary to make it work (although I'm
not convinced of that either).
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From bidibulle at operamail.com Mon Jun 19 10:06:49 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Mon, 19 Jun 2006 18:06:49 +0100
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
Message-ID: <20060619170649.DF61CCA0A4@ws5-11.us4.outblaze.com>
> Now I'm confused. I thought you had identified an issue (in the code
> in "OnDemandServerMediaSubsession.cpp" where we check that a
> newly-created 'groupsock' - with no explicitly-specified port number
> - ends up getting an even-numbered port) that might affect *every*
> user of the library (if they're running a unicast RTSP/RTP
> server).
Actually yes, but it is mainly a performance issue. When trying to get your even RTP port, if you fail twice successfully then you will loop (for a whisle) trying to use the previously failed ports. That's something you don't necessarily see (depending probably on the OS you use and the load of your server) but is a problem for me.
Whisle I was investigating further and trying to fix this by changing the algorithm to get the right port number, I discovered another issue. When you specify the port you want, you may get it, even if the port is used already. Typically, in OnDemandServerMediaSubsession.cpp, when you ask for the RTCP port (RTP port + 1), but also in many test programs (for example testMP3Streamer.cpp). Another thing that is an issue for me (I don't want to receive unexpected data on my UDP sockets).
To fix the first issue, I will specify (in my own code) a "preferred port" number to get my even port number (for example, by trying previous_failed_port + 1) because I think it's the best direction to follow and also because I want to define a range of allowed ports for my server application. But then I will met the second issue for RTP sockets also, which is worst !.
So, I must fix the second issue. And the the best way to do it is to bind() before setting the socket options. You will then be unable to receive a UDP port already used.
But doing so means that you must check the whole liveMedia library to verify with a getSourceName() call that you got the socket number you asked. And if not, you have to handle the error correctly. This is typically not an easy task in the case of OnDemandServerMediaSubsession.cpp because if your RTCP port request fails you have to get back to the RTP port request. So I should change many parts of the code and test all test applications for non-regression. This is too much work, time and risk if I'm te only one complaining about those two issues. Therefore I proposed a minimalist patch that doesn't impact the library code but allow me to keep it used in my server application.
> If there is (although I, personally, have never
> seen the current code fail), then perhaps the best solution would be
> to have a special new Groupsock constructor just for this case: I.e.,
> for creating one that ends up using an arbitrary, but even-numbered
> port.
> This special new constructor could then omit setting the
> SO_REUSE_* flags, if that's necessary to make it work (although I'm
> not convinced of that either).
This is indeed an alternative to the compilation option. Actually, it would be a new constructor with reuseFlag in argument (default value = 1) and would allow to ask for a specific port number, and to get only if it is free. I can provide this new patch to GroupsockHelper class if agreed.
David
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 8 at http://www.opera.com
Powered by Outblaze
From finlayson at live555.com Mon Jun 19 11:15:02 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 19 Jun 2006 11:15:02 -0700
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
In-Reply-To: <20060619170649.DF61CCA0A4@ws5-11.us4.outblaze.com>
References: <20060619170649.DF61CCA0A4@ws5-11.us4.outblaze.com>
Message-ID: <7.0.1.0.1.20060619111238.01f8a7d8@live555.com>
> > If there is (although I, personally, have never
> > seen the current code fail), then perhaps the best solution would be
> > to have a special new Groupsock constructor just for this case: I.e.,
> > for creating one that ends up using an arbitrary, but even-numbered
> > port.
> > This special new constructor could then omit setting the
> > SO_REUSE_* flags, if that's necessary to make it work (although I'm
> > not convinced of that either).
>This is indeed an alternative to the compilation option. Actually,
>it would be a new constructor with reuseFlag in argument (default
>value = 1) and would allow to ask for a specific port number, and to
>get only if it is free. I can provide this new patch to
>GroupsockHelper class if agreed.
Yes, please do. One of my long-planned goals for our RTSP server
implementation has been to select ports from a small range of
'allowable' port numbers (just as DSS, Helix, etc. already do) - to
make firewalling easier. This change will help make that possible.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From braymond at echostorm.net Mon Jun 19 21:07:54 2006
From: braymond at echostorm.net (Brian Raymond)
Date: Tue, 20 Jun 2006 00:07:54 -0400
Subject: [Live-devel] Compile issues on OSX 10.4.6 4.0.1
Message-ID:
I didn't have a lot of time to chase down what the root cause of this
was but I was having the following issue on OSX 10.4.6 with GCC 4.0.1
RTSPServer.cpp: In member function 'void
RTSPServer::incomingConnectionHandler1()':
RTSPServer.cpp:212: error: invalid conversion from 'int*' to
'socklen_t*'
RTSPServer.cpp:212: error: initializing argument 3 of 'int
accept(int, sockaddr*, socklen_t*)'
Casting the parameter to "(socklen_t*)" in all of the code using
SOCKLEN_T solves the problem, HTTPSink.cpp, RTSPServer.cpp,
GroupSockHelper.cpp. I didn't send a patch because I didn't want to
assume that's the way it should be given my use of OSX with 4.0.1 (I'm
battling a couple of GCC bugs with Apple right now so there are some
issues :) ).
From finlayson at live555.com Mon Jun 19 21:51:36 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 19 Jun 2006 21:51:36 -0700
Subject: [Live-devel] Compile issues on OSX 10.4.6 4.0.1
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060619214852.01f5bc70@live555.com>
>Casting the parameter to "(socklen_t*)" in all of the code using
>SOCKLEN_T solves the problem
Did you create your "Makefile"s using
genMakefiles macosx
?
If so, then you should get
-DSOCKLEN_T=socklen_t
on each compile line.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From brainlai1102 at gmail.com Tue Jun 20 22:08:49 2006
From: brainlai1102 at gmail.com (Brain Lai)
Date: Wed, 21 Jun 2006 13:08:49 +0800
Subject: [Live-devel] Is it reasonable for QuickTimeFileSink to
compensate video packet loss?
In-Reply-To: <2dd7fdf10606202032y73f66fb3r9b6cc2a1639888d3@mail.gmail.com>
References: <2dd7fdf10606202032y73f66fb3r9b6cc2a1639888d3@mail.gmail.com>
Message-ID: <2dd7fdf10606202208t611b7f5k382c69eb65e45b6f@mail.gmail.com>
Dear all:
In QuickTimeFileSink, I note that the video is compensated when the variable
seqNumGap (rtpSeqNum - fLastPacketRTPSeqNum) is larger than 1. However, for
a video frame consisting of many rtp packet, the seqNumGap is usually larger
than 1, that is, the compensation is always performed for video frames if
the Sink variable fPacketLossCompensate is set true. Is it reasonable, or I
miss something ?
Reguards
Brain Lai
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From finlayson at live555.com Wed Jun 21 02:26:11 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 21 Jun 2006 02:26:11 -0700
Subject: [Live-devel] Is it reasonable for QuickTimeFileSink to
compensate video packet loss?
In-Reply-To: <2dd7fdf10606202208t611b7f5k382c69eb65e45b6f@mail.gmail.com
>
References: <2dd7fdf10606202032y73f66fb3r9b6cc2a1639888d3@mail.gmail.com>
<2dd7fdf10606202208t611b7f5k382c69eb65e45b6f@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060621022432.01fba9c0@live555.com>
>In QuickTimeFileSink, I note that the video is compensated when the
>variable seqNumGap (rtpSeqNum - fLastPacketRTPSeqNum) is larger than
>1. However, for a video frame consisting of many rtp packet, the
>seqNumGap is usually larger than 1, that is, the compensation is
>always performed for video frames if the Sink variable
>fPacketLossCompensate is set true. Is it reasonable, or I miss something ?
The "fPacketLossCompensate" stuff is just a hack that we added for a
customer several years ago. It shouldn't be relied upon to do
anything useful, and that code will probably be removed completely someday.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From vlcplayer at gmail.com Wed Jun 21 02:50:39 2006
From: vlcplayer at gmail.com (Lukasz Kondrad)
Date: Wed, 21 Jun 2006 12:50:39 +0300
Subject: [Live-devel] [live-devel] receivingInterfaceAddress problem
Message-ID:
Hi Ross,
What does it mean when after specify ReceivingInterfaceAddr =
our_inet_addr("130.230.12.126") I get Unable to determine our source
address: This computer has a invalid IP address 0x0. May you tell me what is
the cause of that??
Best Regards
Lukasz
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From vlcplayer at gmail.com Wed Jun 21 06:41:06 2006
From: vlcplayer at gmail.com (Lukasz Kondrad)
Date: Wed, 21 Jun 2006 16:41:06 +0300
Subject: [Live-devel] [live-devel] Stange problem
Message-ID:
Hello Ross,
I transmit two rtp streams over dvb-h. The first one is generated by
H.264real time encoder (envivio 4caster M2) and send on multimedia
server
(cardinal), the second one is generated from a mp4 file by multimedia server
(cardinal) by itself.
I receive it on linux machine with dvb card. I set dvb network adapter on
proper pid, i tune the card and by tcpdump I see that IP datagrams from
which carry both rtp streams are received by the network adapter. I run vlc
with sdp files. The strange is that I am able to receive the rtp stream
generated by H.264 real time encoder but not the one which multimedia server
generates. I am quite confused and I don't know what is the problem. Below I
presents the sdp files for both streams, maybe there is something what
live555 don't take into consideration.
I thought that the library doesn\t look on proper card but why once it takes
the rtp packets from the card and for the second stream it doesn't.
sdp generated by H.264 encoder
v=0
o=- 1025276472 3155130398 IN IP4 130.230.50.198
s=TUT - test
c=IN IP4 239.252.80.1/128
b=RR:0
t=0 0
a=mpeg4-iod:"data:application/mpeg4-iod;base64,AoIzAE///w/0/wOBPgABQKBkYXRhOmFwcGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBVnNCS2dVZkF5WUF5U0FBWlFRTklCRUFZYWdBQVlhZ0FBR0dvQVlRQUVRQUFWK1FBQUZma0NBQUFBQUFBd0V0QXA4REtRQmxBQVFOUUJVQUJnQUFBTU5RQUFERFVBWVFBRVFBQUx1QUFBRmZrQ0FBQUFBQUEwTURaVzVuBA0BBQAAyAAAAAAAAAAABgkBAAAAAAAAAAADaQACQEZkYXRhOmFwcGxpY2F0aW9uL21wZWc0LWJpZnMtYXU7YmFzZTY0LHdCQVNnVEFxQlcwbUVFSDhBQUFCL0FBQUJFS0NLQ240BBICDQAAIAAAAAAAAAAABQMAAGAGCQEAAAAAAAAAAA=="
m=video 9760 RTP/AVP 96
b=AS:100
a=rtpmap:96 MP4V-ES/90000
a=fmtp:96 profile-level-id=244;
config=000001B0F4000001B50CA020202F000001000000012000C7F9D8D8B0D450B04481498F
a=cliprect:0,0,576,352
a=mpeg4-esid:201
a=x-envivio-verid:00022159
m=audio 9762 RTP/AVP 97
b=AS:50
a=rtpmap:97 mpeg4-generic/48000/2
a=fmtp:97 profile-level-id=15; config=1390; streamtype=5; mode=AAC-hbr;
SizeLength=13; IndexLength=3; IndexDeltaLength=3
a=mpeg4-esid:101
a=lang:eng
a=x-envivio-verid:00022159
sdp generated by multimedia server
v=0
o=QTSS_Play_List 1460042056 502683559 IN IP4 130.230.50.48
s=stream-30
c=IN IP4 239.252.80.3/1
t=0 0
a=x-broadcastcontrol:RTSP
a=mpeg4-iod:"data:application/mpeg4-iod;base64,AoIzAE///w8D/wOBPgABQKBkYXRhOmFwcGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBVnNCS2dVZkF5WUF5U0FBWlFRTklCRUJKUGdBQkpQZ0FBU1Q0QVlRQUVRQUFWK1FBQUZma0NBQUFBQUFBd0V0QXA4REtRQmxBQVFOUUJVQUJnQUFBVitRQUFGZmtBWVFBRVFBQUx1QUFBRmZrQ0FBQUFBQUEwTURaVzVuBA0BBQAAyAAAAAAAAAAABgkBAAAAAAAAAAADaQACQEZkYXRhOmFwcGxpY2F0aW9uL21wZWc0LWJpZnMtYXU7YmFzZTY0LHdCQVNnVEFxQlcwbUVFSDhBQUFCL0FBQUJFS0NLQ240BBICDQAAIAAAAAAAAAAABQMAAGAGCQEAAAAAAAAAAA=="
a=ISMA-compliance:1,1,1
m=video 5004 RTP/AVP 96
b=AS:312
a=rtpmap:96 MP4V-ES/90000
a=fmtp:96 profile-level-id=3;
config=000001B003000001B50CA020202F000001000000012000C7F80808B0D450A04181463F
a=cliprect:0,0,180,320
a=mpeg4-esid:201
a=x-envivio-verid:00022159
a=control:trackID=1
m=audio 5006 RTP/AVP 97
b=AS:106
a=rtpmap:97 mpeg4-generic/48000/2
a=fmtp:97 profile-level-id=15; config=1190; streamtype=5; mode=AAC-hbr;
SizeLength=13; IndexLength=3; IndexDeltaLength=3
a=mpeg4-esid:101
a=lang:eng
a=x-envivio-verid:00022159
a=control:trackID=2
Thank your help.
BR
Lukasz
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From barounis at ceid.upatras.gr Wed Jun 21 07:24:45 2006
From: barounis at ceid.upatras.gr (barounis at ceid.upatras.gr)
Date: Wed, 21 Jun 2006 17:24:45 +0300
Subject: [Live-devel] RTCP RR Handler function
In-Reply-To: <7.0.1.0.1.20060613122349.01f4ddd8@live555.com>
References: <1143586422.4429be76673a7@my.ceid.upatras.gr>
<7.0.1.0.1.20060328150729.01f93568@live555.com>
<1144792970.443c278a9bc41@my.ceid.upatras.gr>
<7.0.1.0.1.20060411151024.01d638a0@live555.com>
<1146055961.444f6d19519e3@my.ceid.upatras.gr>
<1146132663.445098b73a8d8@my.ceid.upatras.gr>
<7.0.1.0.1.20060427053054.01f51830@live555.com>
<1146146329.4450ce19cb6d9@my.ceid.upatras.gr>
<7.0.1.0.1.20060427073327.01f7ed50@live555.com>
<1146923099.445ca85b050bc@my.ceid.upatras.gr>
<7.0.1.0.1.20060506224156.01f868d8@live555.com>
<1147875340.446b300cb6f38@my.ceid.upatras.gr>
<1148308140.4471caacbc138@my.ceid.upatras.gr>
<1148508296.4474d888245e1@my.ceid.upatras.gr>
<7.0.1.0.1.20060524195000.01ec82d8@live555.com>
<1150194652.448e93dc35886@my.ceid.upatras.gr>
<7.0.1.0.1.20060613122349.01f4ddd8@live555.com>
Message-ID: <1150899885.449956ad37f30@my.ceid.upatras.gr>
>
> >There is a basic question I would like to ask. Concerning the above and the
> >ability to stop and play another file, which is the best class (file of the
> >source code) where I should implement this code ?
> >
> >Currently I am trying implementing this code in the
> >RTPTransmissionStats::noteIncomingRR() examining the RTCP report
> >packets sent by
> >the client, but I'am afraid of having problems concerning the scope and the
> >ability of communication between classes and objects.
> >
> >Would it be easy to try to implement the above code on the
> >RTPTransmissionStats::noteIncomingRR() function or this must be done
> somewhere
> >else ?
>
> I don't recommend modifying the supplied library code, if you don't
> have to (because that makes it difficult to upgrade to newer versions
> of the library code).
>
> In your case, I suggest using a separate "RTCP RR handler function"
> that gets called each time a new RTCP "RR" packet arrives, and using
> the "RTCPInstance::setRRHandler()" function to register it.
>
> (See "liveMedia/include/RTCP.hh")
>
>To change the input source for a running stream, you should do the
>following, in order:
> sink->stopPlaying();
> Medium::close(oldSource);
> create newSource
> sink->startPlaying(newSource, ...);
Hello Ross,
concerning the above code I should write, and the handler function, I would like
to ask the following two questions:
Which is the class that the "RTCP RR handler function" must belong to ?
Could you give me a bit explanation about how I should register the "RTCP RR
handler function" with the "RTCPInstance::setRRHandler()" ?
Many thanks
Best regards
----------------------------------------------------
This mail was sent through http://my.ceid.upatras.gr
From bidibulle at operamail.com Wed Jun 21 09:47:38 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Wed, 21 Jun 2006 17:47:38 +0100
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
Message-ID: <20060621164739.043C97B5F5@ws5-10.us4.outblaze.com>
Ross,
Please find here attached the patches corresponding to this issue.
Regards,
David
> Yes, please do. One of my long-planned goals for our RTSP server
> implementation has been to select ports from a small range of
> 'allowable' port numbers (just as DSS, Helix, etc. already do) - to
> make firewalling easier. This change will help make that possible.
>
>
> 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
>
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 8 at http://www.opera.com
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From bidibulle at operamail.com Wed Jun 21 09:48:50 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Wed, 21 Jun 2006 17:48:50 +0100
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
Message-ID: <20060621164850.903327B5D0@ws5-10.us4.outblaze.com>
Last patch ...
David
> Yes, please do. One of my long-planned goals for our RTSP server
> implementation has been to select ports from a small range of
> 'allowable' port numbers (just as DSS, Helix, etc. already do) - to
> make firewalling easier. This change will help make that possible.
>
>
> 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
>
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 8 at http://www.opera.com
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From finlayson at live555.com Wed Jun 21 12:24:48 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 21 Jun 2006 12:24:48 -0700
Subject: [Live-devel] [live-devel] receivingInterfaceAddress problem
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060621122321.01fee7c0@live555.com>
>What does it mean when after specify ReceivingInterfaceAddr =
>our_inet_addr("130.230.12.126") I get Unable
>to determine our source address: This computer has a invalid IP
>address 0x0. May you tell me what is the cause of that??
It seems that the function "ourSourceAddressForMulticast()" is
failing, after you made you change. You will need to trace through
the execution of that code (in "groupsock/GroupsockHelper.cpp") to
figure out why it fails for you.
Remember, You Have Complete Source Code.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Wed Jun 21 12:39:00 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 21 Jun 2006 12:39:00 -0700
Subject: [Live-devel] Groupsock handling in
OnDemandServerMediaSubsession.cpp
In-Reply-To: <20060621164739.043C97B5F5@ws5-10.us4.outblaze.com>
References: <20060621164739.043C97B5F5@ws5-10.us4.outblaze.com>
Message-ID: <7.0.1.0.1.20060621123245.01fae1b0@live555.com>
>Please find here attached the patches corresponding to this issue.
OK, thanks. I'm probably not going to make the changes quite the
same way that you have, because, IMHO, the SO_REUSE_* flags should be
set in almost all circumstances. The only exception that I see is
the case of creating a new socket with an arbitrarily chosen port
number (from some range) - so I'll probably add a new library
function that does specifically that.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From jiri.pinkava at vscht.cz Wed Jun 21 13:24:18 2006
From: jiri.pinkava at vscht.cz (jiri.pinkava at vscht.cz)
Date: Wed, 21 Jun 2006 22:24:18 +0200
Subject: [Live-devel] sapWatch print entire SAP
Message-ID: <20060621222418.9f79a094.jiri.pinkava@vscht.cz>
Hi all,
there is a small problem with sapWatch, it does not show entire SAP packet if payload type present (It is separated by \0... see RFC2974).
It is not most important but I do patch to solve this (otherwise it is unusable).
Ji?? Pinkava
--- sapWatch.cpp.orig 2006-05-18 00:06:50.000000000 +0200
+++ sapWatch.cpp 2006-06-21 22:18:55.000000000 +0200
@@ -59,6 +59,14 @@
continue;
}
+ // convert "application/sdp\0" -> "application/sdp\0x20"
+ // or all other nonprintable characters to blank, except new line
+ unsigned idx = 8;
+ while (idx < packetSize) {
+ if (packet[idx] < 0x20 && packet[idx] != '\n')
+ packet[idx] = 0x20;
+ idx++;
+ }
packet[packetSize] = '\0'; // just in case
printf((char*)(packet+8));
}
From jiri.pinkava at vscht.cz Wed Jun 21 13:30:34 2006
From: jiri.pinkava at vscht.cz (jiri.pinkava at vscht.cz)
Date: Wed, 21 Jun 2006 22:30:34 +0200
Subject: [Live-devel] SAP server
Message-ID: <20060621223034.c462fb49.jiri.pinkava@vscht.cz>
Hi,
there is no SAP server in live555, is in TODO add this feature into this library? What about this?
From finlayson at live555.com Wed Jun 21 13:37:06 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 21 Jun 2006 13:37:06 -0700
Subject: [Live-devel] sapWatch print entire SAP
In-Reply-To: <20060621222418.9f79a094.jiri.pinkava@vscht.cz>
References: <20060621222418.9f79a094.jiri.pinkava@vscht.cz>
Message-ID: <7.0.1.0.1.20060621133642.01fae1b0@live555.com>
Thanks. This change will be included in the next release of the software.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Wed Jun 21 13:45:27 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 21 Jun 2006 13:45:27 -0700
Subject: [Live-devel] SAP server
In-Reply-To: <20060621223034.c462fb49.jiri.pinkava@vscht.cz>
References: <20060621223034.c462fb49.jiri.pinkava@vscht.cz>
Message-ID: <7.0.1.0.1.20060621134402.01ff3238@live555.com>
At 01:30 PM 6/21/2006, you wrote:
>Hi,
> there is no SAP server in live555, is in TODO add this feature
> into this library? What about this?
SAP is rarely used these days. (Instead, RTSP, SIP, or just web
pages are more commonly used to distribute SDP descriptions.)
However, feel free to write a SAP server if you're interested.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From lorenooliveira at gmail.com Wed Jun 21 14:40:57 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Wed, 21 Jun 2006 18:40:57 -0300
Subject: [Live-devel] python
Message-ID:
Hi there,
does anybody knows some python binding for live555??
Loreno
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From spindler at arch.ethz.ch Wed Jun 21 14:45:33 2006
From: spindler at arch.ethz.ch (Torsten Spindler)
Date: Wed, 21 Jun 2006 23:45:33 +0200
Subject: [Live-devel] ffmpeg and live.com
Message-ID: <1150926334.3283.27.camel@didrachmon.ethz.ch>
Hello,
I use MLT (http://mlt.sourceforge.net) to apply some filters on a stream
sent by a Axis-230 MPEG-2 camera. The manipulated stream is then sent
via network further onward. So far I've tried the following:
MLT -> named pipe -> VLC -> multicast -> VLC
MLT -> udp unicast -> VLC -> multicast -> VLC
MLT -> named pipe -> testMPEG1or2VideoStreamer -> multicast -> VLC
MLT -> udp unicast -> testRelay -> multicast -> VLC
As the MLT/ffmpeg stream is quite bursty the resulting stream has either
a high latency (3-4 seconds) or artefacts appear.
Two questions:
1) Is there a patch for ffmpeg so it supports streaming via live.com?
Using MLT/ffmpeg/live in one application should reduce latency.
2) The MLT avformat consumer that stores the video either in a file or
sends it via udp uses av_interleaved_write_frame(oc, &pkt) with oc being
an AVFormatContext and pkt being AVPacket. If 1) is not possible, would
this be the place to hook in and store it for consummation by live.com
DeviceSource?
I'm really no C++ coder, so I'd very much prefer 2) :)
Bye,
Torsten
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From lorenooliveira at gmail.com Wed Jun 21 14:57:36 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Wed, 21 Jun 2006 18:57:36 -0300
Subject: [Live-devel] python
Message-ID:
Hi there,
does anybody knows some python binding for live555??
(sorry if this message is duplicated, I had some problem with my gmail)
Loreno
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From yetanotherid at gmail.com Wed Jun 21 15:08:39 2006
From: yetanotherid at gmail.com (S Mathur)
Date: Wed, 21 Jun 2006 18:08:39 -0400
Subject: [Live-devel] delete -> delete[] in DigestAuthentication.cpp
Message-ID:
Hi Ross,
In DigestAuthentication.cpp, the following 4 delete operations should
be changed to delete[]
void Authenticator::resetRealmAndNonce() {
delete fRealm; fRealm = NULL;
delete fNonce; fNonce = NULL;
}
void Authenticator::resetUsernameAndPassword() {
delete fUsername; fUsername = NULL;
delete fPassword; fPassword = NULL;
fPasswordIsMD5 = False;
}
All four were returned by strDup().
Thanks,
Mathur
From finlayson at live555.com Wed Jun 21 15:04:17 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 21 Jun 2006 15:04:17 -0700
Subject: [Live-devel] ffmpeg and live.com
In-Reply-To: <1150926334.3283.27.camel@didrachmon.ethz.ch>
References: <1150926334.3283.27.camel@didrachmon.ethz.ch>
Message-ID: <7.0.1.0.1.20060621145712.0209a0e0@live555.com>
>MLT -> named pipe -> VLC -> multicast -> VLC
>MLT -> udp unicast -> VLC -> multicast -> VLC
Note that when VLC is used as a *transmitter* (rather than as a
receiver), it does *not* (currently) use the "LIVE555 Streaming
Media" libraries.
>MLT -> named pipe -> testMPEG1or2VideoStreamer -> multicast -> VLC
This should work OK (provided, of course, that the input to
"testMPEG1or2VideoStreamer" is a MPEG-1 or 2 Video Elementary
Stream). You may need to tweak your named pipe somehow (don't ask me
how) to give it enough buffering, though.
>MLT -> udp unicast -> testRelay -> multicast -> VLC
The "testRelay" demo application receives a *multicast* stream. It
will (probably) not work if you try to feed it a unicast stream.
>1) Is there a patch for ffmpeg so it supports streaming via live.com?
Note: We're not "live.com" anymore.
It should not be necessary to 'patch' FFMPEG at all - it already
works just fine, as far as I can tell. However, if you want to use
it to generate a "FramedSource" subclass for delivering encoded data,
then you will need to write new code (using the "LIVE555 Streaming
Media" libraries) that does this. Read the FAQ.
>I'm really no C++ coder
That's a problem - you 'll need to fix this if you want to use our libraries.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Wed Jun 21 15:20:28 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 21 Jun 2006 15:20:28 -0700
Subject: [Live-devel] delete -> delete[] in DigestAuthentication.cpp
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060621152009.0200fa00@live555.com>
Thanks. This will get fixed in the next release.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From tooluck at tom.com Wed Jun 21 23:22:07 2006
From: tooluck at tom.com (tooluck at tom.com)
Date: Thu, 22 Jun 2006 14:22:07 +0800
Subject: [Live-devel] Fw: help me about livemedia
Message-ID: <004c01c695c4$3154d690$5801a8c0@luoqt>
hello,Ross
I have some questions about the "testOnDemandRTSPServer".
I test it use the command : rtsp://myip:554/XXX,the "XXX" is the streamname,and it works ok.
the XXX was preset in the rtsp server,but now ,I want to use rtsp://myip:554/XXX,and the XXX can be a filename or other,
and it was not set in the rtsp server,but the server could know how to do by the XXX.
for examlpe ,there are 3 files: 1.mp4,2.mp4,3.mp4
now I can play them with rtsp://myip:554/*.mp4 (the * is 1,2,or 3),and if i copy 4.mp4 to the floder
now i can use rtsp://myip:554/4.mp4,
how could i do it with livemedia.
another question,I found that in the each test application,the "scheduler" and the "env" had not to be delete,does it a memory leak bug?
thanks.
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From spindler at arch.ethz.ch Thu Jun 22 00:06:54 2006
From: spindler at arch.ethz.ch (Torsten Spindler)
Date: Thu, 22 Jun 2006 09:06:54 +0200
Subject: [Live-devel] ffmpeg and live.com
In-Reply-To: <7.0.1.0.1.20060621145712.0209a0e0@live555.com>
References: <1150926334.3283.27.camel@didrachmon.ethz.ch>
<7.0.1.0.1.20060621145712.0209a0e0@live555.com>
Message-ID: <1150960014.3165.13.camel@didrachmon.ethz.ch>
On Wed, 2006-06-21 at 15:04 -0700, Ross Finlayson wrote:
> >MLT -> udp unicast -> testRelay -> multicast -> VLC
>
> The "testRelay" demo application receives a *multicast* stream. It
> will (probably) not work if you try to feed it a unicast stream.
Works fine for me, there seems to be no problem using 127.0.0.1 3344 for
input and a multicast for output. I merely changed the input|output
AddressStr and input|output Port to take arguments via command line.
> Note: We're not "live.com" anymore.
Sorry, live555.com.
> It should not be necessary to 'patch' FFMPEG at all - it already
> works just fine, as far as I can tell.
It works in the above outlined ways, for sure!
> However, if you want to use
> it to generate a "FramedSource" subclass for delivering encoded data,
> then you will need to write new code (using the "LIVE555 Streaming
> Media" libraries) that does this. Read the FAQ.
After reading the FAQ I thought that there are some chances that someone
has already coupled ffmpeg with live555.com in this way. Guess I'll have
to take a closer look on how to do this ...
Bye,
Torsten
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From chthrivikram at yahoo.com Thu Jun 22 03:36:45 2006
From: chthrivikram at yahoo.com (thrivikram cheethirala)
Date: Thu, 22 Jun 2006 03:36:45 -0700 (PDT)
Subject: [Live-devel] want testMPEG1or2VideoReceiver.cpp
Message-ID: <20060622103645.93389.qmail@web30507.mail.mud.yahoo.com>
Hi,
I have looked into your open source library
"LiveMedia". when i was going through your library i
have seen the testMPEG1or2VideoReceiver.cpp as an
example to work with mpeg1 and mpeg2. similarly if you
could provide me the test file for mpeg4. it will be
of a great help.
regards,
vikram.
thrivikramarao chitirala,
Senior Software Engineer,
RTG,Honeywell Technologies,
Bangalore-76.
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From chthrivikram at yahoo.com Thu Jun 22 07:03:56 2006
From: chthrivikram at yahoo.com (thrivikram cheethirala)
Date: Thu, 22 Jun 2006 07:03:56 -0700 (PDT)
Subject: [Live-devel] Problem in running testPrograms
Message-ID: <20060622140356.63813.qmail@web30510.mail.mud.yahoo.com>
Hi,
When i tried to run testMPEG1or2Reciever, i have
given my command like g++ testMPEG1or2VideoReciever -l
../livemedia/liblivemedia
the message was /usr/bin/ld : can't find
-l../livemedia/liblivemedia.a
i checked for spelling mistakes, there were no such.
if i try with out that l(ell) option it's givving me a
lsit of messages saying undefined reference to
functions.
one such is
: undefined reference to 'strDup(char const*)'
any help is greatly appreciated.
__________________________________________________
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From chthrivikram at yahoo.com Thu Jun 22 07:09:53 2006
From: chthrivikram at yahoo.com (thrivikram cheethirala)
Date: Thu, 22 Jun 2006 07:09:53 -0700 (PDT)
Subject: [Live-devel] Compilation problem
Message-ID: <20060622140953.20036.qmail@web30512.mail.mud.yahoo.com>
Here is one such problem
# g++ testMPEG1or2VideoReceiver.cpp -I
../liveMedia/include -I
/home/NEOS/live/groupsock/include/ -I
/home/NEOS/live/UsageEnvironment/include -I
/home/NEOS/live/BasicUsageEnvironment/include -L
/home/NEOS/live/liveMedia/libliveMedia.a
/tmp/cce7JdwA.o(.text+0x21): In function `main':
testMPEG1or2VideoReceiver.cpp: undefined reference to
`BasicTaskScheduler::createNew()'
/tmp/cce7JdwA.o(.text+0x2f):testMPEG1or2VideoReceiver.cpp:
undefined reference to
`BasicUsageEnvironment::createNew(TaskScheduler&)'
/tmp/cce7JdwA.o(.text+0x4e):testMPEG1or2VideoReceiver.cpp:
undefined reference to
`FileSink::createNew(UsageEnvironment&, char const*,
unsigned int, unsigned int)'
/tmp/cce7JdwA.o(.text+0x78):testMPEG1or2VideoReceiver.cpp:
undefined reference to `our_inet_addr'
/tmp/cce7JdwA.o(.text+0x8f):testMPEG1or2VideoReceiver.cpp:
undefined reference to `Port::Port(unsigned short)'
/tmp/cce7JdwA.o(.text+0xa3):testMPEG1or2VideoReceiver.cpp:
undefined reference to `Port::Port(unsigned short)'
/tmp/cce7JdwA.o(.text+0xcb):testMPEG1or2VideoReceiver.cpp:
undefined reference to
`Groupsock::Groupsock(UsageEnvironment&, in_addr
const&, Port, unsigned char)'
/tmp/cce7JdwA.o(.text+0xf2):testMPEG1or2VideoReceiver.cpp:
undefined reference to
`Groupsock::Groupsock(UsageEnvironment&, in_addr
const&, Port, unsigned char)'
/tmp/cce7JdwA.o(.text+0x10f):testMPEG1or2VideoReceiver.cpp:
undefined reference to
`MPEG1or2VideoRTPSource::createNew(UsageEnvironment&,
Groupsock*, unsigned char, unsigned int)'
/tmp/cce7JdwA.o(.text+0x16c):testMPEG1or2VideoReceiver.cpp:
undefined reference to
`RTCPInstance::createNew(UsageEnvironment&,
Groupsock*, unsigned int, unsigned char const*,
RTPSink*, RTPSource const*, unsigned int)'
/tmp/cce7JdwA.o(.text+0x1ac):testMPEG1or2VideoReceiver.cpp:
undefined reference to
`MediaSink::startPlaying(MediaSource&, void
(*)(void*), void*)'
/tmp/cce7JdwA.o(.text+0x1ed):testMPEG1or2VideoReceiver.cpp:
undefined reference to `Groupsock::~Groupsock()'
/tmp/cce7JdwA.o(.text+0x20d):testMPEG1or2VideoReceiver.cpp:
undefined reference to `Groupsock::~Groupsock()'
/tmp/cce7JdwA.o(.text+0x233):testMPEG1or2VideoReceiver.cpp:
undefined reference to `Groupsock::~Groupsock()'
/tmp/cce7JdwA.o(.text+0x259):testMPEG1or2VideoReceiver.cpp:
undefined reference to `Groupsock::~Groupsock()'
/tmp/cce7JdwA.o(.text+0x29b): In function
`afterPlaying(void*)':
testMPEG1or2VideoReceiver.cpp: undefined reference to
`Medium::close(Medium*)'
/tmp/cce7JdwA.o(.text+0x2ac):testMPEG1or2VideoReceiver.cpp:
undefined reference to `Medium::close(Medium*)'
/tmp/cce7JdwA.o(.text+0x2bd):testMPEG1or2VideoReceiver.cpp:
undefined reference to `Medium::close(Medium*)'
/tmp/cce7JdwA.o(.text+0x2e8): In function
`__static_initialization_and_destruction_0(int, int)':
testMPEG1or2VideoReceiver.cpp: undefined reference to
`SECOND'
/tmp/cce7JdwA.o(.text+0x2f0):testMPEG1or2VideoReceiver.cpp:
undefined reference to `operator*(short, DelayInterval
const&)'
/tmp/cce7JdwA.o(.text+0x326):testMPEG1or2VideoReceiver.cpp:
undefined reference to `operator*(short, DelayInterval
const&)'
/tmp/cce7JdwA.o(.text+0x35c):testMPEG1or2VideoReceiver.cpp:
undefined reference to `operator*(short, DelayInterval
const&)'
collect2: ld returned 1 exit status
Please Help me
thrivikramarao chitirala,
Senior Software Engineer,
RTG,Honeywell Technologies,
Bangalore-76.
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From sdhays.neon.com.tw at gmail.com Thu Jun 22 19:31:21 2006
From: sdhays.neon.com.tw at gmail.com (Scott Hays)
Date: Fri, 23 Jun 2006 10:31:21 +0800
Subject: [Live-devel] Compilation problem
In-Reply-To: <20060622140953.20036.qmail@web30512.mail.mud.yahoo.com>
References: <20060622140953.20036.qmail@web30512.mail.mud.yahoo.com>
Message-ID: <9866ce4f0606221931j28a4162ag4809d6523b0dda7e@mail.gmail.com>
You aren't including all of the libraries.
Scott
On 6/22/06, thrivikram cheethirala wrote:
>
> Here is one such problem
>
> # g++ testMPEG1or2VideoReceiver.cpp -I
> ../liveMedia/include -I
> /home/NEOS/live/groupsock/include/ -I
> /home/NEOS/live/UsageEnvironment/include -I
> /home/NEOS/live/BasicUsageEnvironment/include -L
> /home/NEOS/live/liveMedia/libliveMedia.a
> /tmp/cce7JdwA.o(.text+0x21): In function `main':
> testMPEG1or2VideoReceiver.cpp: undefined reference to
> `BasicTaskScheduler::createNew()'
> /tmp/cce7JdwA.o(.text+0x2f):testMPEG1or2VideoReceiver.cpp:
> undefined reference to
> `BasicUsageEnvironment::createNew(TaskScheduler&)'
> /tmp/cce7JdwA.o(.text+0x4e):testMPEG1or2VideoReceiver.cpp:
> undefined reference to
> `FileSink::createNew(UsageEnvironment&, char const*,
> unsigned int, unsigned int)'
> /tmp/cce7JdwA.o(.text+0x78):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `our_inet_addr'
> /tmp/cce7JdwA.o(.text+0x8f):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `Port::Port(unsigned short)'
> /tmp/cce7JdwA.o(.text+0xa3):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `Port::Port(unsigned short)'
> /tmp/cce7JdwA.o(.text+0xcb):testMPEG1or2VideoReceiver.cpp:
> undefined reference to
> `Groupsock::Groupsock(UsageEnvironment&, in_addr
> const&, Port, unsigned char)'
> /tmp/cce7JdwA.o(.text+0xf2):testMPEG1or2VideoReceiver.cpp:
> undefined reference to
> `Groupsock::Groupsock(UsageEnvironment&, in_addr
> const&, Port, unsigned char)'
> /tmp/cce7JdwA.o(.text+0x10f):testMPEG1or2VideoReceiver.cpp:
> undefined reference to
> `MPEG1or2VideoRTPSource::createNew(UsageEnvironment&,
> Groupsock*, unsigned char, unsigned int)'
> /tmp/cce7JdwA.o(.text+0x16c):testMPEG1or2VideoReceiver.cpp:
> undefined reference to
> `RTCPInstance::createNew(UsageEnvironment&,
> Groupsock*, unsigned int, unsigned char const*,
> RTPSink*, RTPSource const*, unsigned int)'
> /tmp/cce7JdwA.o(.text+0x1ac):testMPEG1or2VideoReceiver.cpp:
> undefined reference to
> `MediaSink::startPlaying(MediaSource&, void
> (*)(void*), void*)'
> /tmp/cce7JdwA.o(.text+0x1ed):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `Groupsock::~Groupsock()'
> /tmp/cce7JdwA.o(.text+0x20d):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `Groupsock::~Groupsock()'
> /tmp/cce7JdwA.o(.text+0x233):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `Groupsock::~Groupsock()'
> /tmp/cce7JdwA.o(.text+0x259):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `Groupsock::~Groupsock()'
> /tmp/cce7JdwA.o(.text+0x29b): In function
> `afterPlaying(void*)':
> testMPEG1or2VideoReceiver.cpp: undefined reference to
> `Medium::close(Medium*)'
> /tmp/cce7JdwA.o(.text+0x2ac):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `Medium::close(Medium*)'
> /tmp/cce7JdwA.o(.text+0x2bd):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `Medium::close(Medium*)'
> /tmp/cce7JdwA.o(.text+0x2e8): In function
> `__static_initialization_and_destruction_0(int, int)':
> testMPEG1or2VideoReceiver.cpp: undefined reference to
> `SECOND'
> /tmp/cce7JdwA.o(.text+0x2f0):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `operator*(short, DelayInterval
> const&)'
> /tmp/cce7JdwA.o(.text+0x326):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `operator*(short, DelayInterval
> const&)'
> /tmp/cce7JdwA.o(.text+0x35c):testMPEG1or2VideoReceiver.cpp:
> undefined reference to `operator*(short, DelayInterval
> const&)'
> collect2: ld returned 1 exit status
>
> Please Help me
>
>
> thrivikramarao chitirala,
> Senior Software Engineer,
> RTG,Honeywell Technologies,
> Bangalore-76.
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
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From chthrivikram at yahoo.com Thu Jun 22 21:08:28 2006
From: chthrivikram at yahoo.com (thrivikram cheethirala)
Date: Thu, 22 Jun 2006 21:08:28 -0700 (PDT)
Subject: [Live-devel] Compilation problem
In-Reply-To: <9866ce4f0606221931j28a4162ag4809d6523b0dda7e@mail.gmail.com>
Message-ID: <20060623040828.27795.qmail@web30503.mail.mud.yahoo.com>
Thanks Scot. I have included libLivemdia,
libBasicUsage,libgroupsock,libUsageEnv,
am i still missing any more libraries.
regards,
vikram.
--- Scott Hays wrote:
> You aren't including all of the libraries.
> Scott
>
> On 6/22/06, thrivikram cheethirala
> wrote:
> >
> > Here is one such problem
> >
> > # g++ testMPEG1or2VideoReceiver.cpp -I
> > ../liveMedia/include -I
> > /home/NEOS/live/groupsock/include/ -I
> > /home/NEOS/live/UsageEnvironment/include -I
> > /home/NEOS/live/BasicUsageEnvironment/include -L
> > /home/NEOS/live/liveMedia/libliveMedia.a
> > /tmp/cce7JdwA.o(.text+0x21): In function `main':
> > testMPEG1or2VideoReceiver.cpp: undefined reference
> to
> > `BasicTaskScheduler::createNew()'
> >
>
/tmp/cce7JdwA.o(.text+0x2f):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to
> > `BasicUsageEnvironment::createNew(TaskScheduler&)'
> >
>
/tmp/cce7JdwA.o(.text+0x4e):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to
> > `FileSink::createNew(UsageEnvironment&, char
> const*,
> > unsigned int, unsigned int)'
> >
>
/tmp/cce7JdwA.o(.text+0x78):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `our_inet_addr'
> >
>
/tmp/cce7JdwA.o(.text+0x8f):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `Port::Port(unsigned
> short)'
> >
>
/tmp/cce7JdwA.o(.text+0xa3):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `Port::Port(unsigned
> short)'
> >
>
/tmp/cce7JdwA.o(.text+0xcb):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to
> > `Groupsock::Groupsock(UsageEnvironment&, in_addr
> > const&, Port, unsigned char)'
> >
>
/tmp/cce7JdwA.o(.text+0xf2):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to
> > `Groupsock::Groupsock(UsageEnvironment&, in_addr
> > const&, Port, unsigned char)'
> >
>
/tmp/cce7JdwA.o(.text+0x10f):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to
> >
>
`MPEG1or2VideoRTPSource::createNew(UsageEnvironment&,
> > Groupsock*, unsigned char, unsigned int)'
> >
>
/tmp/cce7JdwA.o(.text+0x16c):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to
> > `RTCPInstance::createNew(UsageEnvironment&,
> > Groupsock*, unsigned int, unsigned char const*,
> > RTPSink*, RTPSource const*, unsigned int)'
> >
>
/tmp/cce7JdwA.o(.text+0x1ac):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to
> > `MediaSink::startPlaying(MediaSource&, void
> > (*)(void*), void*)'
> >
>
/tmp/cce7JdwA.o(.text+0x1ed):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `Groupsock::~Groupsock()'
> >
>
/tmp/cce7JdwA.o(.text+0x20d):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `Groupsock::~Groupsock()'
> >
>
/tmp/cce7JdwA.o(.text+0x233):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `Groupsock::~Groupsock()'
> >
>
/tmp/cce7JdwA.o(.text+0x259):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `Groupsock::~Groupsock()'
> > /tmp/cce7JdwA.o(.text+0x29b): In function
> > `afterPlaying(void*)':
> > testMPEG1or2VideoReceiver.cpp: undefined reference
> to
> > `Medium::close(Medium*)'
> >
>
/tmp/cce7JdwA.o(.text+0x2ac):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `Medium::close(Medium*)'
> >
>
/tmp/cce7JdwA.o(.text+0x2bd):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `Medium::close(Medium*)'
> > /tmp/cce7JdwA.o(.text+0x2e8): In function
> > `__static_initialization_and_destruction_0(int,
> int)':
> > testMPEG1or2VideoReceiver.cpp: undefined reference
> to
> > `SECOND'
> >
>
/tmp/cce7JdwA.o(.text+0x2f0):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `operator*(short,
> DelayInterval
> > const&)'
> >
>
/tmp/cce7JdwA.o(.text+0x326):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `operator*(short,
> DelayInterval
> > const&)'
> >
>
/tmp/cce7JdwA.o(.text+0x35c):testMPEG1or2VideoReceiver.cpp:
> > undefined reference to `operator*(short,
> DelayInterval
> > const&)'
> > collect2: ld returned 1 exit status
> >
> > Please Help me
> >
> >
> > thrivikramarao chitirala,
> > Senior Software Engineer,
> > RTG,Honeywell Technologies,
> > Bangalore-76.
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam? Yahoo! Mail has the best spam
> protection around
> > http://mail.yahoo.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
>
thrivikramarao chitirala,
Senior Software Engineer,
RTG,Honeywell Technologies,
Bangalore-76.
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
From jiri.pinkava at vscht.cz Fri Jun 23 16:32:40 2006
From: jiri.pinkava at vscht.cz (jiri.pinkava at vscht.cz)
Date: Sat, 24 Jun 2006 01:32:40 +0200
Subject: [Live-devel] PES packet size
Message-ID: <20060624013240.8ded2da0.jiri.pinkava@vscht.cz>
There look like a small bug in MPEG2TransportStreamFromPESSource.cpp
where is specified buffer for PES packet of lenght 65536 (correct is 65535+6).
(but who use packets of this lenght???)
diff -ru live/liveMedia/MPEG2TransportStreamFromPESSource.cpp live-2006.05.17/liveMedia/MPEG2TransportStreamFromPESSource.cpp
--- live/liveMedia/MPEG2TransportStreamFromPESSource.cpp 2005-12-29 14:22:35.000000000 +0100
+++ live-2006.05.17/liveMedia/MPEG2TransportStreamFromPESSource.cpp 2006-05-18 00:06:50.000000000 +0200
@@ -20,7 +20,7 @@
#include "MPEG2TransportStreamFromPESSource.hh"
-#define MAX_PES_PACKET_SIZE 65536
+#define MAX_PES_PACKET_SIZE (65535 + 6)
MPEG2TransportStreamFromPESSource* MPEG2TransportStreamFromPESSource
::createNew(UsageEnvironment& env, MPEG1or2DemuxedElementaryStream* inputSource) {
From finlayson at live555.com Sat Jun 24 00:03:11 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sat, 24 Jun 2006 00:03:11 -0700
Subject: [Live-devel] PES packet size
In-Reply-To: <20060624013240.8ded2da0.jiri.pinkava@vscht.cz>
References: <20060624013240.8ded2da0.jiri.pinkava@vscht.cz>
Message-ID: <7.0.1.0.1.20060624000243.01f9ebb0@live555.com>
At 04:32 PM 6/23/2006, you wrote:
>There look like a small bug in MPEG2TransportStreamFromPESSource.cpp
>where is specified buffer for PES packet of lenght 65536 (correct is 65535+6).
Thanks. This fix will be included in the next release.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From morgan.torvolt at gmail.com Sun Jun 25 03:31:15 2006
From: morgan.torvolt at gmail.com (=?ISO-8859-1?Q?Morgan_T=F8rvolt?=)
Date: Sun, 25 Jun 2006 14:31:15 +0400
Subject: [Live-devel] testMPEG2TransportStreamer to Amino STB
Message-ID: <3cc3561f0606250331y23db0012r812fa40ab9bd2b5e@mail.gmail.com>
Hi guys
I have been trying to send a ts file to a Amino STB for a while now.
Using testMPEG2TransportStreamer with VLC as a reciever works great,
but to Amino STB does not work. All I get is som garbage on the top
part of the picture (max one macroblock in size), and sometimes large
green areas in the top half of the screen. Using VLC as a source works
great with the STB. I have tried unicast and multicast, both without
much success. Since RTSP to STB uses UDP, and that works perfectly
with my Amino box, I would expect that Live is capable of doing this
job, I just don't know how. Has anyone gotten this to work? How?
Thanks
-Morgan T?rvolt-
From soaring_roc at yahoo.com.cn Sun Jun 25 04:46:17 2006
From: soaring_roc at yahoo.com.cn (=?gb2312?q?=D0=A1=B2=A8=20=CC=EF?=)
Date: Sun, 25 Jun 2006 19:46:17 +0800 (CST)
Subject: [Live-devel] A problem about openRTSP
Message-ID: <20060625114617.28035.qmail@web15705.mail.cnb.yahoo.com>
Hi, folks
I use the test program(openRTSP) of liveMedia, the server is DSS. I use it as this: ./openRTSP -V rtsp://xxx.xxx.xxx.xxx/sample_100kbit.mp4 the program can record the Audio and Video tracks in the file named"audio-MPEG4-GENERIC-2" and "video-MP4V-ES-1", follow the discription of the openRTSP like this:
"The received data for each subsession is written into a separate output file, named according to its MIME type. For example, if the session contains a MPEG-1 or 2 audio subsession (RTP payload type 14) - e.g., MP3 - and a MPEG-1 or 2 video subsession (RTP payload type 32), then each subsession's data will be extracted from the incoming RTP packets and written to files named "audio-MPA-1" and "video-MPV-2" (respectively). (You will probably then need to rename these files - by giving them an appropriate filename extension (e.g., ".mp3" and ".mpg") - in order to be able to play them using common media player tools.) "
I add the extension ".mp4" to the file "video-MP4V-ES-1" result in "video-MP4V-ES-1.mp4" follow the discription. then I use the QuickTime player to play it, but it can't work with the error like:"the video-MP4V-ES-1.mp4 is not a valid MPEG-4 file, No 'moov' autom could be found". I am puzzled about the problem. any suggestion would be appreciated! Thanks!
Here is the message when the openRTSP works:
sending request: OPTIONS rtsp://192.168.0.68/sample_100kbit.mp4 RTSP/1.0
CSeq: 1
User-Agent: ./openRTSP (LIVE555 Streaming Media v2006.05.17)
Received OPTIONS response: RTSP/1.0 200 OK
Server: DSS/5.5 (Build/489.7; Platform/Linux; Release/Darwin; )
Cseq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, OPTIONS, ANNOUNCE, RECORD
Sending request: DESCRIBE rtsp://192.168.0.68/sample_100kbit.mp4 RTSP/1.0
CSeq: 2
Accept: application/sdp
User-Agent: ./openRTSP (LIVE555 Streaming Media v2006.05.17)
Received DESCRIBE response: RTSP/1.0 200 OK
Server: DSS/5.5 (Build/489.7; Platform/Linux; Release/Darwin; )
Cseq: 2
Last-Modified: Wed, 31 May 2006 07:28:00 GMT
Cache-Control: must-revalidate
Content-length: 1206
Date: Fri, 23 Jun 2006 13:32:03 GMT
Expires: Fri, 23 Jun 2006 13:32:03 GMT
Content-Type: application/sdp
x-Accept-Retransmit: our-retransmit
x-Accept-Dynamic-Rate: 1
Content-Base: rtsp://192.168.0.68/sample_100kbit.mp4/
Need to read 1206 extra bytes
Read 1206 extra bytes: v=0
o=StreamingServer 3360058323 1149060480000 IN IP4 192.168.0.68
s=/sample_100kbit.mp4
u=http:///
e=admin@
c=IN IP4 0.0.0.0
b=AS:96
t=0 0
a=control:*
a=mpeg4-iod:"data:application/mpeg4-iod;base64,AoJrAE///w/z/wOBdgABQNhkYXRhOmFwcGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBWUVDQVV3Rkh3TklBTWtnQUdVRUx5QVJBRzNkQUFFaytBQUJKUGdGSUFBQUFiRHpBQUFCdFE3Z1FNRFBBQUFCQUFBQUFTQUFoRUQ2S0RBZzhxSWZCaEFBUkFBQUFsZ0FBQUFBSUFBQUFBQURBVElDbndNdUFHVUFCSUNBZ0JSQUZRQVlBQUFBVGlBQUFFNGdCWUNBZ0FJVmtBWVFBRVFBQUI5QUFBQWZRQ0FnQUFBQUF3PT0EDQEFAADIAAAAAAAAAAAGCQEAAAAAAAAAAANpAAJARmRhdGE6YXBwbGljYXRpb24vbXBlZzQtYmlmcy1hdTtiYXNlNjQsd0JBU2daTUNvRmNtRUVIOEFBQUIvQUFBQkVLQ0tDbjQEEgINAABkAAAAAAAAAAAFAwAAYAYJAQAAAAAAAAAA"
a=isma-compliance:1,1.0,1
a=range:npt=0- 70.00000
m=video 0 RTP/AVP 96
b=AS:76
a=rtpmap:96 MP4V-ES/90000
a=control:trackID=3
a=cliprect:0,0,242,192
a=framesize:96 192-242
a=fmtp:96 profile-level-id=1;config=000001B0F3000001B50EE040C0CF0000010000000120008440FA283020F2A21F
a=mpeg4-esid:201
m=audio 0 RTP/AVP 97
b=AS:20
a=rtpmap:97 mpeg4-generic/8000/2
a=control:trackID=4
a=fmtp:97 profile-level-id=15;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1590
a=mpeg4-esid:101
Opened URL "rtsp://192.168.0.68/sample_100kbit.mp4", returning a SDP description:
v=0
o=StreamingServer 3360058323 1149060480000 IN IP4 192.168.0.68
s=/sample_100kbit.mp4
u=http:///
e=admin@
c=IN IP4 0.0.0.0
b=AS:96
t=0 0
a=control:*
a=mpeg4-iod:"data:application/mpeg4-iod;base64,AoJrAE///w/z/wOBdgABQNhkYXRhOmFwcGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBWUVDQVV3Rkh3TklBTWtnQUdVRUx5QVJBRzNkQUFFaytBQUJKUGdGSUFBQUFiRHpBQUFCdFE3Z1FNRFBBQUFCQUFBQUFTQUFoRUQ2S0RBZzhxSWZCaEFBUkFBQUFsZ0FBQUFBSUFBQUFBQURBVElDbndNdUFHVUFCSUNBZ0JSQUZRQVlBQUFBVGlBQUFFNGdCWUNBZ0FJVmtBWVFBRVFBQUI5QUFBQWZRQ0FnQUFBQUF3PT0EDQEFAADIAAAAAAAAAAAGCQEAAAAAAAAAAANpAAJARmRhdGE6YXBwbGljYXRpb24vbXBlZzQtYmlmcy1hdTtiYXNlNjQsd0JBU2daTUNvRmNtRUVIOEFBQUIvQUFBQkVLQ0tDbjQEEgINAABkAAAAAAAAAAAFAwAAYAYJAQAAAAAAAAAA"
a=isma-compliance:1,1.0,1
a=range:npt=0- 70.00000
m=video 0 RTP/AVP 96
b=AS:76
a=rtpmap:96 MP4V-ES/90000
a=control:trackID=3
a=cliprect:0,0,242,192
a=framesize:96 192-242
a=fmtp:96 profile-level-id=1;config=000001B0F3000001B50EE040C0CF0000010000000120008440FA283020F2A21F
a=mpeg4-esid:201
m=audio 0 RTP/AVP 97
b=AS:20
a=rtpmap:97 mpeg4-generic/8000/2
a=control:trackID=4
a=fmtp:97 profile-level-id=15;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1590
a=mpeg4-esid:101
Created receiver for "video/MP4V-ES" subsession (client ports 32772-32773)
Created receiver for "audio/MPEG4-GENERIC" subsession (client ports 32774-32775)Sending request: SETUP rtsp://192.168.0.68/sample_100kbit.mp4/trackID=3 RTSP/1.0CSeq: 3
Transport: RTP/AVP;unicast;client_port=32772-32773
User-Agent: ./openRTSP (LIVE555 Streaming Media v2006.05.17)
Received SETUP response: RTSP/1.0 200 OK
Server: DSS/5.5 (Build/489.7; Platform/Linux; Release/Darwin; )
Cseq: 3
Last-Modified: Wed, 31 May 2006 07:28:00 GMT
Cache-Control: must-revalidate
Session: 584703331794683763
Date: Fri, 23 Jun 2006 13:32:03 GMT
Expires: Fri, 23 Jun 2006 13:32:03 GMT
Transport: RTP/AVP;unicast;source=192.168.0.68;client_port=32772-32773;server_port=6970-6971;ssrc=4F35562D
Setup "video/MP4V-ES" subsession (client ports 32772-32773)
Sending request: SETUP rtsp://192.168.0.68/sample_100kbit.mp4/trackID=4 RTSP/1.0CSeq: 4
Transport: RTP/AVP;unicast;client_port=32774-32775
Session: 584703331794683763
User-Agent: ./openRTSP (LIVE555 Streaming Media v2006.05.17)
Received SETUP response: RTSP/1.0 200 OK
Server: DSS/5.5 (Build/489.7; Platform/Linux; Release/Darwin; )
Cseq: 4
Session: 584703331794683763
Last-Modified: Wed, 31 May 2006 07:28:00 GMT
Cache-Control: must-revalidate
Date: Fri, 23 Jun 2006 13:32:03 GMT
Expires: Fri, 23 Jun 2006 13:32:03 GMT
Transport: RTP/AVP;unicast;source=192.168.0.68;client_port=32774-32775;server_port=6970-6971;ssrc=6367F43F
Setup "audio/MPEG4-GENERIC" subsession (client ports 32774-32775)
Created output file: "video-MP4V-ES-1"
Created output file: "audio-MPEG4-GENERIC-2"
Sending request: PLAY rtsp://192.168.0.68/sample_100kbit.mp4 RTSP/1.0
CSeq: 5
Session: 584703331794683763
Range: npt=0.000-
User-Agent: ./openRTSP (LIVE555 Streaming Media v2006.05.17)
Received PLAY response: RTSP/1.0 200 OK
Server: DSS/5.5 (Build/489.7; Platform/Linux; Release/Darwin; )
Cseq: 5
Session: 584703331794683763
Range: npt=0.00000-70.00000
RTP-Info: url=rtsp://192.168.0.68/sample_100kbit.mp4/trackID=3;seq=12166;rtptime=1202588821,url=rtsp://192.168.0.68/sample_100kbit.mp4/trackID=4;seq=17381;rtptime=776601757
Started playing session
Receiving streamed data (for up to 75.000000 seconds)...
Sending request: TEARDOWN rtsp://192.168.0.68/sample_100kbit.mp4 RTSP/1.0
CSeq: 6
Session: 584703331794683763
User-Agent: ./openRTSP (LIVE555 Streaming Media v2006.05.17)
Received TEARDOWN response: RTSP/1.0 200 OK
Server: DSS/5.5 (Build/489.7; Platform/Linux; Release/Darwin; )
Cseq: 6
Session: 584703331794683763
Connection: Close
Xiaobo Tian
---------------------------------
????????-3.5G???20M???
---------------------------------
????????-3.5G???20M???
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From finlayson at live555.com Sun Jun 25 08:18:02 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 25 Jun 2006 08:18:02 -0700
Subject: [Live-devel] A problem about openRTSP
In-Reply-To: <20060625114617.28035.qmail@web15705.mail.cnb.yahoo.com>
References: <20060625114617.28035.qmail@web15705.mail.cnb.yahoo.com>
Message-ID: <7.0.1.0.1.20060625081330.01fcbe08@live555.com>
> I add the extension ".mp4" to the file "video-MP4V-ES-1"
No, that won't work, because
1/ The ".mp4" file extension denotes a 'MPEG-4 format' file
- *not* a MPEG-4 Video Elementary Stream file.
2/ QuickTime Player cannot play MPEG-4 Video Elementary Stream files
> result in "video-MP4V-ES-1.mp4" follow the discription. then I use
> the QuickTime player to play it, but it can't work with the error
> like:"the video-MP4V-ES-1.mp4 is not a valid MPEG-4 file, No 'moov'
> autom could be found".
The error message is correct. It means exactly what it says.
> I am puzzled about the problem. any suggestion would be appreciated!
To output a proper ".mp4' file (to stdout), use the "-4" option. See
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Sun Jun 25 08:22:27 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 25 Jun 2006 08:22:27 -0700
Subject: [Live-devel] testMPEG2TransportStreamer to Amino STB
In-Reply-To: <3cc3561f0606250331y23db0012r812fa40ab9bd2b5e@mail.gmail.co
m>
References: <3cc3561f0606250331y23db0012r812fa40ab9bd2b5e@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060625081814.02030070@live555.com>
>I have been trying to send a ts file to a Amino STB for a while now.
>Using testMPEG2TransportStreamer with VLC as a reciever works great,
>but to Amino STB does not work.
That's because "testMPEG2TransportStreamer" streams using multicast,
which the Amino STB does not handle.
>I have tried unicast and multicast, both without
>much success.
You can't (easily) modify the "testMPEG2TransportStreamer" to use
unicast. (Just changing the destination address won't be enough,
because the RTSP server still expects to send multicast.)
To stream using unicast, use the "testOnDemandRTSPServer" demo
application. It works with the Amino STB - see
http://www.live555.com/liveMedia/faq.html#stb>
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From morgan.torvolt at gmail.com Sun Jun 25 10:22:08 2006
From: morgan.torvolt at gmail.com (=?ISO-8859-1?Q?Morgan_T=F8rvolt?=)
Date: Sun, 25 Jun 2006 21:22:08 +0400
Subject: [Live-devel] testMPEG2TransportStreamer to Amino STB
In-Reply-To: <7.0.1.0.1.20060625081814.02030070@live555.com>
References: <3cc3561f0606250331y23db0012r812fa40ab9bd2b5e@mail.gmail.com>
<7.0.1.0.1.20060625081814.02030070@live555.com>
Message-ID: <3cc3561f0606251022p688db9e6i1810fbb3b54b3f94@mail.gmail.com>
Hi again. Thanks for the response.
Is there not a way to use Live555 for unicast UDP streaming? In the
testMPEG2TransportStreamer.cpp source code I find this:
// Note: This is a multicast address. If you wish to stream using
// unicast instead, then replace this string with the unicast address
// of the (single) destination. (You may also need to make a similar
// change to the receiver program.)
Just below the IP address. Both unicast works flawlessly and multicast
works partially (I think the problem is my switch actually) from VLC,
so the STB definately knows how to handle both multicast and unicast.
I use these URLs on the STB:
(Streaming server IP: 10.0.0.20, STB IP: 10.0.0.25)
unicast: udp://10.0.0.20:1234
multicast: igmp://239.255.0.1:1234
And of course set the destination IP and port in VLC.
When I use your program i get a little garbage in the top part of my
TV, but that is all.
-Morgan T?rvolt-
On 6/25/06, Ross Finlayson wrote:
>
> >I have been trying to send a ts file to a Amino STB for a while now.
> >Using testMPEG2TransportStreamer with VLC as a reciever works great,
> >but to Amino STB does not work.
>
> That's because "testMPEG2TransportStreamer" streams using multicast,
> which the Amino STB does not handle.
>
> >I have tried unicast and multicast, both without
> >much success.
>
> You can't (easily) modify the "testMPEG2TransportStreamer" to use
> unicast. (Just changing the destination address won't be enough,
> because the RTSP server still expects to send multicast.)
>
> To stream using unicast, use the "testOnDemandRTSPServer" demo
> application. It works with the Amino STB - see
> http://www.live555.com/liveMedia/faq.html#stb>
>
>
> 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
>
From morgan.torvolt at gmail.com Sun Jun 25 10:23:16 2006
From: morgan.torvolt at gmail.com (=?ISO-8859-1?Q?Morgan_T=F8rvolt?=)
Date: Sun, 25 Jun 2006 21:23:16 +0400
Subject: [Live-devel] testMPEG2TransportStreamer to Amino STB
In-Reply-To: <3cc3561f0606251022p688db9e6i1810fbb3b54b3f94@mail.gmail.com>
References: <3cc3561f0606250331y23db0012r812fa40ab9bd2b5e@mail.gmail.com>
<7.0.1.0.1.20060625081814.02030070@live555.com>
<3cc3561f0606251022p688db9e6i1810fbb3b54b3f94@mail.gmail.com>
Message-ID: <3cc3561f0606251023n15e3fe70m729742d21250216e@mail.gmail.com>
BTW. Streaming unicast from testMPEG2TransportStreamer to my computer,
and using VLC for recieve works perfectly.
-Morgan T?rvolt-
On 6/25/06, Morgan T?rvolt wrote:
> Hi again. Thanks for the response.
>
> Is there not a way to use Live555 for unicast UDP streaming? In the
> testMPEG2TransportStreamer.cpp source code I find this:
>
> // Note: This is a multicast address. If you wish to stream using
> // unicast instead, then replace this string with the unicast address
> // of the (single) destination. (You may also need to make a similar
> // change to the receiver program.)
>
> Just below the IP address. Both unicast works flawlessly and multicast
> works partially (I think the problem is my switch actually) from VLC,
> so the STB definately knows how to handle both multicast and unicast.
> I use these URLs on the STB:
> (Streaming server IP: 10.0.0.20, STB IP: 10.0.0.25)
> unicast: udp://10.0.0.20:1234
> multicast: igmp://239.255.0.1:1234
> And of course set the destination IP and port in VLC.
> When I use your program i get a little garbage in the top part of my
> TV, but that is all.
>
> -Morgan T?rvolt-
>
>
>
> On 6/25/06, Ross Finlayson wrote:
> >
> > >I have been trying to send a ts file to a Amino STB for a while now.
> > >Using testMPEG2TransportStreamer with VLC as a reciever works great,
> > >but to Amino STB does not work.
> >
> > That's because "testMPEG2TransportStreamer" streams using multicast,
> > which the Amino STB does not handle.
> >
> > >I have tried unicast and multicast, both without
> > >much success.
> >
> > You can't (easily) modify the "testMPEG2TransportStreamer" to use
> > unicast. (Just changing the destination address won't be enough,
> > because the RTSP server still expects to send multicast.)
> >
> > To stream using unicast, use the "testOnDemandRTSPServer" demo
> > application. It works with the Amino STB - see
> > http://www.live555.com/liveMedia/faq.html#stb>
> >
> >
> > 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
> >
>
From finlayson at live555.com Sun Jun 25 11:18:12 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 25 Jun 2006 11:18:12 -0700
Subject: [Live-devel] testMPEG2TransportStreamer to Amino STB
In-Reply-To: <3cc3561f0606251022p688db9e6i1810fbb3b54b3f94@mail.gmail.co
m>
References: <3cc3561f0606250331y23db0012r812fa40ab9bd2b5e@mail.gmail.com>
<7.0.1.0.1.20060625081814.02030070@live555.com>
<3cc3561f0606251022p688db9e6i1810fbb3b54b3f94@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060625111017.0203b858@live555.com>
>Is there not a way to use Live555 for unicast UDP streaming? In the
>testMPEG2TransportStreamer.cpp source code I find this:
>
> // Note: This is a multicast address. If you wish to stream using
> // unicast instead, then replace this string with the unicast address
> // of the (single) destination. (You may also need to make a similar
> // change to the receiver program.)
This will work, but only because this demo application does not - by
default - have a built-in RTSP server. If you enable the built-in
RTSP server, then you won't be able to use it (i.e., using a
"rtsp://" URL) to receive the unicast stream.
However, as I've noted before, the "testOnDemandRTSPServer"
application *does* allow you to stream (via unicast) to an Amino STB
(provided that you enter the appropriate "rtsp://" URL). This works;
I suggest you use it.
(OTOH, I don't recommend trying to use the Amino STB without entering
a "rtsp://" URL. You might be able to get this to work, but it's not
the proper way to access a/v streams using a media player client.)
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From morgan.torvolt at gmail.com Mon Jun 26 00:49:20 2006
From: morgan.torvolt at gmail.com (=?ISO-8859-1?Q?Morgan_T=F8rvolt?=)
Date: Mon, 26 Jun 2006 11:49:20 +0400
Subject: [Live-devel] testMPEG2TransportStreamer to Amino STB
In-Reply-To: <7.0.1.0.1.20060625111017.0203b858@live555.com>
References: <3cc3561f0606250331y23db0012r812fa40ab9bd2b5e@mail.gmail.com>
<7.0.1.0.1.20060625081814.02030070@live555.com>
<3cc3561f0606251022p688db9e6i1810fbb3b54b3f94@mail.gmail.com>
<7.0.1.0.1.20060625111017.0203b858@live555.com>
Message-ID: <3cc3561f0606260049q67141efbxe02a10551da2f644@mail.gmail.com>
So what you are saying is that VLC and live555 does not do udp unicast
and multicast the same way, and even though it works with VLC, it will
not work with the current live555? Do you have any idea what changes I
would have to make to make it work like VLC in this?
I would prefer to have a solution that would let me stream
continuously from a transport stream device source rather than using
RTSP. If that is the only solution with live555, I will have to
rethink this problem.
-Morgan T?rvolt-
On 6/25/06, Ross Finlayson wrote:
>
> >Is there not a way to use Live555 for unicast UDP streaming? In the
> >testMPEG2TransportStreamer.cpp source code I find this:
> >
> > // Note: This is a multicast address. If you wish to stream using
> > // unicast instead, then replace this string with the unicast address
> > // of the (single) destination. (You may also need to make a similar
> > // change to the receiver program.)
>
> This will work, but only because this demo application does not - by
> default - have a built-in RTSP server. If you enable the built-in
> RTSP server, then you won't be able to use it (i.e., using a
> "rtsp://" URL) to receive the unicast stream.
>
> However, as I've noted before, the "testOnDemandRTSPServer"
> application *does* allow you to stream (via unicast) to an Amino STB
> (provided that you enter the appropriate "rtsp://" URL). This works;
> I suggest you use it.
>
> (OTOH, I don't recommend trying to use the Amino STB without entering
> a "rtsp://" URL. You might be able to get this to work, but it's not
> the proper way to access a/v streams using a media player client.)
>
>
> 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
>
From roy.subrato at gmail.com Mon Jun 26 05:02:42 2006
From: roy.subrato at gmail.com (subrato roy)
Date: Mon, 26 Jun 2006 15:02:42 +0300
Subject: [Live-devel] problem with outputting
Message-ID: <64849090606260502t63376637vf44590029560bc28@mail.gmail.com>
hello ,
I use the framedsource::getNextframe() to read and fill a large buffer
fTo in more than one reads. my question is how should i output this
buffer completly with the afterGetting() function.
can this function do the outputting irrespective of the size of the buffer ?
i tried a simple afterGetting() but on the just next getNextframe() it
complains of being read more than once at same time.
thanks
b r
subrato
From lorenooliveira at gmail.com Mon Jun 26 06:42:43 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Mon, 26 Jun 2006 10:42:43 -0300
Subject: [Live-devel] Problem transmitting audio/video using
testOnDemandRTSPServer and openRTSP
Message-ID:
Hi folks,
I cannot receive audio and video streamed by testOnDemandRTSPServer. The
problem is that the streams are stopped as soon as I try to receive them. In
what follows is the output of openRTSP in the client side.
Does anybony knows what can be wrong here?
Loreno
loreno at COMPOR05:~/testes$ ./openRTSP
rtsp://150.165.63.227:8554/mpeg1or2AudioVideoTest
Opened URL "rtsp://150.165.63.227:8554/mpeg1or2AudioVideoTest", returning a
SDP description:
v=0
o=- 1151327057410004 1 IN IP4 150.165.63.227
s=Session streamed by "testOnDemandRTSPServer"
i=mpeg1or2AudioVideoTest
t=0 0
a=tool:LIVE555 Streaming Media v2006.05.17
a=type:broadcast
a=control:*
a=range:npt=0-
a=x-qt-text-nam:Session streamed by "testOnDemandRTSPServer"
a=x-qt-text-inf:mpeg1or2AudioVideoTest
m=video 0 RTP/AVP 32
c=IN IP4 0.0.0.0
a=control:track1
m=audio 0 RTP/AVP 14
c=IN IP4 0.0.0.0
a=control:track2
Created receiver for "video/MPV" subsession (client ports 32812-32813)
Created receiver for "audio/MPA" subsession (client ports 32814-32815)
Setup "video/MPV" subsession (client ports 32812-32813)
Setup "audio/MPA" subsession (client ports 32814-32815)
Created output file: "video-MPV-1"
Created output file: "audio-MPA-2"
Started playing session
Receiving streamed data (signal with "kill -HUP 9017" or "kill -USR1 9017"
to terminate)...
Received RTCP "BYE" on "audio/MPA" subsession (after 1 seconds)
Received RTCP "BYE" on "video/MPV" subsession (after 1 seconds)
loreno at COMPOR05:~/testes$
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From finlayson at live555.com Mon Jun 26 07:09:14 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 26 Jun 2006 07:09:14 -0700
Subject: [Live-devel] testMPEG2TransportStreamer to Amino STB
In-Reply-To: <3cc3561f0606260049q67141efbxe02a10551da2f644@mail.gmail.co
m>
References: <3cc3561f0606250331y23db0012r812fa40ab9bd2b5e@mail.gmail.com>
<7.0.1.0.1.20060625081814.02030070@live555.com>
<3cc3561f0606251022p688db9e6i1810fbb3b54b3f94@mail.gmail.com>
<7.0.1.0.1.20060625111017.0203b858@live555.com>
<3cc3561f0606260049q67141efbxe02a10551da2f644@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060626065857.01fbedb0@live555.com>
At 12:49 AM 6/26/2006, you wrote:
>So what you are saying is that VLC and live555 does not do udp unicast
>and multicast the same way, and even though it works with VLC, it will
>not work with the current live555?
No, I am saying just that the *testMPEG2TransportStreamer*
application does not stream using UDP unicast, and cannot be modified
to do so without significant changes. If you really want to do this,
however, then
- look at
"MPEG2TransportFileServerMediaSubsession::createNewStreamSource()"
for hints on how to create the input source, and
- for the output sink, create a "BasicUDPSink" rather than a "RTPSink"
Alternatively, as I have explained before, you could just use the
"testOnDemandRTSPServer" application, as is. I don't know why you
don't want to use RTSP, because it is the best way to get what you
want. (Because of this (and because you have an unprofessional email
address), I won't be giving you any more assistance with your problem.)
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Mon Jun 26 07:14:41 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 26 Jun 2006 07:14:41 -0700
Subject: [Live-devel] Problem transmitting audio/video using
testOnDemandRTSPServer and openRTSP
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060626071141.01fbedb0@live555.com>
At 06:42 AM 6/26/2006, you wrote:
>Hi folks,
>
>I cannot receive audio and video streamed by testOnDemandRTSPServer.
>The problem is that the streams are stopped as soon as I try to
>receive them. In what follows is the output of openRTSP in the client side.
>
>Does anybony knows what can be wrong here?
It's hard to say for sure, but I suspect that your "test.mpg" file is
not a proper MPEG-1 or 2 Program Stream file.
If you are sure that your "test.mpg" is, indeed, a MPEG-1 or 2
Program Stream file, then please put it on a web server, and send us
the URL, so I can take a look at it.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From soaring_roc at yahoo.com.cn Mon Jun 26 08:04:48 2006
From: soaring_roc at yahoo.com.cn (=?gb2312?q?=D0=A1=B2=A8=20=CC=EF?=)
Date: Mon, 26 Jun 2006 23:04:48 +0800 (CST)
Subject: [Live-devel] how can I record the output of openRTSP?
Message-ID: <20060626150448.18602.qmail@web15706.mail.cnb.yahoo.com>
Hi, folks
just as we know that openRTSP can output the content of the media fomated of ".mov" or ".mp4" when we use the command option of "-q" or "-4", but it just can output it to the stdout, I want to record the media in a file(which then can be played bu the QuickTime player ), how can I do that? Any suggestion would be appreciated! Thanks!
soaring roc
__________________________________________________
???????????????
http://cn.mail.yahoo.com
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From lorenooliveira at gmail.com Mon Jun 26 10:23:39 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Mon, 26 Jun 2006 14:23:39 -0300
Subject: [Live-devel] how can I record the output of openRTSP?
In-Reply-To: <20060626150448.18602.qmail@web15706.mail.cnb.yahoo.com>
References: <20060626150448.18602.qmail@web15706.mail.cnb.yahoo.com>
Message-ID:
I believe the shortest path for doing that is to redirect the output to a
file with an appropriate extension. For instance:
$ ./openRTSP rtsp://... > some_name.mp3
Loreno
On 6/26/06, ?? ? wrote:
>
> Hi, folks
>
> just as we know that openRTSP can output the content of the media
> fomated of ".mov" or ".mp4" when we use the command option of "-q" or "-4",
> but it just can output it to the stdout, I want to record the media in a
> file(which then can be played bu the QuickTime player ), how can I do
> that? Any suggestion would be appreciated! Thanks!
>
> soaring roc
>
> __________________________________________________
> ???????????????
> http://cn.mail.yahoo.com
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
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From lorenooliveira at gmail.com Mon Jun 26 11:15:33 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Mon, 26 Jun 2006 15:15:33 -0300
Subject: [Live-devel] Problem transmitting audio/video using
testOnDemandRTSPServer and openRTSP
In-Reply-To: <7.0.1.0.1.20060626071141.01fbedb0@live555.com>
References:
<7.0.1.0.1.20060626071141.01fbedb0@live555.com>
Message-ID:
you are right. I tried some other mpg files with success.
However, when receiving the stream of some files I'm getting the following
messages:
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size
exceeds the client's buffer size (20000). 48 bytes of trailing data will be
dropped!
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size
exceeds the client's buffer size (20000). 48 bytes of trailing data will be
dropped!
The effect of this are some broken frames when I playback the saved video
file. Besides one of the files I've tried was streamed without any issue, I
would like to know how to avoid this loss of information. Does anybody has
some hint about this?
Loreno
On 6/26/06, Ross Finlayson wrote:
>
> At 06:42 AM 6/26/2006, you wrote:
> >Hi folks,
> >
> >I cannot receive audio and video streamed by testOnDemandRTSPServer.
> >The problem is that the streams are stopped as soon as I try to
> >receive them. In what follows is the output of openRTSP in the client
> side.
> >
> >Does anybony knows what can be wrong here?
>
> It's hard to say for sure, but I suspect that your "test.mpg" file is
> not a proper MPEG-1 or 2 Program Stream file.
>
> If you are sure that your "test.mpg" is, indeed, a MPEG-1 or 2
> Program Stream file, then please put it on a web server, and send us
> the URL, so I can take a look at it.
>
>
> 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
>
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From finlayson at live555.com Mon Jun 26 15:46:18 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 26 Jun 2006 15:46:18 -0700
Subject: [Live-devel] Problem transmitting audio/video using
testOnDemandRTSPServer and openRTSP
In-Reply-To:
References:
<7.0.1.0.1.20060626071141.01fbedb0@live555.com>
Message-ID: <7.0.1.0.1.20060626154508.02034058@live555.com>
>MultiFramedRTPSource::doGetNextFrame1(): The total received frame
>size exceeds the client's buffer size (20000). 48 bytes of trailing
>data will be dropped!
>
>The effect of this are some broken frames when I playback the saved
>video file. Besides one of the files I've tried was streamed without
>any issue, I would like to know how to avoid this loss of
>information. Does anybody has some hint about this?
Yes, please read the "openRTSP" documentation:
. You'll find an answer there.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From roy.subrato at gmail.com Tue Jun 27 05:37:23 2006
From: roy.subrato at gmail.com (subrato roy)
Date: Tue, 27 Jun 2006 15:37:23 +0300
Subject: [Live-devel] sending large buffer to sink
Message-ID: <64849090606270537h28594a5oa9add1698828c196@mail.gmail.com>
Hello again ,
I am working with mpeg2 ts. with the getNextFrame() in a single call ,
i get 7 ts packets in the buffer. if i call a afterGetting() on this
buffer , it will work fine.But for my work , i have to call it many
times and fill up a buffer with 2800 TS packets. Now i want to ouput
this buffer with afterGetting() . can someone tell me how to do this ?
thanks for all the help
subrato
From tire.tw at gmail.com Tue Jun 27 07:41:05 2006
From: tire.tw at gmail.com (Tire)
Date: Tue, 27 Jun 2006 22:41:05 +0800
Subject: [Live-devel] RTCP Question
Message-ID: <7992557a0606270741p3c52c7dbj4437fb34c6f6bf73@mail.gmail.com>
Dear All :
Is the RTCP in the OnDemandRTSPServer use the UDP to transmit and receive
data ?
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From tire.tw at gmail.com Tue Jun 27 08:01:54 2006
From: tire.tw at gmail.com (Tire)
Date: Tue, 27 Jun 2006 23:01:54 +0800
Subject: [Live-devel] UDP port question
Message-ID: <7992557a0606270801m650110d1yf1d279dbdf47f055@mail.gmail.com>
In the "testOnDemandRTSPServer" demo application, the TCP is used to initial
handshake and UDP is transmit video/audio data.
At the same time, I find the TCP port is fixed and the UDP port will have
the dissimilarity according to the client.
Can the UDP port fix same ?
How to do ?
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From lorenooliveira at gmail.com Tue Jun 27 08:23:08 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Tue, 27 Jun 2006 07:23:08 -0800
Subject: [Live-devel] RTCP Question
In-Reply-To: <7992557a0606270741p3c52c7dbj4437fb34c6f6bf73@mail.gmail.com>
References: <7992557a0606270741p3c52c7dbj4437fb34c6f6bf73@mail.gmail.com>
Message-ID:
No. RTCP and RTSP data is transported in TCP packets.Conversely, RTP data is
transmitted through UDP datagrams.
Loreno
On 6/27/06, Tire wrote:
>
> Dear All :
>
> Is the RTCP in the OnDemandRTSPServer use the UDP to transmit and receive
> data ?
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
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From tire.tw at gmail.com Tue Jun 27 08:49:09 2006
From: tire.tw at gmail.com (Tire)
Date: Tue, 27 Jun 2006 23:49:09 +0800
Subject: [Live-devel] RTCP Question
In-Reply-To:
References: <7992557a0606270741p3c52c7dbj4437fb34c6f6bf73@mail.gmail.com>
Message-ID: <7992557a0606270849s68c27481tbc909f73960a23eb@mail.gmail.com>
Therefore, the RTCP is use the TCP in the "testOnDemandRTSPServer" demo
application.
Can I assign a fixed TCP port to this RTCP ?
How to do ?
2006/6/27, Loreno Oliveira :
>
> No. RTCP and RTSP data is transported in TCP packets.Conversely, RTP data
> is transmitted through UDP datagrams.
>
> Loreno
>
> On 6/27/06, Tire < tire.tw at gmail.com> wrote:
>
> > Dear All :
> >
> > Is the RTCP in the OnDemandRTSPServer use the UDP to transmit and
> > receive data ?
> >
> > _______________________________________________
> > 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
>
>
>
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From lorenooliveira at gmail.com Tue Jun 27 10:41:38 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Tue, 27 Jun 2006 14:41:38 -0300
Subject: [Live-devel] verifying streamed files
Message-ID:
Is there any way of veifying whether a file received through streaming is
identical or not to the original file?
Well, I'm using testOnDemandRTSPServer and openRSTP for streaming/receiving
an MP3 file. The idea is to check the received file and detect if it were
packet losses during the streaming.
My first approach was verifying the cheksums of the files (original and
received). Oddly the received file is lower than the original file in some
bytes, but without neither any perceived damage on the file when reproducing
it nor error messages in the console during the stream reception.
Some idea??
loreno
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From finlayson at live555.com Tue Jun 27 11:17:31 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 27 Jun 2006 11:17:31 -0700
Subject: [Live-devel] RTCP Question
In-Reply-To:
References: <7992557a0606270741p3c52c7dbj4437fb34c6f6bf73@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060627111529.01fb5aa0@live555.com>
At 08:23 AM 6/27/2006, you wrote:
>No. RTCP and RTSP data is transported in TCP packets.
That's not (usually) correct. RTCP packets are, by default, sent
(and received) via UDP, as are RTP packets.
(RTP and RTCP packets use TCP only if the server is specifically
requested to use RTP-over-TCP streaming.)
>Therefore, the RTCP is use the TCP in the "testOnDemandRTSPServer"
>demo application.
No, not usually.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From lorenooliveira at gmail.com Tue Jun 27 11:25:15 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Tue, 27 Jun 2006 15:25:15 -0300
Subject: [Live-devel] RTCP Question
In-Reply-To: <7.0.1.0.1.20060627111529.01fb5aa0@live555.com>
References: <7992557a0606270741p3c52c7dbj4437fb34c6f6bf73@mail.gmail.com>
<7.0.1.0.1.20060627111529.01fb5aa0@live555.com>
Message-ID:
Sorry, my fault :-)
On 6/27/06, Ross Finlayson wrote:
>
> At 08:23 AM 6/27/2006, you wrote:
> >No. RTCP and RTSP data is transported in TCP packets.
>
> That's not (usually) correct. RTCP packets are, by default, sent
> (and received) via UDP, as are RTP packets.
>
> (RTP and RTCP packets use TCP only if the server is specifically
> requested to use RTP-over-TCP streaming.)
>
> >Therefore, the RTCP is use the TCP in the "testOnDemandRTSPServer"
> >demo application.
>
> No, not usually.
>
>
> 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
>
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From roy.subrato at gmail.com Wed Jun 28 23:57:02 2006
From: roy.subrato at gmail.com (subrato roy)
Date: Thu, 29 Jun 2006 09:57:02 +0300
Subject: [Live-devel] buffer question unanswered
Message-ID: <64849090606282357g29d64f89pbe749f715c6a6cc6@mail.gmail.com>
hello dear all ,
I guess i am doing something wrong if i dont receive any answers here.
My question was is it possible to do more than 1 reads on a mpeg2 TS
stream (i am calling getNextFrame() multiple times to get 2800 TS
packets in a buffer) and then output the resulting buffer using
afterGetting().
For 2 reads , i attempted to call afterGetting() 2 times before any
new call to getNextFrame(). I get a black screen with this in
vlc(output stream).
thank you all
b r
subrato
From zhangzx at rcs-9000.com Thu Jun 29 01:30:09 2006
From: zhangzx at rcs-9000.com (Zhixue Zhang)
Date: Thu, 29 Jun 2006 16:30:09 +0800
Subject: [Live-devel] "PAUSE" with new version
Message-ID: <000601c69b56$3e697830$347457c6@rcs9000.com>
Dear Ross, I have met a strange problem.
When my application is built with live.2006.01.05, it works fairly well
when I issued "PAUSE" and "RESUME" command. But when I update live555 to
live.2006.05.17 to support RTP over TCP, problems happened. When I
issued "PAUSE", the program is frozen at singlestep().
I have tried my best to test it, but failed.:(
Can you tell me possible reason and possible solution?
Thanks a lot.
Zhixue Zhang
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From finlayson at live555.com Thu Jun 29 01:56:04 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 29 Jun 2006 01:56:04 -0700
Subject: [Live-devel] "PAUSE" with new version
In-Reply-To: <000601c69b56$3e697830$347457c6@rcs9000.com>
References: <000601c69b56$3e697830$347457c6@rcs9000.com>
Message-ID: <7.0.1.0.1.20060629015329.01fb68b0@live555.com>
>When my application is built with live.2006.01.05, it works fairly
>well when I issued "PAUSE" and "RESUME" command. But when I update
>live555 to live.2006.05.17 to support RTP over TCP, problems
>happened. When I issued "PAUSE", the program is frozen at singlestep().
>
>I have tried my best to test it, but failed.:(
>Can you tell me possible reason and possible solution?
Not really. Note that "RTSPClient::pauseMediaSession()" deliberately
does not look for a response to the "PAUSE" when were're streaming
RTP-over-TCP (because in that case, the TCP connection is being read
instead by a handler that's extracting embedded RTP/RTCP
packets). That should have prevented any problems with "PAUSE" and
RTP-over-TCP, so I don't know why you might still be experiencing
problems with this.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From mohamed.lalmi at netcentrex.net Thu Jun 29 02:08:09 2006
From: mohamed.lalmi at netcentrex.net (M.LALMI)
Date: Thu, 29 Jun 2006 11:08:09 +0200
Subject: [Live-devel] "PAUSE" with new version
Message-ID: <200606290908.k5T987301598@mailer.paris.netcentrex.net>
Hello,
I am reading an H263+ stream, some frames arrived in many packets with the same TimeStamp,
Each packet is sent directly to ffmpeg.
How can I do to make Live wait for a complete frame before to send it to ffmpeg ?
From mohamed.lalmi at netcentrex.net Thu Jun 29 05:41:03 2006
From: mohamed.lalmi at netcentrex.net (M.LALMI)
Date: Thu, 29 Jun 2006 14:41:03 +0200
Subject: [Live-devel] traces
Message-ID: <200606291241.k5TCf1316266@mailer.paris.netcentrex.net>
is it possible to activate traces of livemedia ?
From dsmurl at yahoo.com Thu Jun 29 10:05:10 2006
From: dsmurl at yahoo.com (Sam b)
Date: Thu, 29 Jun 2006 10:05:10 -0700 (PDT)
Subject: [Live-devel] Prospective newbie question.
Message-ID: <20060629170510.3387.qmail@web38404.mail.mud.yahoo.com>
Hello,
I need to write an encoder head that can:
1. receive an elementry stream from and encoder I
have writtin
2. Packetize it rtsp/rtp
3. then send it out to a port to be recieved later
by vlc or helix
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
From dsmurl at yahoo.com Thu Jun 29 10:07:56 2006
From: dsmurl at yahoo.com (Sam b)
Date: Thu, 29 Jun 2006 10:07:56 -0700 (PDT)
Subject: [Live-devel] Prospective newbie question [Full]
Message-ID: <20060629170756.16784.qmail@web38403.mail.mud.yahoo.com>
Hello,
I need to write an encoder head that can:
1. receive an elementary stream from and encoder I
have written
2. Packetize it rtsp/rtp
3. then send it out to a port to be received later
by vlc or helix
Is this the right project for this purpose and where
might I start? Any help would be appreciated.
- Thanks,
Sam Bagot
ps. sorry about the last incomplete post.
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
From finlayson at live555.com Thu Jun 29 10:39:28 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 29 Jun 2006 10:39:28 -0700
Subject: [Live-devel] Prospective newbie question [Full]
In-Reply-To: <20060629170756.16784.qmail@web38403.mail.mud.yahoo.com>
References: <20060629170756.16784.qmail@web38403.mail.mud.yahoo.com>
Message-ID: <7.0.1.0.1.20060629103613.01ff0418@live555.com>
> I need to write an encoder head that can:
>
> 1. receive an elementary stream from and encoder I
>have written
>
> 2. Packetize it rtsp/rtp
>
> 3. then send it out to a port to be received later
>by vlc or helix
>
> Is this the right project for this purpose
Yes, perhaps - although I'm not really sure what you mean by your
step 3. Do you plan to use VLC or Helix as a server (i.e., a
transmitter)? If so, then you really don't need to do this, because
we have our own RTSP/RTP server implementation.
> and where might I start?
Read the FAQ, and play around with the demo applications (especially
"testOnDemandRTSPServer").
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From zhangzx at rcs-9000.com Thu Jun 29 17:05:46 2006
From: zhangzx at rcs-9000.com (Zhixue Zhang)
Date: Fri, 30 Jun 2006 08:05:46 +0800
Subject: [Live-devel] "PAUSE" with new version
In-Reply-To: <7.0.1.0.1.20060629015329.01fb68b0@live555.com>
Message-ID: <000501c69bd8$f2b77b70$347457c6@rcs9000.com>
Dear Ross
I didn't explain my problem clearly. I want my application to support
both UDP and TCP.
When streaming with unicast, "PAUSE" works well with live.2006.01.05,
and frozen in while(1) {singlestep()} with new version.
thanks for your reply.
Zhixue Zhang
Nari-relays Electric Corporation, Ltd.
Tel:025-52100626
From mohamed.lalmi at netcentrex.net Fri Jun 30 04:25:23 2006
From: mohamed.lalmi at netcentrex.net (M.LALMI)
Date: Fri, 30 Jun 2006 13:25:23 +0200
Subject: [Live-devel] processSpecialHeader
Message-ID: <200606301125.k5UBPM322632@mailer.paris.netcentrex.net>
what does this function ?
processSpecialHeader ( processSpecialHeader(BufferedPacket* packet, unsigned& resultSpecialHeaderSize)
what is the role of : resultSpecialHeaderSize ??
thank you in advance for your help
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From finlayson at live555.com Fri Jun 30 07:53:11 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 30 Jun 2006 07:53:11 -0700
Subject: [Live-devel] processSpecialHeader
In-Reply-To: <200606301125.k5UBPM322632@mailer.paris.netcentrex.net>
References: <200606301125.k5UBPM322632@mailer.paris.netcentrex.net>
Message-ID: <7.0.1.0.1.20060630075209.01fd9e18@live555.com>
At 04:25 AM 6/30/2006, you wrote:
>what does this function ?
>
>processSpecialHeader ( processSpecialHeader(BufferedPacket* packet,
>unsigned& resultSpecialHeaderSize)
>
>what is the role of : resultSpecialHeaderSize ??
This should be clear from its use in the code:
liveMedia/include/MultiFramedRTPSource.hh
liveMedia/MultiFramedRTPSource.cpp
and the various subclasses of "MultiFramedRTPSource" that redefine
this virtual function.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From kgurganus at 650dialup.com Fri Jun 30 08:54:12 2006
From: kgurganus at 650dialup.com (Keith Gurganus)
Date: Fri, 30 Jun 2006 08:54:12 -0700
Subject: [Live-devel] RTSP "Manual Unicast
In-Reply-To: <200606291241.k5TCf1316266@mailer.paris.netcentrex.net>
Message-ID:
Hi,
Is it possible to do a RTSP "Manual Unicast" as opposed to an "Automatic
Unicast (Announce) with live.com?
- Keith
From finlayson at live555.com Fri Jun 30 09:13:11 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 30 Jun 2006 09:13:11 -0700
Subject: [Live-devel] RTSP "Manual Unicast
In-Reply-To:
References: <200606291241.k5TCf1316266@mailer.paris.netcentrex.net>
Message-ID: <7.0.1.0.1.20060630091218.01b9a680@live555.com>
>Is it possible to do a RTSP "Manual Unicast" as opposed to an "Automatic
>Unicast (Announce) with live.com?
I don't understand this question at all. Please elaborate.
Also, we're not "live.com" anymore.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From braymond at echostorm.net Fri Jun 30 11:48:14 2006
From: braymond at echostorm.net (Brian Raymond)
Date: Fri, 30 Jun 2006 14:48:14 -0400
Subject: [Live-devel] Compile issues on OSX 10.4.6 4.0.1
In-Reply-To: <7.0.1.0.1.20060619214852.01f5bc70@live555.com>
References:
<7.0.1.0.1.20060619214852.01f5bc70@live555.com>
Message-ID: <9B2BB506-8606-4D09-83EF-4C387AEF92C0@echostorm.net>
Sorry about the delay in replying, this email slippsed down my inbox.
Yes, that was the reason I was confused, I ran it a couple of times
just to make sure.
From the Makefile:
COMPILE_OPTS = $(INCLUDES) -I. -DBSD=1 -O -DSOCKLEN_T=socklen_t
- Brian
On Jun 20, 2006, at 12:51 AM, Ross Finlayson wrote:
>
>> Casting the parameter to "(socklen_t*)" in all of the code using
>> SOCKLEN_T solves the problem
>
> Did you create your "Makefile"s using
> genMakefiles macosx
> ?
>
> If so, then you should get
> -DSOCKLEN_T=socklen_t
> on each compile line.
>
>
> 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
>
>
From braymond at echostorm.net Fri Jun 30 11:49:42 2006
From: braymond at echostorm.net (Brian Raymond)
Date: Fri, 30 Jun 2006 14:49:42 -0400
Subject: [Live-devel] RTSP "Manual Unicast
In-Reply-To:
References:
Message-ID: <62EC3CA6-890D-461C-9C64-E27838D8FE29@echostorm.net>
I take it you are referring to Darwin Streaming Server since this is
the verbage QT uses when referring to how to stream. Unless you use
the DarwinInjector class to announce a stream they are actually all
"Manual Unicast".
- Brian
On Jun 30, 2006, at 11:54 AM, Keith Gurganus wrote:
> Hi,
>
> Is it possible to do a RTSP "Manual Unicast" as opposed to an
> "Automatic
> Unicast (Announce) with live.com?
>
> - Keith
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>