yueyanbin@vimicro.com | 29 Jul 06:17 2014

How to mke a URL with username & password?

Hello everyone:

If I use the Authentication in the rtspsverver (as define ACCESS_CONTROL in testOnDemandRTSPServer.cpp), how could I make the url with the user name and password?
We use the testOnDemandRTSPServer as rtsp server,  and vlc as the client, use the url rtsp://IP:port/test.h264 to play, then vlc show a username & password input dialog.
Even we use the url  rtsp://IP:port/test.h264&user=username&password=password to play, the input dialog show all the same.

Could anyone tell me how to make the url with username & password?

Thanks

yueyanbin <at> vimicro.com
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Cường Lê | 27 Jul 12:57 2014
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RTSP Server lost packet when stream multi-file.

Hi!
I'm use your class rtsp server.
if I creat below, it is operator very good with multi-client connected to ServerMediaSession. No packet loss.
      ServerMediaSession* sms =    ServerMediaSession::createNew(*env,           streamName,  streamName,  descriptionString);
    sms->addSubsession(H264VideoFileServerMediaSubsession
               ::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);


but if I creat below, each ServerMediaSession have 1 client connected, the packet loss is large.

ServerMediaSession* sms =    ServerMediaSession::createNew(*env,           streamName,  streamName,  descriptionString);
    sms->addSubsession(H264VideoFileServerMediaSubsession
               ::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);

ServerMediaSession* sms2 =    ServerMediaSession::createNew(*env,           streamName2,  streamName2,  descriptionString);
    sms2->addSubsession(H264VideoFileServerMediaSubsession
               ::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms2);



can you help me explain and fix this problem?
Thank you






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ChaSeop Im | 24 Jul 14:31 2014
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Problem with RTPInterface::sendPacket method

Hi.

I have implemented a RTSP server with own recording files.

The problem occurs when removed stream socket while sending packet over tcp sockets.

When packet sending is failed in RTPInterface::sendDataOverTCP method, it called RTPInterface::removeStreamSocket method.

call stack like this : 
RtspPlaybackServerTest.exe!RTPInterface::removeStreamSocket(int sockNum=444, unsigned char streamChannelId='') 
RtspPlaybackServerTest.exe!RTPInterface::sendDataOverTCP(int socketNum=444, const unsigned char * data=0x000000528417e860, unsigned int dataSize=4, bool forceSendToSucceed=false) 
RtspPlaybackServerTest.exe!RTPInterface::sendRTPorRTCPPacketOverTCP(unsigned char * packet=0x000000528b219070, unsigned int packetSize=1448, int socketNum=444, unsigned char streamChannelId='\0') 
RtspPlaybackServerTest.exe!RTPInterface::sendPacket(unsigned char * packet=0x000000528b219070, unsigned int packetSize=1448) 


RTPInterface::removeStreamSocket delete streamPtr. This occur crash in RTPInterface::sendPacket. because values of streams and streams->fNext are not valid.

When I added break; after success = False; statement, crash is not occured. but, I don't know this is right. 

how to fix it?


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Xuan | 24 Jul 03:03 2014
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How to combine live H.264 source and AAC source

 

 

发件人: Xuan [mailto:lkxkfl <at> hotmail.com]
发送时间: 2014723 23:01
收件人: live-devel <at> lists.live555.com
主题: How to combine live H.264 source and AAC source

 

Hello,

  I am working on a testing program based on live555. In my program, I use a thread to capture video frames by webcam, and another thread to capture audio frames. In each thread, both video and audio frames are encoded with ffmpeg. Then I use a third thread to stream one of them.

  My problem is that streaming either of them is Ok, but how can I stream them together? If I do like this in a thread:

sms->addSubsession(PassiveServerMediaSubsession::createNew(*vSink, rtcp));

sms->addSubsession(PassiveServerMediaSubsession::createNew(*aSink, audioRtcp));

rtspServer->addServerMediaSession(sms);

H264RealTimeStreamSource* naluSource =  H264RealTimeStreamSource::createNew(*env_live,&pThis->videoCList2,frame_rate);

h264Source = H264VideoStreamDiscreteFramer::createNew(*env_live, naluSource);

adtsSource = ADTSRealTimeStreamSource::createNew(*env_live,&pThis->audioCList,1,44100,2,NULL);

pThis->startVideoLive = vSink->startPlaying(*h264Source, afterPlayingLiveH264, NULL);

pThis->startAudioLive = aSink->startPlaying(*adtsSource, afterPlayingLiveAAC, NULL);

I can only receive video frames with VLC player, and also timestamp of VLC player is rather unstable.

Could I do anything wrong, or maybe I should stream h.264 and aac in different thread rather than in one thread at the same time?

 

Thank you a lot. Looking forward for reply!

 

Xuan

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Jon Shemitz | 24 Jul 20:39 2014

Decode Secure RTP on Android, using MediaCodec + MediaCrypto?

I hope this isn’t spam: I’m using VLC to stream camera output over RTSP and, so far as I know, VLC relies on live555 code for that.

 

Anyhow (as per http://stackoverflow.com/questions/24920200/which-mediacrypto-uuid-do-you-use-with-secure-rtp) I’ve written  some Android code that does a nice job decoding H264 format RTSP video, using the MediaCodec. But when I have VLC use Secure RTP, I need to pass the MediaCodec a MediaCrypto instance … which requires a UUID and a byte array of “initialization data”.

 

Can anyone here point me in the right direction?

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Marco Porsch | 24 Jul 18:34 2014
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RFC4175 uncompressed video streaming: variable payload header length

Hello,

I am currently trying to implement uncompressed video streaming according to RFC4175. An issue that I
struggle with, is that the payload header is variable in length (see [1]). The header length depends on
packet size, image line length and current fragmentation offset. (I use some simplifications
concerning color sampling for now. Also the payload header is not yet evaluated except for the
continuation flag.)

I can calculate the header length and write the header accordingly in my subclass'
doSpecialFrameHandling() function that calls setSpecialHeaderBytes(). But on the receiver side I
always see image line offset artefacts, i.e. one or multiple lines shifted left/right.
The receiver code should be alright, as it just looks for the "continuation" flag to determine the header
length and skips the header by setting "resultSpecialHeaderSize" accordingly.

The issue seems to be in the order of events concerning doSpecialFrameHandling() and
specialHeaderSize(). My subclass' specialHeaderSize() is called called from
MultiFramedRTPSink.cpp before the header is written in my doSpecialFrameHandling(). Thus, it seems I
have to predict the header size one fragment ahead? But in that case I do not yet know how large
"numBytesInFrame" is, which seems to change depending on my previously written header sizes...

I also tried using setFrameSpecificHeaderBytes() and frameSpecificHeaderSize(), but the result was
just a totally garbled and twisted image on the receiver side.

Yes, this is all a bit confusing. Maybe I am just doing something totally wrong. Maybe you could clarify on
how to cope with variable per-frame payload header lengths at all?

Cheers,
Marco Porsch

[1]
http://tools.ietf.org/html/rfc4175

PS: Please let's not have a discussion on the craziness of uncompressed video streaming in high
resolution. I am aware that it is. =)
Michael Rahlff | 23 Jul 17:17 2014
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Problem setting rtpmap on Flir A320 thermal camera.

Dear Sirs.
I am having trouble trying to get a thermal camera (Flir A320) to stream the raw data.

My goal is to receive raw data from a Flir A320 (rtpmap 102 [See below output] ).

I have successfully connected to and streamed MPEG4 encoded video from the camera with VLC, but as mentioned, I need to change to rtpmap 102 to get the raw data instead.

I have downloaded “live555” and looked at testRTSPClient.cpp example, but cannot see where to set rtpmap.

Is it correct understanding, I need to make my own class and then inherence from the RTPSource class?

According to the Flir A320 manual, the transport format is as described in RFC4175 (RTP - payload format for uncompressed video)

Flir A320 SDP DESCRIBE response:
v=0
o=- 0 0 IN IP4 169.0.0.2
s=IR stream
i=Live infrared
t=now-
c=IN IP4 169.0.0.2
m=video 0 RTP/AVP  96 97 98 99 100 102 103
a=control:rtsp://169.0.0.2/sid=96
a=framerate:30
a=rtpmap:96 MP4V-ES/90000
a=framesize:96 640-480
a=fmtp:96 profile-level-id=1;config=000001B003000001B509000001010000012002045D4C28A021E0A4C7
a=rtpmap:97 MP4V-ES/90000
a=framesize:97 320-240
a=fmtp:97 profile-level-id=1;config=000001B003000001B509000001010000012002045D4C285020F0A4C7
a=rtpmap:98 MP4V-ES/90000
a=framesize:98 160-128
a=fmtp:98 profile-level-id=1;config=000001B003000001B509000001010000012002045D4C282820A0A4C7
a=rtpmap:99 FCAM/90000
a=framesize:99 320-240
a=fmtp:99 sampling=mono; width=320; height=240; depth=16
a=rtpmap:100 FCAM/90000
a=framesize:100 160-120
a=fmtp:100 sampling=mono; width=160; height=120; depth=16
a=rtpmap:102 raw/90000
a=framesize:102 320-240
a=fmtp:102 sampling=mono; width=320; height=240; depth=16
a=rtpmap:103 raw/90000
a=framesize:103 160-120
a=fmtp:103 sampling=mono; width=160; height=120; depth=16


Thanks in advance

Best regards

Michael Rahlff
Holsteinsgade 35B
8300 Odder

Tlf: 407-408-22


www.m-rahlff.dk






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RTSP range clock/npt SDP parse

Hi,
looking into file liveMedia/MediaSession.cpp I believe there is an
extra space in parseRangeAttribute method.

Instead of:
return sscanf(sdpLine, "a=range: npt = %lg - %lg", &startTime, &endTime) == 2;
I think it should read:
return sscanf(sdpLine, "a=range:npt=%lg - %lg", &startTime, &endTime) == 2;

Instead of:
int sscanfResult = sscanf(sdpLine, "a=range:clock=%[^-\r\n]-%[^\r\n]", as, ae);
I think it should read:
int sscanfResult = sscanf(sdpLine, "a=range:clock=%[^-\r\n]-%[^\r\n]", as, ae);

Could you confirm if this is indeed the case ? Or can you at least
support both cases ?

Thank you,
Paulo Vitor
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Cường Lê | 18 Jul 08:50 2014
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RTSP Server: streaming filr .mp4

Hi!
I 'm using your live555. I very like it.
I have builded and streamed file .mkv, .h264, .m4v..., it can operate very good.
When I try to streamed file .mp4, i have used VLC client capture URL but it can't display and error.
Can you help me to stream file .mp4 in RTSP server?
Thank you very much!
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Alix Frombach | 17 Jul 13:26 2014
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Using openRTP for receiving MPEG TS

Hello,

I am able to use the openRTSP client for receiving H.264 and MP4 data from an IP camera fine, but cannot seem to find a way to receive MPEG TS from the same camera.  Is this a supported feature of openRTSP?  I saw a method of piping the received H.264 video data to MPEG TS using testH264VideoToTransportStream, but this is not desired as I would like to capture video and audio simultaneously.

Thanks in advance
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rajesh gupta | 16 Jul 11:21 2014
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Streaming from live source

Dear All,
             Is that possible from Live555MediaServer can stream the data form my webcamera .
 
Regards
Rajesh
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