Re: [Live-devel] Streaming to RTSP clients with different ports

> So in rephrased question, which live555 server module can give me this > functionality, i.e.: > > [FFmpeg (RTP (AAC) audio)] --> [live555 server module] --> [FFmpeg (RTSP > (AAC) audio)] If the first RTP AAC audio source is accessible via RTSP (i.e., if it has a RTSP server), then you should b

Re: [Live-devel] Streaming to RTSP clients with different ports

Message-ID: <4d4ba832-f4d3-4634-9a6e-c36b0e040...@live555.com> Content-Type: text/plain; charset="utf-8" > I have a test scenario, where I am streaming webcam input + mic over to a remote server using RTP protocol, which re-streams this to another client (other client pulls data through RTSP): >

Re: [Live-devel] Adding streams to already-running event loop

> OK, so the “addNewStream” triggered event, where the RTSPClient is created, > would add the pointer to it then. I’m a bit concerned about thread-safety > here, since the struct is created on the GUI thread, then manipulated on the > live555 thread. I suppose you’re implying that safety would c

Re: [Live-devel] Adding streams to already-running event loop

> On Nov 12, 2014, at 12:31 PM, Ross Finlayson wrote: > >> I’ve got this working along the lines you’ve described, but the problem I’m >> running into is a need for some kind of external (to live555 thread) >> reference to each RTSPClient, in order to trigger other events (stopping >> one, fo

Re: [Live-devel] Adding streams to already-running event loop

> I’ve got this working along the lines you’ve described, but the problem I’m > running into is a need for some kind of external (to live555 thread) > reference to each RTSPClient, in order to trigger other events (stopping one, > for example). That’s why I thought about creating the clients out

Re: [Live-devel] Adding streams to already-running event loop

Hi Ross, Thanks for your help so far. I’ve got this working along the lines you’ve described, but the problem I’m running into is a need for some kind of external (to live555 thread) reference to each RTSPClient, in order to trigger other events (stopping one, for example). That’s why I though

Re: [Live-devel] Possible authentication bug opening several RTSPClient from the same event loop

> To be honest, i sometimes observe this kind of situation in VLC that even > with good User:Pass combination - VLC was giving me 401 auth problem. But i > never fired the bug here as you manytimes statded that VLC is not your > software. That’s correct. However, if you see such an error with

Re: [Live-devel] Possible authentication bug opening several RTSPClient from the same event loop

Yes - my fault! I’ve just installed a new version (2014.11.12) that fixes this. Thanks for the report. Ross Finlayson Live Networks, Inc. http://www.live555.com/ ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailma

Re: [Live-devel] Possible authentication bug opening several RTSPClient from the same event loop

Hello Ross, To be honest, i sometimes observe this kind of situation in VLC that even with good User:Pass combination - VLC was giving me 401 auth problem. But i never fired the bug here as you manytimes statded that VLC is not your software. Looks like it is not realted to VLC directly. Marci

Re: [Live-devel] Possible authentication bug opening several RTSPClient from the same event loop

Hi Ross, Looking at the code fAllowBasicAuthentication is never initialized. 2014-11-12 19:01 GMT+01:00 Francisco Feijoo : > Hi Ross, > > Thanks for your support. > > Looks like I can reproduce the problem with the unmodified openRTSP > command. It works on VLC. > > ./openRTSP rtsp://admin:12345

Re: [Live-devel] Possible authentication bug opening several RTSPClient from the same event loop

Hi Ross, Thanks for your support. Looks like I can reproduce the problem with the unmodified openRTSP command. It works on VLC. ./openRTSP rtsp://admin:12345@192.168.1.222:554/h.264/ch7/main/av_stream Opening connection to 192.168.1.222, port 554... ...remote connection opened Sending request: O

Re: [Live-devel] Opus test file

> Would it be possible to provide the opus test file you use for testing opus > RTP streaming on live555.com or in the tar file? I’ve left an example “.opus” file in http://www.live555.com/liveMedia/public/opus/ This can be streamed by our

Re: [Live-devel] current position, time problem

> We have "Amino A139 STB" , and live555 server running on "Ubuntu 14.04.1 LTS > (GNU/Linux 3.13.0-32-generic x86_64)”. Linux ‘distributions’ (such as ‘Ubuntu’) usually contain old, out-of-date versions of the LIVE555 Streaming Media software. Please make sure that you are running an up-to-dat

[Live-devel] Opus test file

Hi Ross, Would it be possible to provide the opus test file you use for testing opus RTP streaming on live555.com or in the tar file? Thanks, Ralf -- This message is subject to the CSIR's copyright terms and conditions, e-mail legal notice, and implemented Open Document Format (ODF) stan

Re: [Live-devel] Streaming to RTSP clients with different ports

> I have a test scenario, where I am streaming webcam input + mic over to a > remote server using RTP protocol, which re-streams this to another client > (other client pulls data through RTSP): > > FFmpeg (RTP AAC audio + RTP video) -> Server -> (RTSP AAC audio + RTSP video) > FFmpeg client Y

Re: [Live-devel] Possible authentication bug opening several RTSPClient from the same event loop

Can you reproduce the problem using the *unmodified* “openRTSP” command - or does the problem occur only with your custom RTSP client application (that is opening multiple RTSP streams)? If it’s the latter, then I suspect that your problem may be that you are using the same “Authenticator” obje

[Live-devel] current position, time problem

Thanks in advanced , for this great job you have done. We have "Amino A139 STB" , and live555 server running on "Ubuntu 14.04.1 LTS (GNU/Linux 3.13.0-32-generic x86_64)". When playing TS streams , or Audio TS files, the "current position" or "current time" is "-1" which means Amino cannot receiv

[Live-devel] Streaming to RTSP clients with different ports

Hi, I have a test scenario, where I am streaming webcam input + mic over to a remote server using RTP protocol, which re-streams this to another client (other client pulls data through RTSP): FFmpeg (RTP AAC audio + RTP video) -> Server -> (RTSP AAC audio + RTSP video) FFmpeg client In

[Live-devel] Possible authentication bug opening several RTSPClient from the same event loop

Hello, After updating live555 to the latest live.2014.11.07 I'm seeing some issues where some connections are failing with: "Failed to get a SDP description: 401 Unauthorized " I open several RTSP streams from the same event loop using one RTSPClient for each stream. Some of those streams conne