I'm using ffmpeg to stream to YouTube live. I have everything working perfectly.
When I enable DVR on my channel I immediately start receiving playback errors on all iOS devices. When I disable DVR I immediately fix the DVR issue.
I've tried changing the GOV and my KeyFrames. YouTube live's control panel does not indicated any errors.
Anyone else seen this?
Related
YouTube streaming get automatically disable itself. We are doing live streaming through VMIX software. It works fine but sometimes it automatically stop the streaming on the VMIX software we checked streaming looks good.
We stopped and restart the streaming on VMIX but it didn't make any difference until you go to open YouTube live streaming page and then stop and start the streaming from VMIX.
Can anyone face this type of issue? Please guide if you have any clue on this thanks.
I have tried gone through YouTube documentation but didn't got anything on this.
I’m obtaining H.264 video from a DJI drone in an Android library I wrote. From there the video is distributed via WebRTC to many subscribers. This works.
Now one came to the idea if it would be possible to have an RTMP stream aside, so that a parallel publishing of the video to platforms like YT or FB would be possible.
I integrated the code which does H.264 to FLV to RTMP and it works perfectly with at least two open source solutions I have tested: OSS/SRS (https://github.com/ossrs/srs) and node-media-server (https://github.com/illuspas/Node-Media-Server). I publish to instances running here in my LAN and view that by VLC. That works fine.
It doesn’t work if I publish to YT directly. Then I thought I try to insert restream.io into the chain. But it also does not work reliably. Restream at least is a bit more chatty regarding what’s happening, but not chatty enough: What I see is, that I can connect and disconnect - the dashboard window reacts promptly. Same as YT does. I see that the RS dashboard shows bitrate, frame rate and key frame rate and the statistics confirms that. Just - the screen remains black (as with YT) and there is just this spinning wheel.
I can exclude, that I have any kind of weird firewall problem, since I can perfectly uploading H264 as FLV stream using FFMPEG from the command line.
So what is the state: I have two open source RTMP servers, which tell me, all is fine. I have two major public RTMP servers, which don’t say much, but don’t confirm that it works either…
I'm looking for some hints to find out, what is wrong with my stream :)
The simple reason was: YT REQUIRES audio. My stream didn't contain any audio. So I multiplexed a silent fake audio stream into the upload and it worked.
I'm using MobileVLCKit to stream video and audio from Wowza RTMP server. At the same time I'm using VideoCore]1 to stream audio to Wowza RTMP server (I closed off the video channel in VideoCore). Now I'm attempting to make this sort of a teleconferencing solution. Now I'm limited to RTMP or RTSP, not teleconferencing solution (WebRTC or SIP or what not...I am not familiar with these at the moment) because of the limitation on the other end of the line.
The above setup doesn't work. Turning the both functions (video and audio streaming down and audio streaming up) individually runs fine. But not when run simultaneously as audio cannot be heard on the other end. In fact, when app started with VideoCore streaming audio upstream, as soon as I started to downstream via MobileVLCKit, audio cannot be heard on the other end, even though the stream is open. It appears that microphone is somehow wrested away from VideoCore, even though MobileVLC should not need the microphone.
However, when I made the two into two apps and allow them to run in the background (audio & airplay background mode), the two runs fine with one app stream down video & audio and the other picking up microphone voices and stream to the other end.
Is there any reason why the two functions appear to apparently be in conflict within the same app, and any ideas how to resolve the conflict?
I encountered the same problem. Say I have two objects, a vlc player and the other audio processor which listens the microphone. It works fine in simulator for operating both functions in the same time. But conflict in the iPhone device. I think the root cause is that there is only one position or right for listening the microphone. And vlc occupies the right so that my audio processor cannot work. But for some reasons, I cannot modify the vlc code. So I'd to figure out the workaround resolution. And I found one.
The problem comes from vlc which occupies the right but doesn't event use the microphone, and my audio processor did. So the way appears clearly. That is, vlc player plays first and then we new the other object instance, audio processor in my case, which needs to listen the microphone. Since audio processor comes after vlc player, it takes back the right of microphone listening. And they both work properly.
For your reference and hope it can help you.
I am having trouble streaming H.264 video to my FMS v.5 server. I am using an iOS app as the stream source; called streamsie.
Whenever I click the play button, it buffers, but never shows content.
Is there a way to resolve this issue, or do you know how to do so?
Thank you.
H.264 video via RTMP is not supported on iOS. You can find many topics in Adobe forums.
I have RTSP feed coming from my IP Camera as my source input and need to publish it to HTTP Live Streaming (HLS) for playback on iOS devices. I have already tried RTSP-to-RTSP on iOS and it worked; this time I want to try RTSP-to-HLS. I already have Wowza and ffmpeg installed but I just don't know what commands I should run to produce HLS. I've googled for the right commands but I just couldn't find them. What should I run?
Safari supports HLS. Does it also mean that UIWebView supports HLS as well?