Search results

Hi Max, We still experiencing choppy, crackly sound on rtmp stream. I have emailed you a video on the demo and SSH details for the origin and edge server (this is only one edge which runs these days). Please can we look into this? First day after the update it was fine but since last 2-3 days...

We update the origin to latest version 768 and voice on rtmp seems to be good as well. Not sure why it went bad as when we deployed 744 few weeks ago, we didnt had any issues until last few days.

But if it was the ISP provider issue, it should happen on WebRTC as well. It's not a random issue, voice is continuously worse on the rtmp re-publishing Today, I could not see any packet lost but the sound is still choppy. Voice is very good on WebRTC but not on delayed rtmp re-published stream.

Hi Max, What do you mean by channel issue? and what can we do about it? Is updating the origin only enough? I can confirm all the settings on the mentioned post are present. Regards Azhar

Hi Max, We have been having issues between the quality of the sound between webrtc stream and LITE stream. We have a delay of 10sec using the buffer settings but the quality of sound is very poor on the rtmp but its crispy and clear on the webrtc. I have emailed you the link of the recorded...

Thanks Max, that fixed the issue. Delay has been working nicely past couple days. Thanks for all the help. Really appreciate the hard work you do.

Hi max, You misunderstood me..what I meant was client is connected to the rtmp stream but after 2mins it disconnects itself. As suggested above the rtp timer is also 2mins and its audio only stream. Could the video side of things disconnecting it as there are packets for that?

Hi Max, Another problem we are having is, rtmp_stream auto disconnects after roughly 2mins. In our case there might be times when noone is talking for several mins, as its a news service. How can we stop that? It doesn't happen for the WebRTC stream.

Hi Max, I don't have this file on my server at this location. /usr/local/FlashphonerWebCallServer/media_transponder.sdp Is that correct? I can create it and would it get picked up itself as its present now? thanks for the help

Hi Max, I have updated the live servers and in terms of delay all is working fine. We are starting with 10second delay. However, we have an issue with the quality of the sound on the rtmp version of the stream. WebRTC to WebRTC audio quality if pretty sharp but the RTMP version, audio comes...

Thanks Max, that probably was the reason for it. I will do the test again. What's the RTP activity for? Do we need it for audio-only streams? There are few settings for rtp to disabled rtp_activity_audio , rtp_activity_video, is both needs to be turned off? Is it enough to turn them off on...

Hi Max, Update to 744 and tested and it still does not work for audio-only stream. I modified the media devices examples to republish to rmtp as well. I tested with audio only stream and same problem happened. When playing the rtmp_streamName, the player stayed blank and did not play any audio...

Thanks max, will test. One question, does origin needs to update only or edge servers needs to be on 744 as well. Just checking as we would need to update them and then create templates etc. for autoscale purpose.

Thanks Max. I completely understand and appreciate your effort. Will wait for the update with the fix.

Thanks for letting me know. I will be patient but do you have an estimate on it. We have a new service which we want to put live in about 23 days and this a key part of it. We do have a workaround to do multiple streams delayed one from OBS but OBS one sometimes gets choppy sound as they both...

Hi Max, I modified the media devices examples to republish to rmtp as well. I tested with audio only stream and same problem happened. When playing the rtmp_streamName, the player stayed blank. So it seems bufferization only works when both video and audio tracks are available. I tested this...

Hi Max, thanks for that. I will test this with audio-only as this is our use case but I was testing this on demo app just for ease. Regards Azhar

Hi Max, I set up a test server and installed v728 on it. I updated the settings to the following: Test 1 rtmp_transponder_stream_name_prefix =rtmp_ rtmp_transponder_full_url=false rtmp_out_buffer_enabled=true rtmp_out_buffer_start_size=30000 rtmp_out_buffer_initial_size=30000 I modified the...

thanks

Thats interesting, will def give it a go..seems like most solid solution. I take it if I adjust the buffer parameters I can add a delay of 1min? And rmtp should work with cdn setup we have?