Recent content by Azhar Ali

Hi, We use flashphoner server to broadcast content to our listeners using webrtc. We are interested to know if it's possible to use AI text to voice tools and auto stream the generate voice. There are many tools out there with API which will return from mp3 or some format of voice stream...

Hi Max, We did tried to use that but when phone gets locked or App goes hidden, after some time stream would get closed itself. We want it to keep playing in the background until user stops manually. As our usecase is breaking news broadcasting, stream is on all day but only voice when we...

Hi Max, Any update on this? We have managed to get it working using native API's but finding it impossible to connect to media volume. Its working with Call volume but thats not ideal. Have you got an update or guidance on it? We have used this to build it...

Hi Max, thanks for the reply. Is there any indication the timeframe for that ticket?

Hi, We have an app built in react native for our audio only stream. Do you have any guidance on how to implement play only stream using that? Regards Azhar

Hi Max, thanks for the detailed response. I can confirm that was the problem and it's now fixed. Thank you for your help. Regards Azhar

Also to add to the above, I have tried the demo.flashphoner.com and examples are working on the chrome we have so not sure where its going wrong.

Hi Max, thank you for checking this for us. I have updated the server and it still has the same problem. I am getting the same error, its not working on the demo site either. Works on FF I am only testing the simple example of streamer from the demo site. My Chrome build is Version...

Hi, Our users are reporting after updating the chrome, our audio-only stream has stopped working. Getting connection error message "Failed by DTLS error" Any ideas? I have updated chrome on dev machine and its same error. Its quite urgent if you can help. Our server is running on 5.2.768...

Hi Max, Is there any ETA on this? We have been getting regular complaints that WebRTC stream is delaying by 3-4 seconds. Do you think its down to TCP? We never had this issue before we made that change for RTMP stream to work. Regards Azhar

Hi Max, Thanks for checking, it seems to be intermittent. I can confirm, we didn't had any issues either yesterday. Regards Azhar

Hi Max, any update on this? Voice still crackles a bit but nowhere near how it was on udp protocol. Regards Azhar

Hi Max, I have updated the client application to TCP and updated the origin settings as suggested. Should be good to collect the logs. thanks for the help. Regards Azhar

Should i enable tcp for only rtmp stream users?

Would TCP effect the latency on webrtc stream? We cant afford to increase the latency for paid subscribers?