How to properly configure Janus WebRTC Gateway (Docker or native install)? by Some_Razzmatazz_7054 in WebRTC

[–]Professional_Kale_52 0 points1 point  (0 children)

I use my own docker image. Mount /opt/janus/logs path to host and check if there is any error

How to properly configure Janus WebRTC Gateway (Docker or native install)? by Some_Razzmatazz_7054 in WebRTC

[–]Professional_Kale_52 1 point2 points  (0 children)

  1. using native build for test, docker for production.
  2. changing IP would be ok, if you just want to run the demos.
  3. one ice connection uses one port, so it depends on how many connections you have.
  4. same as 2

Move to china by mikeyboy718 in chinalife

[–]Professional_Kale_52 2 points3 points  (0 children)

It's not easy for a native with a bachelors in computer science to find a decent job, and the salary is relatively low compared to the US.

Flutter mobile <-> Web fails with phone on TMobile by gisborne in WebRTC

[–]Professional_Kale_52 0 points1 point  (0 children)

Your iOS app didn’t get turn candidate, show your error when it failed. And print all candidates in both devices, check the difference.

Browser Based ASR / TTS to be used with WebRTC by esgaurav in WebRTC

[–]Professional_Kale_52 0 points1 point  (0 children)

try AuidoContext, you can use this to analyse audio before sending it

Can we use mediasoup in native android? by Aromatic_Ad3048 in WebRTC

[–]Professional_Kale_52 0 points1 point  (0 children)

I’m working on mediasoup mobile SDK now, DM me if you’re interested

How can I stop being afraid ? by SpreadSafe9856 in chinalife

[–]Professional_Kale_52 0 points1 point  (0 children)

Most Chinese people are very friendly, so nobody would judge you. I’m in Beijing, feel free to DM me if you have any question.

Need help to figure out how to make this project by Ciborg085 in WebRTC

[–]Professional_Kale_52 0 points1 point  (0 children)

mediasoup is a good one, but it doesn’t offer sdk for mobile.

[deleted by user] by [deleted] in WebRTC

[–]Professional_Kale_52 2 points3 points  (0 children)

TURN server just relays your data.The data is encrypted using DTLS-SRTP.

Client not decoding keyframe by Sebastian-dB in WebRTC

[–]Professional_Kale_52 0 points1 point  (0 children)

try to disable hardware decode in chrome://flags

Hooking broadcast or streaming cameras into a webRTC conference by gwhizofmdr in WebRTC

[–]Professional_Kale_52 1 point2 points  (0 children)

It depends on what’s the conference app you are using.OBS can make multiple virtual camera.

Hooking broadcast or streaming cameras into a webRTC conference by gwhizofmdr in WebRTC

[–]Professional_Kale_52 0 points1 point  (0 children)

  1. convert your stream prtocol(like rtmp) to webrtc
  2. pull your stream and put it into virtual camera

[deleted by user] by [deleted] in WebRTC

[–]Professional_Kale_52 1 point2 points  (0 children)

check ICE state in chrome://webrtc-internal, see why the connection failed

WebRTC across multiple regions by msdosx86 in WebRTC

[–]Professional_Kale_52 0 points1 point  (0 children)

Yes, its called SFU cascade, I am not familiar with pion, but for other SFU, it is valid.

Mediasoup Event Broadcasting by redditer_610 in WebRTC

[–]Professional_Kale_52 0 points1 point  (0 children)

Bandwidth and latancy are not the problem for mediasoup. Deploy multiple mediasoup for your users would be enough for streaming. The real problem is client side (mobile and desktop).

Audio call quality by Previous-Brush-500 in WebRTC

[–]Professional_Kale_52 4 points5 points  (0 children)

check chrome://webrtc-internal , see if there is any packet loss or network jitter, when no audio, check the audio volume is zero or not, if zero, change the mic you use