webrtc

To publish rtmp://ip/live/livestream?k=v to SRS, the correct way to use OBS:

• Server: rtmp://ip/live
• StreamKey: livestream?k=v

However, sometimes the StreamKey has maken lots of people confused, they will literally use as bellow, easpecially for fresh man:

• Server: rtmp://ip/live/livestream?k=v
• StreamKey: Empty

And it's easy to understand.

Solution

So, SRS mi

Summa

Hello!

When text line in list is too long and part of it is moved on the next line, list number is placed on the last line. When I use <br>, list number is placed on the first line of multiline text.

Demo is below:

Demo text:

1. [ ] task 1
2. [ ] task 2 task 2 task 2

Is your feature request related to a problem? Please describe.

Fonoster Voice could benefit from re-using the files generated by a TTS Engine. It will make things faster and reduce costs for the users.

Describe the solution you'd like

The Voice server could maintain a list (perhaps in memory). The list should be a hash of the parameters used to create the transcriptions.

Also,

It would be nice to use shortcuts for common functions, like:

• s to trigger screen sharing (start/stop)
• someKey to switch between available streams
• f to trigger full screen
• c to copy the room url

Currently node selectors follow a strict inheritance. For example, RegionAwareSelector is built on top of SystemLoadSelector. This provides limited flexibility if a user wanted to use CPU load or bandwidth as a metric.

I think it would be good to restructure this to be more flexible, where the selection could happen on multiple metrics (even with region as another dimension). For example, y

Since the URLs are just a gate pass for the video call, there might be a chance of getting intruders into the call with a simple brute force.

Users feel more secure if we have some passcode or approval from the creator to join into call.