-
AVPacket can't write to WebRTC directly because of this.
test code
```golang
func TestRTMPConnect(t *testing.T) {
assert := assert.New(t)
conn, err := rtmpUtil.Dial("some url")
assert.NoErr…
-
Would be really powerful if we can stream a vlc stream (or any http video stream) through here.
BTW: awesome project!
-
https://alexxit.github.io/go2rtc/api/#tag/Produce-stream
Please provide an example for the purpose/use case of this API.
Thanks
-
```
Hello!
Thank you for this project!
Siprtmp looks good as flash2sip gateway. But it uses tcp for rtmp part.
Initial version of rtmfp code on python as added to repository! Maybe it's time
for ne…
-
```
Hello!
Thank you for this project!
Siprtmp looks good as flash2sip gateway. But it uses tcp for rtmp part.
Initial version of rtmfp code on python as added to repository! Maybe it's time
for ne…
-
We'll be focused on making improvements that help viewers have a reliable experience viewing streams on Glimesh, even in poorer network conditions. One of the main objectives will be supporting the Gl…
-
> Note: Please read FAQ before file an issue, see #2716
## Description
Please description your issue here
1. SRS Version: 6.0
2. SRS Log:
```
2023-05-23 11:03:19.371][INFO][805727][3f7…
-
### Short description
We have an app that as soon a streams starts it republishes to 2 output locations, one to nimble and the other one to an internal Ant App. After a few days (so a lot of stre…
mekya updated
3 years ago
-
Hi.
Your experimens are great but i didn't find solution for my trouble.
I need to translate real-time audiostream from WebRTC-supported browser to ffmeg and later to rtmp media server. Reason i…
-
clone git and build it successfully.
run it on android 9 os. push mp4 video file with following command:
./ffmpeg -re -i video.mp4 -c copy -f flv -y rtmp://ip:1935/live/mp4test
then webrtc://ip/l…