-
**Description**'
Please ensure that the markdown structure is maintained.
> Local use of SRS for pushing and pulling streams is normal, but after deploying to the server, RTMP pushing and pulling…
-
-
Hello,
I'm worried, this CPU usage is normal? OME has only 5 incoming streams and 1 outgoing.
I'm afraid to use OME on production... Other solutions is using ~20% CPU max.
There is only one outpu…
-
Configuration
```
# main config for srs.
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
#srs_log_tank file;
#srs_log_file ./objs/srs.log;…
-
**Description**'
Please ensure that the markdown structure is maintained.
Memory increasing unlimited when using a gb28181 input stream and view output strem by webrtc
> Please describe the issue …
-
> Note: Before asking a question, please read the FAQ (Please read FAQ before filing an issue) https://github.com/ossrs/srs/issues/2716
**Description**
> Previously, I compiled version 4.0.161 and…
-
### 现象描述
调用`addStreamProxy`接口时,如果拉流正在处理,再次提交相同的请求时,会提前返回
例如,如果向`http://10.1.123.123:9000/index/api/addStreamProxy` 接口提交了GET 请求
```
GET /index/api/addStreamProxy?secret=abcde&url=rtmp:%…
-
### Describe the problem you are having
Good morning everyone, if you saw my other support post i recently started a very basic deployment of Frigate, and after some self imposed stupidity its now up…
-
## Description
Please description your issue here
In the process of rtmp/srt-WebRTC with AAC-opus, the opus bitrate is too low, it seems to be only 64Kbps, causing severe audio quality loss. I hop…
-
**Description**
Platforms like Facebook abroad require encrypted transmission for security reasons, and they require support for the RTMPS protocol for streaming. It is hoped that SRS can also add …