# mod_srs

**Repository Path**: doibest/mod_srs

## Basic Information

- **Project Name**: mod_srs
- **Description**: No description available
- **Primary Language**: Unknown
- **License**: Not specified
- **Default Branch**: master
- **Homepage**: None
- **GVP Project**: No

## Statistics

- **Stars**: 0
- **Forks**: 0
- **Created**: 2024-01-23
- **Last Updated**: 2024-01-23

## Categories & Tags

**Categories**: Uncategorized

**Tags**: None

## README

# mod_srs

FreeSWITCH endpoint module to play and publish video streams from/to [SRS](https://github.com/ossrs/srs).

## Build

Build and install FreeSWITCH. If you never installed FreeSWITCH from source, you can try to do that first <https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Installation/Linux/Debian_67240088/>.

Note, mod_srs need some patches to work with bunded rtp streams, you can find the patches from the master branch in <<https://github.com/rts-cn/xswitch>, see <https://github.com/ossrs/srs/discussions/3625> for more information. Or simply build and install from <https://github.com/rts-cn/xswitch> .

To build and install mod_srs, from any directory, run:

```bash
git clone https://git.xswitch.cn/xswitch/mod_srs.git
cd mod_srs
cmake .
make
make install
```

Load mod_srs

```
fs_cli
freeswitch> load mod_srs
```

## connect to srs

create a srs channel and publish a local video to srs:

```bash
bgapi originate {video_use_audio_ice=true,rtp_payload_space=106,absolute_codec_string=OPUS\,H264,url=http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream}srs/auto_answer &playback(/tmp/test.mp4)
```

or simply

```bash
bgapi originate {url=http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream}srs/auto_answer &playback(/tmp/test.mp4)
```

bridge to a null channel:

```bash
bgapi originate {null_video_codec=H264}null/1234 &bridge({video_use_audio_ice=true,rtp_payload_space=106,absolute_codec_string=OPUS\,H264,url=http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream}srs/auto_answer)
```

call a local sip extension and push to srs:

```bash
bgapi originate user/1006 &bridge({video_use_audio_ice=true,rtp_payload_space=106,absolute_codec_string=OPUS\,H264,url=http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream}srs/auto_answer)
```

or create and extension and call `livestream` from any SIP video phone to push to srs:

```xml
<extension name="srs">
    <condition field="destination_number" expression="^livestream$">
        <action application="bridge" data="{video_use_audio_ice=true,rtp_payload_space=106,absolute_codec_string=OPUS\,H264,url=http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream}xrtc/auto_answer"/>
    </condition>
</extension>
```

screen shot:

![](http://freeswitch.org.cn/images/2023/08/srs_result.png)

pull srs stream from srs and play it to a conference:

```bash
conference 3000 bgdial {video_use_audio_ice=true,rtp_payload_space=106,absolute_codec_string=OPUS\,H264,url=http://localhost:1985/rtc/v1/whip-play/?app=3000&stream=livestream}srs/auto_answer
```

## More information

- [如何在FreeSWITCH中对接SRS](http://freeswitch.org.cn/blog/2023/08/use_srs/) (Chinese)
- https://ossrs.net/lts/zh-cn/docs/v5/doc/introduction
- https://ossrs.net/lts/zh-cn/docs/v4/doc/getting-started
- https://github.com/ossrs/srs/issues/3459
- https://github.com/ossrs/srs/discussions/3625
- https://ossrs.net/lts/zh-cn/docs/v4/doc/http-callback
- https://github.com/ossrs/srs-unity#usage-player