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https://github.com/q191201771/lalmax
lalserver with srt/webrtc/whip/whep/http-fmp4/hls(fmp4)/low latency hls(llhls)/gb28181. 卍解
https://github.com/q191201771/lalmax
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lalserver with srt/webrtc/whip/whep/http-fmp4/hls(fmp4)/low latency hls(llhls)/gb28181. 卍解
- Host: GitHub
- URL: https://github.com/q191201771/lalmax
- Owner: q191201771
- License: mit
- Created: 2023-04-07T05:15:00.000Z (over 1 year ago)
- Default Branch: master
- Last Pushed: 2024-07-24T08:30:36.000Z (2 months ago)
- Last Synced: 2024-07-24T10:00:20.109Z (2 months ago)
- Language: Go
- Homepage:
- Size: 3.29 MB
- Stars: 84
- Watchers: 7
- Forks: 17
- Open Issues: 4
-
Metadata Files:
- Readme: README.md
- License: LICENSE
Awesome Lists containing this project
README
# lalmax
lalmax是在lal的基础上集成第三方库，可以提供SRT、RTC、mp4、gb28181、onvif等解决方案
# 编译
./build.sh
# 运行
./run.sh或者./lalmax -c conf/lalmax.conf.json
# 配置说明
lalmax.conf.json配置主要由2部分组成
(1) lal_config_path表示lal配置文件的路径,用于加载lal本身的配置,具体配置说明见[lal配置](https://pengrl.com/lal/#/ConfigBrief)
(2) 剩余的配置则为lalmax的配置,具体配置说明见[config.md](./document/config.md)
# docker运行
```
docker build -t lalmax:init ./
docker run -it -p 1935:1935 -p 8080:8080 -p 4433:4433 -p 5544:5544 -p 8083:8083 -p 8084:8084 -p 30000-30100:30000-30100/udp -p 1290:1290 -p 6001:6001/udp lalmax:init
```
# 架构
# 支持的协议
## 推流
(1) RTSP
(2) SRT
(3) RTMP
(4) RTC(WHIP)
(5) GB28181
具体的推流url地址（除了srt/whip）
https://pengrl.com/lal/#/streamurllist
## 拉流
(1) RTSP
(2) SRT
(3) RTMP
(4) HLS(S)-TS
(5) HTTP(S)-FLV
(6) HTTP(S)-TS
(7) RTC(WHEP)
(8) HTTP(S)-FMP4
(9) HLS(S)-FMP4/LLHLS
具体的拉流url地址见https://pengrl.com/lal/#/streamurllist（除了srt/whep）
## [SRT](./document/srt.md)
（1）使用gosrt库
（2）暂时不支持SRT加密
（3）支持H264/H265/AAC
（4）可以对接OBS/VLC
```
推流url
srt://127.0.0.1:6001?streamid=#!::h=test110,m=publish
拉流url
srt://127.0.0.1:6001?streamid=#!::h=test110,m=request
```
## [WebRTC](./document/rtc.md)
（1）支持WHIP推流和WHEP拉流,暂时只支持POST信令
（2）支持H264/G711A/G711U/OPUS
（3）可以对接OBS、vue-wish
（4）WHEP支持对接Safari HEVC
（5）支持datachannel,只支持对接jessibuca播放器
（6）WHIP支持对接OBS 30.2 beta HEVC
datachannel播放地址：webrtc://127.0.0.1:1290/webrtc/play/live/test110
```
WHIP推流url
http(s)://127.0.0.1:1290/webrtc/whip?streamid=test110
WHEP拉流url
http(s)://127.0.0.1:1290/webrtc/whep?streamid=test110
```
## Http-fmp4
(1) 支持H264/H265/AAC/G711A/G711U
```
拉流url
http(s)://127.0.0.1:1290/live/m4s/test110.mp4
```
## HLS(fmp4/Low Latency)
(1) 支持H264/H265/AAC/OPUS
```
拉流url
http(s)://127.0.0.1:1290/live/hls/test110/index.m3u8
```
## [GB28181](./document/gb28181.md)
(1) 作为SIP服务器与设备进行SIP交互,使用单端口/多端口收流
(2) 提供相关API获取设备信息、播放某通道等功能
(3) 支持H264/H265/AAC/G711A/G711U
(4) 支持TCP/UDP
## Onvif
(1) onvif接入设备进行pull拉流
(2) 支持tcp/udp拉流
(3) pull api:http://127.0.0.1:1290/api/ctrl/onvif/pull
```
{
"addr": , // 设备IP:PORT
"username": , // 用户名
"password": , // 密码
"rtspmode": , // rtsp拉流模式,0-tcp, 1-udp
"pullallprofiles": // 是否请求所有的profiles
}
```
## Room
（1）集成livekit实现房间功能