SRT Live Server

Introduction

srt-live-server (SLS) is an open source live streaming server for low latency based on Secure Reliable Tranport (SRT). Normally, the latency of transport by SLS is less than 1 second in internet.

Requirements

Please install the SRT library first, refer to SRT for system enviroment setup. SLS can only run on Unix-based operating systems.

Compilation

git submodule update --init
mkdir build && cd build
cmake ../ -DCMAKE_BUILD_TYPE=Release
make -j

Binaries are created in build/bin/ directory.

Usage

cd build

Help information

./srt_server -h

Run with default configuration file

./srt_server -c ../sls.conf

Configuration

Configuration directives are documented on the wiki page.

SRTLA / Bonded Connection Support

SRT Live Server supports both SRTLA (bonded cellular) and direct SRT connections on the same server using separate publisher ports:

server {
    listen_player 4000;               # All streams playable here
    listen_publisher 4001;            # Direct SRT (OBS, FFmpeg)
    listen_publisher_srtla 4002;      # SRTLA/bonded (via srtla_rec)
    ...
}

• listen_publisher (for direct SRT connections, standard behavior)
• listen_publisher_srtla (for SRTLA/bonded connections, enables SRTLA patches automatically)
• listen_player (playback for streams from both publisher types)

Multiple ports per role listen_player, listen_publisher, and listen_publisher_srtla each accept more than one port. Provide a comma separated list, inclusive ranges (a-b), or a mix. One listener is created per port, so a client may connect on any of them.

server {
    listen_player 4000,4010,5000-5005;   # players may connect on any of these
    listen_publisher 4001;
    ...
}

Why separate ports? SRTLA bonded connections require special SRT patches that disable dynamic reorder tolerance and periodic NAK reports. Using the wrong setting causes glitching:

• Direct SRT with SRTLA patches = dropped packets
• SRTLA without patches = spurious retransmissions

Testing

srt-live-server only supports the MPEG-TS format streaming.

Test with FFmpeg

You can push camera live stream using FFmpeg. FFmpeg must be compiled with --enable-libsrt flag - to obtain appropriate binaries, download FFmpeg sourcecode from https://github.com/FFmpeg/FFmpeg, then compile FFmpeg with --enable-libsrt.

srt library is installed in folder /usr/local/lib64.

If ERROR: srt >= 1.3.0 not found using pkg-config occurs during the compilation of FFmpeg, please check the ffbuild/config.log file and follow its instruction to resolve this issue. In most cases it can be resolved by executing the following command:

export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig:/usr/local/lib64/pkgconfig

If error while loading shared libraries: libsrt.so.1 occurs, please add srt library path to the runtime linker configuration file, /etc/ld.so.conf, then refresh the cache by running the comand /sbin/ldconfig as root.

Push stream from webcam to SRT

./ffmpeg -f avfoundation -framerate 30 -i "0:0" -vcodec libx264  -preset ultrafast -tune zerolatency -flags2 local_header  -acodec libmp3lame -g  30 -pkt_size 1316 -flush_packets 0 -f mpegts "srt://[your.sls.ip]:8080?streamid=uplive.sls/live/test"

Play a SRT stream using FFplay

./ffplay -fflags nobuffer -i "srt://[your.sls.ip]:8080?streamid=live.sls/live/test"

Test with OBS

OBS supports SRT protocol to publish streams from version v25.0 onwards. To publish SRT stream from OBS to SRT Live Server you can use the following url:

srt://[your.sls.ip]:8080?streamid=uplive.sls/live/test

You can also add a SRT stream as an input source. To do this, add a Media source to OBS, enter mpegts as input format and set the following input URL:

srt://[your.sls.ip]:8080?streamid=live.sls/live/test

Test with SRT Live Client

There is a test tool in SLS which can be used as a performance test - it has no codec overhead, only network overhead. The SRT Live Client can play a SRT stream to a TS file, or push a TS file to a SRT stream.

Push a TS file via SRT

./srt_client -r srt://[your.sls.ip]:8080?streamid=uplive.sls/live/test -i [the full file name of exist ts file]

Play a SRT stream

./srt_client -r srt://[your.sls.ip]:8080?streamid=live.sls/live/test -o [the full file name of ts file to save]

Use SLS with docker

Please refer to: https://hub.docker.com/r/ravenium/srt-live-server

Development

To build a debug build of the SRT Live Server, run the following commands:

git submodule update --init
mkdir build && cd build
cmake ../ -DCMAKE_BUILD_TYPE=Debug
make -j

Note:

• SLS refers to the RTMP url format (domain/app/stream_name), example: www.sls.com/live/test. The URL must be set in streamid parameter of SRT, which will be the unique identification a stream.

• How to distinguish the publisher and player of the same stream? In the configuration file file, you can set parameters of domain_player/domain_publisher and app_player/app_publisher to resolve it. Importantly, the two combination strings of domain_publisher/app_publisher and domain_player/app_player must not be equal in the same server block.

• I supplied a simple android app for testing SLS, which can be downloaded from https://github.com/Edward-Wu/liteplayer-srt