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Version: 4.0

WebRTC

https://github.com/ossrs/srs/issues/307

Config

There are some config for WebRTC:

  • full.conf: Section rtc_server and vhost rtc.vhost.srs.com is about WebRTC.
  • rtc.conf: WebRTC to WebRTC clients.
  • rtmp2rtc.conf: Covert RTMP to WebRTC.
  • rtc2rtmp.conf: Covert WebRTC to RTMP.

Config: Candidate

Please note that candidate is essential important, and most failure is caused by wrong candidate, so be careful.

As it shows, candidate is server IP to connect to, SRS will response it in SDP answer as candidate, like this one:

type: answer, sdp: v=0
a=candidate:0 1 udp 2130706431 192.168.3.6 8000 typ host generation 0

So the 192.168.3.6 8000 is an endpoint that client could access. There be some IP you can use:

  • Config as fixed IP, such as candidate 192.168.3.6;
  • Use ifconfig to get server IP and pass by environment variable, such as candidate $CANDIDATE;
  • Detect automatically, first by environment, then use server network interface IP, such as candidate *;, we will explain at bellow.
  • Specify the ?eip=x in URL, such as: webrtc://192.168.3.6/live/livestream?eip=192.168.3.6
  • Normally API is provided by SRS, so you're able to use hostname of HTTP-API as candidate, we will explain at bellow.

Configurations for automatically detect the IP for candidate:

  • candidate *; or candidate 0.0.0.0; means detect the network interface IP.
  • use_auto_detect_network_ip on; If disabled, never detect the IP automatically.
  • ip_family ipv4; To filter the IP if automatically detect.

Configurations for using HTTP-API hostname as candidate:

  • api_as_candidates on; If disabled, never use HTTP API hostname as candidate.
  • resolve_api_domain on; If hostname is domain name, resolve to IP address. Note that Firefox does not support domain name.
  • keep_api_domain on; Whether keep the domain name to resolve it by client.

Note: Please note that if no candidate specified, SRS will use one automatically detected IP.

In short, the candidate must be a IP address that client could connect to.

Use command ifconfig to retrieve the IP:

# For macOS
CANDIDATE=$(ifconfig en0 inet| grep 'inet '|awk '{print $2}')

# For CentOS
CANDIDATE=$(ifconfig eth0|grep 'inet '|awk '{print $2}')

# Directly set ip.
CANDIDATE="192.168.3.10"

Pass it to SRS by ENV:

env CANDIDATE="192.168.3.10" \
./objs/srs -c conf/rtc.conf

For example, to run SRS in docker, and setup the CANDIDATE:

export CANDIDATE="192.168.3.10"
docker run --rm --env CANDIDATE=$CANDIDATE \
-p 1935:1935 -p 8080:8080 -p 1985:1985 -p 8000:8000/udp \
ossrs/srs:4 \
objs/srs -c conf/rtc.conf

Note:About the usage of srs-docker, please read srs-docker.

Stream URL

Online demo URL:

The streams for SRS:

HTTP API

About the API for WebRTC, please read publish and play.

RTMP to RTC

Please use conf/rtmp2rtc.conf as config.

export CANDIDATE="192.168.1.10"
docker run --rm --env CANDIDATE=$CANDIDATE \
-p 1935:1935 -p 8080:8080 -p 1985:1985 -p 8000:8000/udp \
ossrs/srs:4 \
objs/srs -c conf/rtmp2rtc.conf

Note: Please set CANDIDATE as the ip of server, please read CANDIDATE.

Use FFmpeg docker to push to localhost:

docker run --rm --network=host ossrs/srs:encoder ffmpeg -re -i ./doc/source.flv \
-c copy -f flv rtmp://localhost/live/livestream

Play the stream in browser:

RTC to RTC

Please use conf/rtc.conf as config.

export CANDIDATE="192.168.1.10"
docker run --rm --env CANDIDATE=$CANDIDATE \
-p 1935:1935 -p 8080:8080 -p 1985:1985 -p 8000:8000/udp \
ossrs/srs:4 \
objs/srs -c conf/rtc.conf

Note: Please set CANDIDATE as the ip of server, please read CANDIDATE.

Play the stream in browser:

Remark: Note that if not localhost, the WebRTC publisher should be HTTPS page.

RTC to RTMP

Please use conf/rtc2rtmp.conf as config.

export CANDIDATE="192.168.1.10"
docker run --rm --env CANDIDATE=$CANDIDATE \
-p 1935:1935 -p 8080:8080 -p 1985:1985 -p 8000:8000/udp \
ossrs/srs:4 \
objs/srs -c conf/rtc2rtmp.conf

Note: Please set CANDIDATE as the ip of server, please read CANDIDATE.

The streams:

SFU: One to One

Please use conf/rtc.conf as config.

export CANDIDATE="192.168.1.10"
docker run --rm --env CANDIDATE=$CANDIDATE \
-p 1935:1935 -p 8080:8080 -p 1985:1985 -p 8000:8000/udp \
ossrs/srs:4 \
objs/srs -c conf/rtc.conf

Note: Please set CANDIDATE as the ip of server, please read CANDIDATE.

Then startup the signaling, please read usage:

docker run --rm -p 1989:1989 ossrs/signaling:1

Use HTTPS proxy httpx-static as api gateway:

export CANDIDATE="192.168.1.10"
docker run --rm -p 80:80 -p 443:443 ossrs/httpx:1 \
./bin/httpx-static -http 80 -https 443 -ssk ./etc/server.key -ssc ./etc/server.crt \
-proxy http://$CANDIDATE:1989/sig -proxy http://$CANDIDATE:1985/rtc \
-proxy http://$CANDIDATE:8080/

To open http://localhost/demos/one2one.html?autostart=true

Or by the IP https://192.168.3.6/demos/one2one.html?autostart=true

Note: For self-sign certificate, please type thisisunsafe to accept it.

SFU: Video Room

Please follow SFU: One to One, and open the bellow demo pages.

To open http://localhost/demos/room.html?autostart=true

Or by the IP https://192.168.3.6/demos/room.html?autostart=true

Note: For self-sign certificate, please type thisisunsafe to accept it.

Room to Live

Please follow SFU: One to One, and please convert RTC to RTMP, for FFmpeg to mix the streams.

export CANDIDATE="192.168.1.10"
docker run --rm --env CANDIDATE=$CANDIDATE \
-p 1935:1935 -p 8080:8080 -p 1985:1985 -p 8000:8000/udp \
ossrs/srs:4 \
objs/srs -c conf/rtc2rtmp.conf

If use FFmpeg to mix streams, there is a FFmpeg CLI on the demo page, for example:

ffmpeg -f flv -i rtmp://192.168.3.6/live/alice -f flv -i rtmp://192.168.3.6/live/314d0336 \
-filter_complex "[1:v]scale=w=96:h=72[ckout];[0:v][ckout]overlay=x=W-w-10:y=H-h-10[out]" -map "[out]" \
-c:v libx264 -profile:v high -preset medium \
-filter_complex amix -c:a aac \
-f flv rtmp://192.168.3.6/live/merge

Input:

  • rtmp://192.168.3.6/live/alice
  • rtmp://192.168.3.6/live/314d0336

Output:

  • rtmp://192.168.3.6/live/merge

Winlin 2020.02