migrate_v4_EN_webrtc

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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

The play will definitely fail if the CANDIDATE of config is not correct, because it's the RTP server address(IP and port) in SDP, the IP is CANDIDATE:

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

So the CANDIDATE is 192.168.3.6 here, which is the IP of server. We could config it by:

• Use plaintext IP directly, such as candidate 192.168.3.6;
• Use command ifconfig to retrieve IP of network interface, pass it by ENV, such as candidate $CANDIDATE;
• Try to read the intranet IP from network interface, for example, candidate *;
• Read from query string of stream URL, such as webrtc://192.168.3.6/live/livestream?eip=192.168.3.6

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:

• Publish：webrtc://d.ossrs.net/live/show
• Play：webrtc://d.ossrs.net/live/show

The streams for SRS usage:

• VLC(RTMP): rtmp://localhost/live/livestream
• H5(HTTP-FLV): http://localhost:8080/live/livestream.flv
• H5(HLS): http://localhost:8080/live/livestream.m3u8
• H5(WebRTC): webrtc://localhost/live/livestream

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:

• WebRTC：webrtc://localhost/live/livestream
• HTTP-FLV：http://localhost:8080/live/livestream.flv

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:

• Publish by WebRTC：webrtc://localhost/live/show
• Play by WebRTC：webrtc://localhost/live/show

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:

• Publish by WebRTC：webrtc://localhost/live/show
• Play by WebRTC：webrtc://localhost/live/show
• HTTP-FLV：http://localhost:8080/live/show.flv
• RTMP by VLC：rtmp://localhost/live/show

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