baresip-webrtc
Baresip WebRTC Demo
-
Install libre and librem
-
Install baresip dev:
$ sudo make install install-dev -C ../baresip
- Compile this project:
make
- Start it:
$ ./baresip-webrtc
Local network address: IPv4=en0|10.0.1.12
medianat: ice
mediaenc: dtls_srtp
aucodec: opus/48000/2
aucodec: G722/16000/1
ausrc: aufile
auplay: aufile
vidcodec: H264
vidcodec: H264
vidcodec: H263
vidcodec: H265
avcodec: using H.264 encoder 'libx264' -- libx264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10
avcodec: using H.264 decoder 'h264' -- H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10
avcodec: using H.265 encoder 'libx265' -- libx265 H.265 / HEVC
avcodec: using H.265 decoder 'hevc' -- HEVC (High Efficiency Video Coding)
vidsrc: fakevideo
vidisp: fakevideo
ausrc: avformat
vidsrc: avformat
ausrc: ausine
demo: listening on HTTP 0.0.0.0:9000
demo: listening on HTTPS 0.0.0.0:9001
- Open this URL in Chrome and follow the instructions:
http://localhost:9000/
Protocol Diagram
This diagram shows how a WebRTC capable browser can connect to baresip-webrtc. Baresip-WebRTC has a small embedded HTTP(S) Server for serving JavaScript files and for signaling.
The media stream is compatible with WebRTC, using ICE and DTLS/SRTP as media transport. The audio codecs are Opus, G722 or G711. The video codecs are VP8, H264.
(Signaling)
.----------. SDP/HTTP .-----------.
| Browser |<-------------------->| Baresip |
| (Chrome) | | WebRTC |<==== A/V Backend
| |<====================>| |
'----------' ICE/DTLS/SRTP '-----------'
(Audio,Video)
API Mapping
| WebRTC: | this: |
|---|---|
| MediaStream | n/a |
| MediaStreamTrack | struct media_track |
| RTCConfiguration | struct configuration |
| RTCPeerConnection | struct peer_connection |
| RTCSessionDescription | struct session_description |
| RTCRtpTransceiver | struct stream |