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sipML5 - Janus Gateway
Asterisk WebRTC frontier:
make client SIP Phone with
Alessandro Polidori
@ale_polidori
Fosdem 2019 - Brussels
Realtime DevRoom
Alessandro Polidori
Software Engineer @Nethesis
#Node.js #WebRTC #OpenSource
@ale_polidori
alepolidori
https://medium.com/@ale_polidori
ale_polidori
WebRTC: Web Real Time Communication
● 2011 Google project
● Javascript API for audio/video communications
● 3 Components
○ getUserMedia: camera, microphone, screen access
○ RTCPeerConnection: negotiation, encoding, decoding, nat traversal
○ RTCDataChannel: exchange data between browsers
● Other APIs: getStats, MediaRecorder
ale_polidori
Protocols
● 2 groups
○ RTP → transport
○ SIP → signaling
● SRTP
○ secure real-time transport protocol
○ encryption
○ message authentication
● SDP - audio & video description
● STUN, TURN, ICE
Architecture
VoIP
Provider
Company
network
Web App
Internet
NethVoice PBX (Asterisk)
VoIP Gateway
router
PSTN
ale_polidori
CoDec
● Reduction / Compression / Decompression of data flow
● Bandwidth / Quality (MOS) / Latency
● Audio
○ G.711 (64 kbps)
○ Opus (6-510 kbps - dynamic bitrate)
● Video
○ VP8, VP9, AV1
○ H.264
sipML5
ale_polidori
sipML5
● First Open Source HTML5 SIP Client (Doubango Telecom)
● 100% Javascript: NO PLUGIN !!!
● Media stack on WebRTC
● SIP over WebSocket (UDP, TCP, TLS)
● Audio / Video Calls / Instant Messaging / Screen share
● Desktop & Mobile
● Google I/O 2012
ale_polidori
sipML5 Architecture
Javascript SIP
Javascript SDP
WebRTC
websocket
UDP/TCP/TLS
SRTP/SRTCP/ICE
HTML5 Client
PSTN
Sip Net
NethVoice PBX (Asterisk)
ale_polidori
sipML5: how to use
1. Engine initialization
2. Start SIP Stack
3. Extension registration
4. Start Audio/Video call
ale_polidori
sipML5: how to use
1. Engine initialization
2. Start SIP Stack
3. Extension registration
4. Start Audio/Video call
ale_polidori
sipML5: how to use
1. Engine initialization
2. Start SIP Stack
3. Extension registration
4. Start Audio/Video call
ale_polidori
sipML5: how to use
1. Engine initialization
2. Start SIP Stack
3. Extension registration
4. Start Audio/Video call
...more code...
ale_polidori
sipML5: the library
ale_polidori
1. sipML5: engine initialization
ale_polidori
2. sipML5: start SIP stack
ale_polidori
2. sipML5: start SIP stack
ale_polidori
2. sipML5: start SIP stack
ale_polidori
2. sipML5: start SIP stack
ale_polidori
2. sipML5: start SIP stack
ale_polidori
2. sipML5: start SIP stack
ale_polidori
3. sipML5: extension registration
ale_polidori
4. sipML5: call
Janus Gateway
ale_polidori
Janus
● Gateway general purpose by Meetecho
● WebRTC audio/video
● JSON messages
● Plugin Architecture → SIP Plugin
● Monitoring
● Interfaces HTTP, WebSocket, RabbitMQ
ale_polidori
server
Janus Architecture
janus.js
PBX (Asterisk)
HTTPS
Apache ProxyPass
UDP/TCP/TLS
HTML5 Client
PSTN
Sip Net
ale_polidori
Janus: how to use
1. Engine initialization
2. Create a session
3. Link SIP plugin
4. Start Audio/Video call
ale_polidori
Janus: how to use
1. Engine initialization
2. Create a session
3. Link SIP plugin
4. Start Audio/Video call
ale_polidori
Janus: how to use
1. Engine initialization
2. Create a session
3. Link SIP plugin
4. Start Audio/Video call
ale_polidori
Janus: how to use
1. Engine initialization
2. Create a session
3. Link SIP plugin
4. Start Audio/Video call
...more code...
ale_polidori
Library
● WebRTC adapter
webrtc/adapter
● Janus Client lib
meetecho/janus-gateway
ale_polidori
1. Janus: engine initialization
ale_polidori
2. Janus: create a session
ale_polidori
3. Janus: link SIP plugin
Handle to interact with plugin
ale_polidori
4. Janus: audio/video call
Call destination
VoIP PBX
ale_polidori
NethServer VoIP PBX
● NethServer Linux distro
● Asterisk & FreePBX based
● Open Source - community.nethserver.org
● NethVoice Enterprise version
● NethCTI WebApp
Demo
ale_polidori
Start your NethServer VoIP PBX1
ale_polidori
go to:
https://alepolidori.github.io/janus-webrtc-phone2
References
https://github.com/alepolidori/fosdem-2019
https://bloggeek.me
@ale_polidori
alepolidori
https://medium.com/@ale_polidori
Thank you !

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Asterisk WebRTC frontier: make client SIP Phone with sipML5 and Janus Gateway