This document discusses integrating FreeSWITCH with Janus for WebRTC services. It describes Janus as a general purpose WebRTC gateway with plugins that can be used to build applications like video conferencing. The architecture presented uses Janus as a Selective Forwarding Unit (SFU) for Web users, while FreeSWITCH acts as a Multipoint Control Unit (MCU) for legacy SIP users, with both connecting to the same media streams. Some challenges are noted, as directly using SIP is not optimal and managing multiple media connections for each participant is not ideal.

1. INTEGRATING FreeSWITCH
WITH JANUS FOR WEBRTC
SERVICES
Luca Pradovera
ClueCon 2017

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INTRODUCING MYSELF
• Luca Pradovera
• Principal at Mojo Lingo (Atlanta, GA and Milan, Italy)
• Real time application developers
• Adhearsion maintainers
• We build IVRs, dialers, WebRTC apps, mobile apps
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FREESWITCH AND WEBRTC
• Verto is great!
• Outstanding codec support
• Fantastic transcoding
• All the FreeSWITCH features
• It is an MCU
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WEBRTC TOPOLOGIES
• MCU: Multipoint Conferencing Unit
• Media is mixed by the server and everyone gets the same stream
• SFU: Selective forwarding unit
• Each stream is sent to participants separately
• Mesh
• Streams are sent peer to peer (no server involved in the media)
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MCU
• What FreeSWITCH does
• One media stream from
client to server
• One stream out for each
client
• Low bandwidth usage
• Everyone gets the same
video
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SFU
• One media stream from
client to server
• N media streams from
server to client
• Independent handling of
each video stream
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MESH
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DON’T. BUT YOU ALREADY KNEW THAT.

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WEBRTC IS:
• Signaling: JSON, SDP, JSEP
• Connection and NAT traversal: STUN, TURN, ICE
• Media: RTP, SRTP, DTLS
• Codecs: VP8, H264, Opus (not an acronym! Yay!)
• Data: SCTP
• Oh my!
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OUR REQUIREMENTS
• SFU for UI and UX purposes
• SIP legacy support with audio and video
• Recording of simultaneous media streams in one
place
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I HAVE AN HEADACHE NOW
• The goal is to have a general-purpose architecture
• What do we need to build
• Video conferencing SFU with PSTN support
• Streaming and teaching platform
• White-label web conferencing
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SO WHAT DID WE PICK?
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ENTER JANUS
• General purpose WebRTC gateway
• Small footprint
• C-based, all functionality is in plugins
• Developed by MeetEcho
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JANUS CAPABILITIES
• JSON signaling
• RTP/SRTP relaying
• Data messages
• No transcoding
• Recording is in its own format and for single streams
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JANUS FEATURES
• REST or Websocket API if used directly
• Provides a JS library to use in browsers
• Ability to wrap API in your own service
• RabbitMQ/MQTT/UnixSockets
• Admin API to control and monitor instance
• Event emitters
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JANUS PLUGINS
• A plugin is your application or a feature of your
application
• The core only handles the signaling and sessions
• Functionality goes in plugins
• Written in C
• Examples: video room, echo test, SIP interop, screen
cast…
• The SIP plugin uses the Sofia stack
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SAMPLE INTERACTION
• POST to server root to get session ID
• POST to attach a plugin (activate a feature)
• Plugin-level API interactions (create room, join room,
kick participant…)
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SOUNDS GREAT. WHAT ABOUT
FREESWITCH?
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BACK TO OUR USE CASE
• Providing SFU to web users and MCU to legacy users
• Until Unified Plan multicast comes, we have to make
one connection for each participant to the SFU
• Currently using SIP, moving to Verto
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OUR ARCHITECTURE
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SFU Media
SIP
RTP
MCU Media
SIP

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FREESWITCH SETUP
• Just using mod_conference
• No actual actions, everything is handled through
another API
• SIP users are in the FS conference only
• H.264 and Opus only, low resolution streams
(640x480)
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CURRENT CHALLENGES
• There is no real reason to use SIP
• The media that comes from the SIP side is the MCU
video
• Multiple media connections are not exactly the best
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THANK YOU!
My name is: Luca Pradovera
I am a Voice Application Developer
at Mojo Lingo.
Web: https://mojolingo.com
Twitter: @lucaprado
GitHub: lpradovera
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