The document discusses the integration of Janus and FreeSWITCH, two open-source real-time communication projects, to build a SIP/WebRTC conferencing system suitable for heterogeneous environments. It highlights the differences between SFU (Selective Forwarding Unit) and MCU (Multipoint Control Unit) approaches, detailing their use cases and benefits in real-time communications. The presentation also outlines potential challenges, integrations, and benefits of combining Janus and FreeSWITCH to leverage their unique capabilities in delivering an efficient conferencing solution.

1. An SFU/MCU integration for
heterogeneous environments
Building a SIP/WebRTC conferencing system with Janus
and FreeSWITCH
Giacomo Vacca - RTC Architect at Nexmo

2. Introduction

3. Overview
Janus and FreeSWITCH are two open source real-time communications
projects that can be used to build conferencing systems.
Among other things, Janus implements an SFU (Selective Forwarding Unit).
FreeSWITCH includes a conference module, which works as MCU (Multipoint
Control Unit).
Often we are asked to build solutions integrating existing applications and
services: this presentation shows an approach and example.

4. The three big integration points
WebRTC / SIP
SFU / MCU
Janus / FreeSWITCH

5. About me
Started working in the VoIP area in 2001, and since have worked for Truphone,
Libon and others, while currently on the SIP infrastructure at Nexmo.
User, promoter and contributor of open source projects in the RTC field.
I design and maintain solutions based on Kamailio, FreeSWITCH, Asterisk,
RTPEngine, Janus, Homer, and love debugging/troubleshooting.

6. About Nexmo
Nexmo provides API for RTC: messaging, voice, video, and many other features.
The SIP stack connects the Nexmo Voice API with the external world (SIP
customers and PSTN carriers).
Programmable SIP and SIP Connect: Link a DID to an application, reach it from
PSTN or SIP, and program the user experience of your customers.
The core SIP stack is built on Kamailio, Asterisk, nginx, with Voice API providing
the main features with FreeSWITCH. Hepic (Homer) for troubleshooting and
QoS analysis.

7. Try it out!
https://www.nexmo.com/voice
Coupon code:
JNCNX19
with €10 credit, available from Sep 22nd to Oct 25th.

8. Why combining MCU and SFU?
MCU integrates features on the server - heavier processing -
more suitable for traditional/mobile clients
SFU routes streams but delegates features to clients - lighter
processing - suitable for modern browsers

9. About FreeSWITCH

10. FS as MCU and gateway to SIP/PSTN
FS implements an MCU with mod_conference
FS also integrates SIP clients (and so PSTN) and
WebRTC clients (via mod_verto) directly.
Currently FS does not offer an SFU approach.
FS is widely adopted as B2BUA and conference
server.
The MCU approach allows for creative mixing of
audio/video.
A single stream for the entire conference is well
suited for mobile clients and other “traditional”
VoIP clients.

11. Not just SIP
endpoints
FS supports:
- SIP endpoints
- PSTN GWs
- WebRTC clients (verto protocol)
- WebSockets to exchange media
- Opus
- Transcoding
About FS and opus:
https://freeswitch.org/confluence/display/FREESWI
TCH/FreeSWITCH+And+The+Opus+Audio+Codec

12. FS operations for this project
SIP and verto
At ClueCon 2014 on using Kamailio and FS to
build a WebRTC service:
https://www.slideshare.net/GiacomoVacca/top-
5-challenges-to-add-web-calls-to-truphone-voip-
platform
JsSIP + WebSocket + Ephemeral Auth + coturn
verto protocol:
- JSON-based, exchanging SDP and
other call information
- verto.invite, verto.media, verto.answer,
verto.bye, verto.attach
Great documentation here:
http://evoluxbr.github.io/verto-docs/

13. FS - mod_conference
- Conferencing (MCU)
- Recording
- Whisper, ASR
- Opus + Opus at low sample rate for
PSTN*
- Canvas composition
- Single stream to each participant
- Transcoding
- Mute/Unmute streams (see relate
later)
* About Opus session negotiation:
https://www.giacomovacca.com/2016/09/opus-negotiation-for-practical-man.html

14. About Janus

15. How does the Janus Videoroom work?
SFU implementation
Subscriber/Publisher pattern
“A router of streams”
Lightweight processing of streams
Many opportunities to design the UI
on the client side
Main operations:
- Client creates a transaction
- Attaches to the plugin (PeerConnection)
- Joins a room (as a Publisher once or as a
Subscriber many times)
- Configure audio (then exchange media)

16. The PoC

17. PoC - High level architecture
FreeSWITCH Janus
Controller
API
verto(W
ebSocket)
WebRTC
Client
SIP Client
SIP
RTP/SRTP
ESL(TCP)
DTLS SRTP
PC
,
D
TLS
SRTP
Web Server HTTPS

18. Controller application
The controller application for the PoC is written in nodejs.Any other
technology can be used.
There are many libraries implementing Janus and FS clients.
e.g. https://janus.conf.meetecho.com/docs/resources#stacks
Connects to FS as a verto client (WebSocket) and to the ESL interface (TCP).
Connects to Janus HTTP API (or over WebSocket).

19. Expected RTP flow 1/2
FreeSWITCH Janus
SIP Client
A
SIP Client
B
WebRTC
Client
C
WebRTC
Client
D
A
C
D
C
D
DA
A
A
CD
C

20. Expected RTP flow 2/2
FreeSWITCH Janus
SIP Client
A
SIP Client
B
WebRTC
Client
C
WebRTC
Client
D
A
B
C
D
C
D
D
C
A
A
A
CD
AB
AB
AB
BCD
ACD

21. FS becomes a Publisher
FreeSWITCH Janus
SIP Client
A
Controller
1. INVITE SDPa
2. 200 SDPfsa
4. verto.invite
SDPfsb
5.create/attach/offer SDPfsb
6. answer SDPja
7.verto.answer SDPja
8. ICE + DTLS3. RTP
9. DTLS SRTP

22. FS becomes a Publisher
Note: “sendonly” and
“trickle”: false, otherwise
Janus will keep waiting for
ICE candidates from FS and
never start the DTLS
negotiation.

23. FS becomes a Subscriber
FreeSWITCH Janus
Controller
6. verto.media
SDPf
3. attach/join
4. attached
SDPjc
1. C
8. ICE + DTLS
9. DTLS SRTP
WebRTC
Client
C
5. verto.invite SDPjc
2. publishers list
7. start room
SDPf

24. FS becomes a subscriber
“trickle”: false, otherwise
Janus will keep waiting for
ICE candidates from FS.

25. Prestidigitation with the streams
- In an SFU model streams are separate
- An MCU mixes the streams
- WebRTC clients receive already streams from Janus directly
FS needs to eliminate the streams from WebRTC clients for the feeds towards
other WebRTC clients
- ‘relate’ function in mod_conference. “nospeak” option.

26. Testing it

27. Demo
Code used for the PoC:
https://github.com/giavac/janus_freeswitch_integration

28. Can this be
achieved in a
different way?
An MCU plugin could be developed for Janus.
An SFU module could be developed for FS.
Likely these solutions would be more cohesive
and efficient, but would require a larger
development effort.
To be evaluated case by case.
Performance assessment was out of scope in
this PoC.
Browsers support/compatibility also to be
verified.

29. Scalability
Performance was not in the scope of this PoC. However…
Another benefit of this modularity is that Janus and FS can scale
independently.
Each one with its own best practice (farming/clustering for FS, cascading for
Janus).

30. Not considered in this PoC
Many things!
- Creating and configuring rooms in Janus
- Clustering of FS and Janus instances (scaling)
- Dynamic dialplan in FS
- ...

31. Integrating Janus and FS
can bring the best of two
worlds together.
Think of this as an option
when building a
conferencing system in an
heterogeneous environment.

32. References
FS verto documentation: http://evoluxbr.github.io/verto-docs/
Janus VideoRoom documentation: https://janus.conf.meetecho.com/docs/videoroom
RFC 7667, https://tools.ietf.org/html/rfc7667 - RTP Topologies
“FreeSWITCH 1.8”, A. Minessale, G. Maruzzelli, Packt

33. Thanks!
Get in touch:
@giavac
https://www.linkedin.com/in/gia
comovacca/