Janus/Asterisk @ Astricon 2017

1. Facilitating WebRTC access to Asterisk-based services with Janus Lorenzo Miniero @elminiero

2. Who am I? Lorenzo Miniero • Ph.D @ UniNA • Chairman @ Meetecho • Open source believer • Viking wannabe (←) Contacts and info • lorenzo@meetecho.com • https://twitter.com/elminiero • https://www.slideshare.net/LorenzoMiniero

3. You may remember me from asterisk-opus

4. Before moving on, what’s Meetecho? • Co-founded in 2009 as an academic spin-off • University research efforts brought to the market • Recently got my Ph.D @ UniNA • Completely independent from the University • https://www.meetecho.com • Focus on real-time multimedia applications • Web conferencing only, at ﬁrst • Then widened the scope to multimedia in general • Strong perspective on standardization and open source • WebRTC rulez! • Proudly brewed in sunny Napoli(*), Italy

7. ((*) Napoli ruined by our ugly faces)

8. ((*) Much better...)

9. How do you pay your bills? • Meetecho Web Conferencing and Collaboration • Meetings, conference calls, webinars, etc. • Subscription, one-shot, ... • Streaming of live events • Internet Engineering Task Force (IETF) • ACM SIGCOMM • Consultancy services • Mostly (but not only) involving WebRTC and Janus • Installation and conﬁguration • Custom plugins for custom use cases • Wrapping/orchestration of Janus resources • Sponsored development on new features or improvements • ... • Commercial support and Janus licenses

13. WebRTC reference architecture: peer-to-peer

14. WebRTC reference architecture: peer-to-peer

15. The WebRTC protocol suite • Signalling (well, sort of) and Negotiation • Javascript Session Establishment Protocol (JSEP) • Session Description Protocol (SDP) adaptation • Connection Establishment and NAT Traversal • Session Traversal Utilities for NAT (STUN) • Traversal Using Relay NAT (TURN) • Interactive Connectivity Establishment (ICE) • Media Transport and Control • Real-time Transport (and Control) Protocol (RTP/RTCP) • Secure Extensions to RTP (SRTP) • Datagram Transport Layer Security (DTLS) • Multimedia codecs • Opus audio codec (MTI, Mandatory-to-implement) • VP8 and H.264 video codecs (MTI, Mandatory-to-implement) • Generic Data • WebRTC Data Channels (SCTP)

16. The WebRTC protocol suite • Signalling (well, sort of) and Negotiation • Javascript Session Establishment Protocol (JSEP) • Session Description Protocol (SDP) adaptation • Connection Establishment and NAT Traversal • Session Traversal Utilities for NAT (STUN) • Traversal Using Relay NAT (TURN) • Interactive Connectivity Establishment (ICE) • Media Transport and Control • Real-time Transport (and Control) Protocol (RTP/RTCP) • Secure Extensions to RTP (SRTP) • Datagram Transport Layer Security (DTLS) • Multimedia codecs • Opus audio codec (MTI, Mandatory-to-implement) • VP8 and H.264 video codecs (MTI, Mandatory-to-implement) • Generic Data • WebRTC Data Channels (SCTP)

17. Involving a server as a peer (and applications)

22. Involving different technologies as well

25. Janus: a general purpose WebRTC server “In ancient Roman religion and myth, Janus [..] is the god of beginnings and transitions, and thereby of gates, doors, passages, endings and time. He is usually depicted as having two faces, since he looks to the future and to the past.” — http://en.wikipedia.org/wiki/Janus

26. Janus: a general purpose WebRTC server • A door between the communications past and future • Legacy technologies (the “past”) • WebRTC (the “future”) Janus General purpose, open source WebRTC server and gateway • https://github.com/meetecho/janus-gateway • Demos and documentation: https://janus.conf.meetecho.com • Community: https://groups.google.com/forum/#!forum/meetecho-janus

27. Modular architecture • The core only implements the WebRTC stack • JSEP/SDP, ICE, DTLS-SRTP, Data Channels, ... • Recording/simulcasting/monitoring/etc. natively done here as well • Plugins expose Janus API over different transports • Currently HTTP / WebSockets / RabbitMQ / Unix Sockets / MQTT • “Application” logic implemented in plugins too • Users attach to plugins via the Janus core • The core handles the WebRTC stuff • Plugins route/manipulate the media/data • Plugins used as “bricks” to compose application • Streaming + VideoRoom = Social TV • VideoRoom + AudioBridge or SIP + TextRoom = Webinar • ...

31. Extensible Architecture and API

32. Extensible Architecture and API

33. What can Janus do to help with SIP? • Since day one, SIP support has been available as a Janus plugin • Demo: https://janus.conf.meetecho.com/siptest • Basically a WebRTC-to/from-SIP gateway • WebRTC on one side, SIP(S)/(S)RTP on the other end • Janus SIP plugin acts as a SIP endpoint • SIP stack implemented with Soﬁa-SIP • WebRTC users only see the Janus API (JSON) • No transcoding, media is only relayed • Simpliﬁes life for web developers • No need to worry about a SIP stack (only SIP URIs) • Basic methods/events to handle call (call, answer, hangup) • Allows headers injection (REGISTER, INVITE)

37. Yeah, yeah, but why should I care? • Asterisk supports WebRTC out of the box • Signalling done via SIP over WebSockets • ICE support via pjnath • DTLS-SRTP supported too • Recently, added support for rtcp-mux too • Sometimes helpful when it doesn’t have to, though • Asterisk may already be responsible for a ton of other things • What if we can at least remove the burden of WebRTC management? • Lighten Asterisk, and WebRTC/services scaling can be done separately • ... but having Janus in front of Asterisk can be useful for other reasons too • More in the next few slides!

40. [PLEASE DON’T HIT ME] Asterisk 11/12! • We already mentioned the “infamous” rtcp-mux constraint • Normally RTP and RTCP are transported in different ports (e.g., RTCP = RTP + 1) • rtcp-mux = muxing RTP and RTCP over the same transport • Made mandatory since Chrome 57 • http://www.juandebravo.com/2017/02/15/rtcp-mux-in-webrtc • Huge impact on SIP endpoints • Most don’t support the new stuff WebRTC is adding, including rtcp-mux • Deﬁnitely impacted Asterisk as well, when the news came out • https://nimblea.pe/monkey-business/2017/01/19/webrtc-asterisk-and-chrome-57/

44. [PLEASE DON’T HIT ME] Asterisk 11/12! • Asterisk developers promptly fixed this! • http://blogs.asterisk.org/2017/04/26/rtcp-mux-webrtc/ • ... but ONLY for Asterisk >= 13.15.0 and >= 14.4.0 • Older versions don’t have this fix • Now, you REALLY shouldn’t use older versions of Asterisk... • ... but if for some reason you HAVE TO, Janus can fill that gap!

45. You may not want SIP in the browser • By default, Asterisk requires you to do SIP in the browser • SIP over WebSockets (e.g., via JsSIP) • We explained how the Janus SIP plugin does SIP internally, instead • A SIP stack is created for each new user • The client exchanges messages via a JSON-based API • There are indeed reasons why you may NOT want SIP in the browser... 1 Handling a whole SIP stack on the client side can be a nightmare 2 Using a simpler JSON API makes it easier for web developers 3 Wrapping SIP helps protecting from “well known” attacks (e.g., vulnerabilities) 4 Much easier to make “assumptions” on the trafﬁc you’ll handle 5 Masking SIP also helps hiding the topology of the SIP infrastructure

53. You may not want SIP in the browser

54. You may not want SIP in the browser

55. Making debugging/troubleshooting a bit less “hell” • Establishing WebRTC PeerConnections might fail for different reasons • ICE failures, DTLS handshake errors, MTU issues, etc. • Can be even more problematic when you use different servers for the same app • Different servers may handle different scenarios in a different way • What if one works and the other doesn’t under the same conditions? • Each WebRTC server has its libraries and debugging approaches • This may be one more reason to deploy Janus as a frontend for WebRTC • If SIP and, e.g., RTSP fail because of WebRTC, you know where to look at • Several tools to use for debugging and troubleshooting • Admin API (pull based) • Event Handlers (async and modular, integrates with Homer/HEPIC) • text2pcap support for dumping unencrypted trafﬁc to .pcap (NEW!)

59. HEPIC is Epic! https://fosdem.org/2017/schedule/event/homer/

60. “One API to rule them all!” • As explained before, it’s very common to mix plugins to build complex apps • e.g., VideoRoom + SIP for hybrid videoconferencing with PSTN support • Some real-world examples in just a few slides, stay tuned! • If you’re using Janus for other features, why not extend it to SIP as well? • Using a single and consistent API will make your web developer happier • Besides, see previous point on debugging...

61. What is Janus used for today, and by whom? • We use it ourselves for many things (obviously) • Web conferencing, Webinars, SIP gateway, ... • Many folks/companies also using it in creative ways! • E-learning • Coworking • Contact centers • TV broadcasting and Social TV • Videogames streaming and coplaying • Surveillance systems • E-health • Home automation & Internet of Things • Mobile devices, Raspberry Pis, wearables, drones, etc. • Several third-party tools available already • https://janus.conf.meetecho.com/docs/resources

64. Streaming IETF meetings with Janus https://www.vuc.me/2017/vuc640-ietf-remote-participation-with-lorenzo-miniero/

69. Voxbone (click-to-call) https://click2vox.com/

70. Lenovo’s AirClass (e-learning) https://www.airclass.com

71. Convoso (contact centers) https://www.convoso.com

72. Veeting rooms (web conferencing) https://www.veeting.com

73. Evercast (collaboration & media production) https://www.evercast.us/

74. What to do next? • Improve the WebRTC implementation • Work on effective renegotiation (ICE restarts almost ready) • Implement multistream (Uniﬁed Plan) • SIP-wise, we have some alternatives almost ready • SIPre (same as existing plugin, but based on libRE) • NoSIP (media-only gateway, signalling left to the application) • https://www.slideshare.net/LorenzoMiniero/janussip-opensips-2017 • WIP: a Janus Lua plugin! • Media routing and manipulation done at the C level • All the logic/signalling in Lua instead Your application here! • What if this was just what you needed? Play with it! • ... and why not, write your own applications/wrappers/plugins!

78. Thanks! Questions? Comments? Follow us! • https://twitter.com/elminiero • https://twitter.com/meetecho

Janus/Asterisk @ Astricon 2017

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Janus/Asterisk @ Astricon 2017