Janus/SIP @ OpenSIPS 2019

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Put some Webin your RTC SIP Architecture with Janus Lorenzo Miniero @elminiero OpenSIPS Summit April 30th 2019,

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“By The PowerOf VoIP!” Lorenzo Miniero • Ph.D @ UniNA • Chairman @ Meetecho • Barber shop avoider Contacts and info • lorenzo@meetecho.com • https://twitter.com/elminiero • https://www.slideshare.net/LorenzoMiniero

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A few wordson Meetecho • Co-founded in 2009 as an academic spin-off • University research efforts brought to the market • Completely independent from the University • Focus on real-time multimedia applications • Strong perspective on standardization and open source • Several activities • Consulting services • Commercial support and Janus licenses • Streaming of live events (IETF, ACM, etc.) • Proudly brewed in sunny Napoli(*), Italy

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((*) You may haveheard of it)

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Why SIP andWebRTC? • A lot of reasons why it makes sense to use WebRTC and SIP together • WebRTC stacks are avalailable everywhere, so making clients is easier now • Almost all of you have a SIP infrastructure already, and want to reuse it • SIP and WebRTC are similar enough that gatewaying isn’t impossible • PSTN integration is a common scenario • “Hey granma, I’m calling from my browser!” • Contact centers can really beneﬁt from that • It’s much easier for customers to get in touch (button on the website) • It’s much easier to remotize agents (they just need a browser) • Re-using existing infrastructure saves a lot of money • SIP-based conferencing can be enhanced too • e.g., audio conferencing involving WebRTC and PSTN users • ... maybe with video only for WebRTC users? (hybrid approach) • ... or, why not, video for everyone with the help of an MCU

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Sometimes it’s notthat easy, though... Try adding WebRTC support to THAT!

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How similar areSIP and WebRTC?

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How similar areSIP and WebRTC?

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The WebRTC protocolsuite: scary! • 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)

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Talking to aserver instead

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... maybe SIPand other stuff as well!

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Enter Janus! Janus General purpose,open source WebRTC server • https://github.com/meetecho/janus-gateway • Demos and documentation: https://janus.conf.meetecho.com • Community: https://groups.google.com/forum/#!forum/meetecho-janus

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Modular architecture • Thecore only implements the WebRTC stack • JSEP/SDP, ICE, DTLS-SRTP, Data Channels, Simulcast, VP9-SVC, ... • Plugins expose Janus API over different “transports” • Currently HTTP / WebSockets / RabbitMQ / Unix Sockets / MQTT / Nanomsg • “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 can be combined on client side as “bricks” • Video SFU, Audio MCU, SIP gatewaying, broadcasting, etc.

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Extensible Architecture andAPI

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Extensible Architecture andAPI

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Janus + 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 • Built-in recording (separate media legs) • 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)

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The SIP pluginin a nutshell

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Is SIP inthe browser always a good idea? • Most WebRTC gateways require 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 SIP infrastructure can be completely clueless about WebRTC 2 Handling a whole SIP stack on the client side can be troublesome 3 Using a simpler JSON API makes it easier for web developers 4 Wrapping SIP helps protecting from “well known” attacks (e.g., vulnerabilities) 5 Much easier to make “assumptions” on the trafﬁc you’ll handle 6 Masking SIP also helps hiding the topology of the SIP infrastructure

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Using the SIPplugin in Janus instead

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Using the SIPplugin in Janus instead

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What if youDO want SIP in the browser? • You may want to have more control on SIP messaging • e.g., to re-use stacks like JsSIP or SIP.js instead of JSON, or other reasons • The SIP plugin doesn’t allow for that • Complexity hidden from users, on purpose • Only partial control (e.g., custom headers, INFO DTMF, negotiating security) • NoSIP plugin to the rescue! • Different Janus plugin that doesn’t care about signalling, only SDP • You pass it a WebRTC SDP, it gives back a plain SDP • You pass it a plain SDP, it gives back a WebRTC SDP • Plugin bridges the media between WebRTC user and legacy client • No "register", "call", etc., that’s up to you (e.g., SIP or others)

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Using the NoSIPplugin in practice

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Using the NoSIPplugin in practice

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How scalable isall this? • Did a talk on Janus scalability just last year, at CommCon • https://2018.commcon.xyz/speakers/lorenzo-miniero/ • https://www.youtube.com/watch?v=zxRwELmyWU0&t=264s • Recently did many measurements on Janus in general as well • https://janus.conf.meetecho.com/citeus • So how easy is to scale the SIP plugin? Quite easy, actually! 1 Load increase is pretty much linear 2 Caller and callee are independent of each other (tied by SIP infrastructure) 3 Very easy to spawn new instances on a per-need basis 4 Different strategies possible depending on how load balancing needs to be done

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Different scalability andHA strategies

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Did I mentionJanus works great with HOMER? https://www.opensips.org/events/Summit-2018Amsterdam/

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Next steps • Whilebasically complete, the SIP plugin could use some enhancements • No support for blind/attended transfers yet, for instance • Additional security, e.g., via authenticated headers • A SIP trunking mode might also help • Maybe multiple active calls at the same time on the same PeerConnection? • Lawful Interception is a commonly asked feature as well Your application here! • Janus SIP plugin already used in production by many • Play with it and see if it can help you too!

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Next steps • Whilebasically complete, the SIP plugin could use some enhancements • No support for blind/attended transfers yet, for instance • Additional security, e.g., via authenticated headers • A SIP trunking mode might also help • Maybe multiple active calls at the same time on the same PeerConnection? • Lawful Interception is a commonly asked feature as well Your application here! • Janus SIP plugin already used in production by many • Play with it and see if it can help you too!

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Thanks! Questions? Comments? Getin touch! • https://twitter.com/elminiero • https://twitter.com/meetecho • http://www.meetecho.com

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See you soonin Napoli! September 23-25, 2019, Napoli — https://januscon.it

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See you soonin Napoli! September 23-25, 2019, Napoli — https://januscon.it

Janus/SIP @ OpenSIPS 2019

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