Rethink the PBX

Rethinking the PBX. The SIP building blocks for changing how we communicate.@oej & @saghul

SIP has changed.• SIP is no longer the same as 10 years ago• RFC 3261 is no longer the single reference• There’s a new kid in town. Meet him!

Meet the new SIP!

Meet the new SIP! SIMPLE presence GIN - pbx ICE registrationsSIP outbound SIP SIP identity 3261 GRUU RTP multiplexing TLS SPLICES

WebRTC WebRTC The browser takes over. Beware, old phone!

WebRTC The new kid on the block• Cooperation between the W3C and IETF• Bidirectional media between browsers• Audio, video, text• The platform for new services• SIP in the browser

WebRTC The vision• An open service where we can communicate freely with each other from any device and any network• First wave propably just between users of the same web service• Many of us wants open federation - it requires a shared address space and protocol

WebRTC Dependencies• The architecture is still discussed• Will propably depend on ICE, which means dependencies on TURN/STUN as well• Do we need full PSTN interoperability?• What about security?

WebRTC WebRTC• Platform for new cool • We’ll still have NAT applications and ﬁrewall issues• Built into the web • Will it be standardized browser enough• Security-enabled from • Will we need SBCs to start (hopefully) handle the connections?+ -

ICE ICETaking us out of the NAT darkness.

ICE Ice: Show me yours, and I’ll show you mine. NATted network • All UAs ﬁnd all their SIP addresses, using STUN SIP Alice • May allocate an address using TURN • Sends all addresses as candidates in SDP • Receipient tries to contact addresses and select best media path Turn • Supports both IPv4 and IPv6 Bob Media relay • IPv6 UAs allocate IPv4 NATted network Turn address Cecilia

ICE ICE • Finds the best media path • Takes time at call between two nodes setup • Supports IPv4 and IPv6 • Hard for b2bua’s to deployments support • Binds SIP+SDP to actual • Complex for media developers + -

OUTBOUND SIP Outbound Stay connected. And reconnect if it fails. NATted network SIP SIP SIP Location server/Registrar Ingres proxys RFC 5626

OUTBOUND NATted network SIP SIP Client initiated connections SIP Location server/Registrar Ingres proxys• The client is responsible for keeping the connection open• Clients has a UUID, device identifier that stays the same - ALWAYS!• The SIP proxy sees that one device has multiple registrations and use only one at a time• The Registrar or Ingres proxy assigns a flow ID that is unique for each flow• A dialog stays on one connection until it fails

SIP outboundOUTBOUND• Makes TLS easier • Adds number of connections• Better deﬁnition for NAT traversal support • Not implemented in many devices• Identiﬁes devices in a unique way• Makes TCP/TLS failover much, much quicker+ -

Globally Routable device addressesGRUU Example.com SIP SIP Alice The AOR for Alice and Bob belongs to their proxy. Bob has one Builds on SIP outbound AOR for multiple UAs. UUID URN’s. SIP astritech.com BobNATted network Bob

Globally Routable device addressesGRUU Example.com SIP SIP Alice The AOR for Alice and Bob belongs to their proxy. Bob has one Builds on SIP outbound AOR for multiple UAs. UUID URN’s. SIP astritech.com Bob The GRUU points to a device. It is allocatedNATted network at registration and belongs to the domain, thus Bob can be used globally!

GRUU Device URIs• Makes transfers and • Complex RFC other SIP in-dialog functions work across • Adds a bit of domains complexity to the UA• A Contact without IPv4/ IPv6 dependencies• Opens up for multi- device calls (SPLICES)+ -

GIN PBX trunk registration One REGISTER for multiple phone numbers PBX SIP Trunk SIP PSTN• Created by The SIP Forum for SIPconnect 2.0• RFC 6140• Only for E.164 phone numbers• 200 OK to register includes all the phone numbers• Location server adds one AOR contact binding per number• Use GRUUs, which depends on SIP outbound

GIN GIN - PBX REGISTER• Supports current usage • Adds complexity in by PBX vendors and SIP registrar and client trunk providers• Standardizes something that was no standard• Cleans up+ -

SPLICES• IETF working group• Adding remote devices to an existing SIP session• Add your TV with webcam to a call on your smartphone

Rethinking Realtime communication Note: This is not science ﬁction.

Rethink the client• The client is not a ”phone”• People are not phone numbers• The client is in the browser or a separate app• It’s in all your devices - smartphone, laptop, pad, desktop• Possibly in your car, set-top box, TV

Rethink the server• It’s not one application, one PBX• It’s a group of servers producing SIP services• Your domain is your cloud.

Rethink the user• The user does not want to run SIP• The user wants to communicate with another user or entity• Wants to manage the session - move between multiple devices during a session• From your kid to your grandpa

Meet the future.• Start a call with your wife in your car• Walk into the house, add the TV to the call• Invite your grandpa to the call• Show video from vacation to all participants in the call• Wife stops the video feed from her device

Finding you and setting up a session.• I ﬁnd you in my address book, buddy list or on a web page• I start a session and get a menu of our common media types - right now• If the session goes over PSTN, I notice it by the limited audio quality• I don’t start with selecting device and media type, I start with selecting YOU and checking if your are available.• Your phone number is not relevant any more. It’s a gateway to the past.

Rethinking yourself.• Your next PBX is not a PBX.• Open up for new services• Open up for modern communication• Open up for personal communication you can trust.

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Rethink the PBX

by Olle E Johansson

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