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!

ICE

SIP outbound

GRUU

TLS

GIN - pbx registrations

SIP identity

RTP multiplexing

SPLICES

SIP3261

SIMPLE presence

WebRTCThe browser takes over. Beware, old phone!

WebRTC

The new kidon 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

applications

• Built into the web browser

• Security-enabled from start (hopefully)

• We’ll still have NAT and firewall issues

• Will it be standardized enough

• Will we need SBCs to handle the connections?

+ -

WebRTC

ICE

Taking us out of the NAT darkness.

ICE

Ice: Show me yours, and I’ll show you mine.

SIP

NATted network

NATted network

Alice

Bob

Cecilia

SIP

Media relay

•All UAs find all their addresses, using STUN

•May allocate an address using TURN

•Sends all addresses as candidates in SDP

•Receipient tries to contact addresses and select best media path

•Supports both IPv4 and IPv6

•IPv6 UAs allocate IPv4 Turn address

ICE

Turn

ICE• Finds the best media path

between two nodes

• Supports IPv4 and IPv6 deployments

• Binds SIP+SDP to actual media

• Takes time at call setup

• Hard for b2bua’s to support

• Complex for developers

+ -

ICE

SIP OutboundStay connected. And reconnect if it fails.

SIP

SIPSIP

NATted network

Location server/Registrar

Ingres proxys

OUTBOUND

RFC 5626

Client initiated connections

SIP

SIPSIP

NATted network

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

OUTBOUND

SIP outbound• Makes TLS easier

• Better definition for NAT traversal support

• Identifies devices in a unique way

• Makes TCP/TLS failover much, much quicker

• Adds number of connections

• Not implemented in many devices

+ -

OUTBOUND

Globally Routable device addresses

SIP

NATted network

Alice

Bob

Bob

SIP

SIP

Example.com

astritech.com

The AOR for Alice and Bobbelongs to their proxy. Bob has one

AOR for multiple UAs.

The GRUU points to a device. It is allocatedat registration and belongs to the domain, thus

can be used globally!

GRUU

Builds on SIP outboundUUID URN’s.

Device URIs• Makes transfers and

other SIP in-dialog functions work across domains

• A Contact without IPv4/IPv6 dependencies

• Opens up for multi-device calls (SPLICES)

• Complex RFC

• Adds a bit of complexity to the UA

+ -

GRUU

PBX trunk registrationOne REGISTER for multiple phone numbers

GIN

SIP PSTNPBXSIP Trunk

• 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 - PBX REGISTER

• Supports current usage by PBX vendors and SIP trunk providers

• Standardizes something that was no standard

• Cleans up

• Adds complexity in registrar and client

+ -

GIN

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.