Slides from the talk given at 4KConference in Buenos Aires (november 2011) together with Olle E Johansson (http://slideshare.net/oej).

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

2. 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!

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

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

5. 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

6. 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

7. 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?

8. 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?
+ -

9. ICE
ICE
Taking us out of the NAT darkness.

10. 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

11. 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
+ -

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

13. 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

14. SIP outbound
OUTBOUND
• 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
+ -

15. Globally Routable device addresses
GRUU
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 allocated
NATted network at registration and belongs to the domain, thus
Bob can be used globally!

16. 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)
+ -

17. 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

18. 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
+ -

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

20. Rethinking Realtime
communication
Note: This is not science fiction.

21. 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

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

23. 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

24. 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

25. 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.

26. 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.