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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!
                     SIMPLE presence

                                        GIN - pbx
         ICE
                                       registrations


SIP 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
             ICE




Taking 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 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



+                                -
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!
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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Rethinking the PBX

  • 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.