1. The VoIP Net: From POTS to Quality Unified Communications Globally
Prepared for: Ingate Systems 3 Day Seminar
Unified Communications: SIP Trunking, Video, Collaboration and More
ITEXPO Conference, Austin, September 2011
By: Karl Erik Ståhl
President Intertex Data AB
CEO and Chairman Ingate Systems AB
Also see Live Demo Presentation from ITEXPO SIP Trunking Summit Miami, February 2011!
© 2011 Intertex Data AB

2. Intertex & Ingate Same parent company
Intertex: SMB, SOHO and home SIP Firewalls and E-SBCs
For volume deployment
Ingate: Enterprise and SMB SIP Firewalls and E-SBCs
SIParators® for enterprises and projects
Cooperation in management and development
Co-developed SIP code
Ingate represents Intertex in the US

3. From POTS to Global Quality UC
POTS and mobile have global reach, but Low Fi voice only…
UC is rich communication, but today only in islands (the Enterprise UC LAN, Skype, Google Talk and the others)
UC should be global, with quality and with SIP-addresses as well as phone numbers!
Will it happen?
How?
When? 3

4. Yes, Telcos are Concerned About their Core Business!
Are Telcos just becoming bandwidth providers?
IP has just been used to replicate POTS Telephony
Where is the global Live IP Communication: Multimedia or UC?
The “Beyond POTS” islands are taking over:
at the Enterprise UC LAN
by Skype, Google Talk and the others
We can Go Beyond POTS and Beyond Skype Now!
Why not better and beyond?
Telcos can bring it together and offer better! 4

5. Connect Us Together, add Functionality and Quality!
Bringing the Islands together is a Telco core business!
So is bringing Functionality, Quality and Reliability!
Give us a SIP addresses (same as ) to each phone number!
and not with eight dect phones either!
We cannot get stuck here
Phones can be
More Than Smart? – What about better than AM-radio… 5

6. Shouldn’t We be Far Beyond PSTN and POTS?
RJ11
POTS and PSTN have been there for 100 years
Black Phone
3.5 kHz isn’t HiFi, but MOS is 5!
RJ45
LAN Intranet Internet
Now we have a new global network: The IP Networks
And we have a new standard: SIP
WiFi Mobile
Soft Client
IP Phone
Presence
Messaging
Video
Voice

7. But IP Used to Replace Bits of the PSTN, keeping POTS 
Gateway
Toll
Bypass
Europe US
VPN
Tunnel
IP PBX
IP PBX
Gateway
PBX
Are we stuck
with old POTS
telephony
over new wires?
PSTN
Gateway
Voice over
Broadband
Soft
Switch
Very seldom VoIP connectivity between the VoIP IP clouds!
Most broadband VoIP providers still run calls between each other over the PSTN!

8. SIP wasn’t Meant for Islands or Voice Only!
DNS
partco.com
To receive SIP calls globally:
- A SIP server (Proxy Registrar)
- SIP server domain published in DNS
Proxy Registrar
for partco.com
RING!
RING!
RING!
Internet
To initiate SIP calls:
- A proxy capable of routing (=DNS lookup!) - Add ENUM to use E.164 numbers
Outbound proxy for smartco.com
CALL
CalleeProxy
CallerProxy
Callee
Caller
The SIP tapeziod
Magic? – It’s just the SIP standard…

9. The SIP Standard: Global and More Than Voice!
over the Internet, but then:
not always sufficient quality
difficult to bill by usage (Telcos’ core business…)
and the NAT/Firewall traversal issue must be resolved
Telcos have feared another Skype…
But Telcos don’t like another Skype. Need to offer more to bill…

10. Instead IP and SIP has been Used to Replace Pieces of the PSTN Maintaining Old Structures  POTSoIP
We see a telephony overlay structure of Soft Switches and SBCs (or IMS) on top of IP that is ill suited for anything beyond old time voice.
And Carriers Peer their Networks PSTN Style…
It is even destructive for the 160 years old Fax service*
* Mike Coffee, CEO of Commetrex: Work in progress by SIP Forum’s FoIP Task Group i3 Forum.
T.38 works fine in one hop!

11. Telephony Peering: A Show Stopper for Global UC!
Carrier Peering points are mostly TDM type  POTS only
There are VoIP Peering – But still
only for Voice Minutes  POTSoIP
What is a UC Minute? – A dead end!
A price list not even possible…
Voice, HD Voice, Codec, Video resolution, Video codec, IM, Presence message?
Quality level used - From Internet to critical telepresence?
And for new services?
Simple solution:
The SP must only charge its own customers – A great simplification!
Any charging between carriers should only be IP based – Not application specific!
The IMS model will not work!
Even if IMS gets it technically working– Global connectivity will fail due to lack of multimedia price lists between SPs.
There is STILL NO IMS peering after all these years (only voice peering used). 11

12. But We Can Do It and We Are Close!
Deutsche Telecom
Internet
AT&T
Qwest
TeliaSonera Internet
MPLS
MPLS
QoS IP Network
QoS IP Network
MPLS
ENUM
CDR
CDR
SIParator
IX78
Let’s see:
- What is missing?
- How can we do it?

13. Missing 1: The Global Quality IP Network
We have the Internet…
But you cannot get priority – Not always good enough, e.g. for Telepresence
We have the Telco’s better VoIP networks
But they are VoIP islands or (in best case) connected via SBCs for voice peering only
We need a higher quality IP network IP Peered globally, the “IQ-Net”
Just like for the Internet – Settlement free peering (or simple data volume charges)
For real time traffic only, and with quality bits (DSCP/TOS) honored
Routed to the Internet will give interoperability – Same network. Here we simply have access to higher quality. (TeliaSonera’s VoIP network has used that model since long)
AT&T
TeliaSonera Internet
Deutsche Telecom
Internet
Qwest
Soft Switch
IQ-Net = Quality bits (DSCP/TOS) honored 13

14. Multiple QoS Separated WAN Pipes are Common (Telia Network)
Internet
The Multimedia LAN
All services must be available to multimedia terminals! – Over controlled high QoS pipes as well as the Internet.
IMS
VoIP
TR-069
IP-TV
VoD
Internet
Application Innovation Requires it!
WiFi
VLANs or ADSL Virtual Circuits
The Multimedia LAN
Telepresence
IP-
PBX
     
But only the Internet is IP-peered globally.  IP-peer the VoIP networks = IQ-Net!
PDA

15. Missing 2: Billing by Usage
Usage of the better IQ-Net must be billed separately from the Internet
If not charged separately, it would be used for everything and we are back at all usage being at the same quality level.
Service Providers should only charge their customers
No UC minute charge between carriers. Cannot be defined. – Great simplification!
Settlement free IP peering has worked for the Internet. Any exchange between carriers shall only be based on IP level usage (data transported at a certain quality level).
Measure usage at certain quality levels = Fair pricing model
Usage, for each call, voice, video and data transferred and at the quality level used can be put in CDRs and then billed as usual to the customer.
Can be done by clever E-SBCs*
* Some operators already require the E-SBC to measure the quality – MOS value – of each call.
The E-SBC can also classify the traffic and assure the correct quality pipe is used.
Such E-SBCs are Telco deployed, Owned and Managed. TR-069 is a secure, highly scalable management protocol allowing such usage. 15

16. 3-Party Video Conference with CDRs including Call Quality Metrics

17. Now also with Video Call Metrics and Pipe Used!
Billing – A Key Thing
Now also with Video Call Metrics and Pipe Used!
CDRs with Call Quality Metrics – View from iEMS (our TR-69 management system)

18. Missing 3: Delivery to the LAN Users
NAT/Firewall Traversal needs to be handled!
Has hampered deployment and usage of SIP for 10 years. (Applies to all such protocols: The need to connect to users on private LANs and to have media flows on separate ports is not trivial.)
Skype’s clever traversal was a main cause of its immediate success
But workaround methods (STUN, TURN, ICE, Far End NAT Traversal) have their limits. Reliability, battery draining of mobiles, no QoS (for Internet, not for IQ-Net)
Handled simply: VoIP services often terminates in analog voice ports (logically outside the Firewall)  POTS only
SIP Trunking of PBX’s already often use E-SBCs for NAT traversal (and more)
SERVER FW
HTTP, SMTP, etc.
NAT/FW designed for this, but… FW
PERSON
SIP (and H.323…) connects Person-to-Person
Locate the person - Set up a session - Open real time media streams 18

19. A CPE is Most Often Required Anyway
A SIP Proxy Based E-SBC, also routing media, can do it all
Such are also used for security, SIP normalization,
QoS, failover and features
And while routing SIP, it can add the
functions found in Soft Switches
Centralized SBCs doing the NAT traversal, require an individual pipe (e.g. MPLS or other “VPN”) to the each LAN (security concern also)
…often ending up in an expensive Telco deployed CPE anyway
Handling many users, makes it a critical point of failure – Duplication and failover capability by an E-SBC is often required 19

20. Not Missing: PSTN/POTS connectivity
We have the SIP Trunks
SIP Trunking of PBXs is really about connecting to PSTN/POTS over IP
Leave today’s SIP Trunks – Add the UC communication to the same endpoints. The clever E-SBC can do it.
For the future – if the operator’s SIP Trunk routes to the IQ-Net – the SIP Trunk can carry multimedia.
We have the hosted VoIP services
Leave them just for PSTN/POTS usage. Clever E-SBC can add the UC communication to the same endpoints.
For the future – if the operator’s Soft Switch routes to the IQ-Net – it can carry multimedia. 20

21. Reusing the SIP Trunking E-SBC
Telco owned E-SBCs are already used for (voice) SIP Trunking
Full operator control
Service provider’s demarcation point
Enables the SIP Trunking: for NAT/Firewall traversal, PBX interoperability and Security - But Video is not very different from Voice at this level…
Reuse the same E-SBC for Video Calling and other UC!
In the Ingate and Intertex E-SBCs, it is all there:
Classify outgoing calls (as Video, HD voice or plain voice)
Assure right quality pipe and/or quality marking is used
Route the call directly to the other party
Use ENUM (public or private) for E.164 number to SIP address resolution
Only settlement free IP peering between operators required
Can fallback to best effort IP peering (Internet) in operator network
Produce and deliver CDRs for each call
Report Minutes and Data used
Include video and voice quality metrics (including MOS scores)
Deliver via Radius, Syslog, Management system (TR-069 informs) or method by choice 21

22. A SIP Trunking E-SBC May Look Like a Specific “Gateway”!
SIP Trunking Provider
SIP System GW
PSTN
IP-PBX
SIParator®
Data & VoIP LAN

23. …but can be an Enabler for SIP Services and Users Everywhere!
SIP Trunking Provider
SIP System GW
PSTN
UC Voice Mail
Remote Users
Ingate/Intertex E-SBCs enable SIP based Live UC Across the Borders!
(SIP does not traverse ordinary NAT/Firewalls.)
IP-PBX
SIParator®
Data & VoIP LAN

24. …and can Today Realize the Global UC Network!
Deutsche Telecom
Internet
AT&T
Qwest
TeliaSonera Internet
MPLS
MPLS
QoS IP Network
QoS IP Network
MPLS
ENUM
CDR
CDR
SIParator
IX78

25. For the Telcos To Do
Provide high quality IP pipes for Video and HD Voice (e.g. MPLS)
If on separate layer 2 networks for quality, still make them routable to the Internet (for fallback to best effort peered carriers).
Enter users in ENUM (public or private)
E.164 numbers to SIP address resolution
Settlement Free Peering between Carriers for high QoS IP networks
Just like for the Internet - Now also for high quality IP network (e.g. by MPLS)
Share ENUM (number/SIP addresses between the Carriers)
Deploy same CPEs (E-SBCs) as for SIP Trunking
Can also be general SIP enablers (at least Intertex’ and Ingate’s) for offering all types of SIP based services
Process the CDRs from the E-SBC as usual for Billing
Via Intertex’ TR-069 server (ACS) is a very good solution 25

26. And What Is Required to “UC Communicate” Globally?
Talk: SIP!
It is the standard for global live person-to-person communication.
Connect over the IQ-Net - being billed for that usage.
And, the E-SBC can seamlessly integrate*:
Connect over the Internet - probably just flat rate Internet billing.
Connect to the PSTN over the SIP Trunk, being billed as for POTS voice (or it may be a service on the IQ-Net)
ATA
We don’t need new standards – They already exist!
But standard deviating endpoints need to adapt or have gateways.
And minimum requirements for a certain application may of course be required, e.g. Codec G711 u-law for voice telephony, T.38 for fax, H.26?... for video etc.
* See: 26

27. Can the “Core” Soft Switch/SBC Participate?
Sure, but since IP peered quality network will be used for the transport (instead of voice specific telephony peering), the “Soft Switching” is not really used. But Soft Switches could route calls just like any other SIP Proxy.
Their role will be more of a hosted service, as the SIP Registrar, and for applying policies for incoming calls (as an alternative of having it in a local PBX or in the E-SBC).
And the E-SBCs will still be required for various reason, e.g. Security, Interoperability and (unless a separate level 2 pipe (”VPN”) is provided from a central SBC to each customer) also for NAT/Firewall traversal.

28. Is it coming? The OVCC Initiative (by Polycom)!
A network just for Video Calling or the start of the common global UC network?
Key points:
A global quality IP network
Service Providers only charge their own customers
SIP is the standard
SIP addresses ( -like) and E.164 numbers 28

29. E-SBCs & SIP Capable Firewalls
See us at ITEXPO Room 9C!
Intertex Data AB
Rissneleden 45
SE Sundbyberg
Sweden
Tel:
Ingate Systems Inc.
7 Farley Road
Hollis, NH 03049
United States
Ph: +1 (603)
Tel sv:

30. E-SBCs & SIP Capable Firewalls See us at ITEXPO Room 9C!
Some extra slides, with further details follows.
See us at ITEXPO Room 9C!
Intertex Data AB
Rissneleden 45
SE Sundbyberg
Sweden
Tel:
Ingate Systems Inc.
7 Farley Road
Hollis, NH 03049
United States
Ph: +1 (603)
Tel sv:

31. Going Beyond POTS Means Leaving the POTSoIP “Session Delivery Networks”
View it as an improved Internet (driving real time communication)
With IP level QoS honored
Not restricted to just flat billing: Also billing by Quality level and Usage
(Access can be billed separately from the Service )
With delivery of SIP Communication to the LAN (traversing the NAT/Firewall)
With SIP addresses = E.164 numbers (via ENUM)
Where a SIP service can be anywhere. The “PBX” or Soft Switch should be the SIP registrar.
Where today’s SIP Trunks will be the gateways into the PSTN
Clever E-SBCs can do it all
NAT/Firewall traversal – delivers SIP to the LAN
Can be the registrar (eases proper SIP addresses:
Can be the (multimedia) PBX
Can lookup ENUM and route
Can measure usage and generate CDRs
Can enable access, based on authenticated E-SBC
A great simplification of network architecture, giving large savings for the Telcos! 31

32. Quality is Really an Advantage Only the Telcos can Bring!
Bringing the Islands together is a Telco core business!
So is bringing Functionality, Quality and Reliability!
Some basics around IP QoS and why better Internet QoS cannot be for free:
A. On the Internet we have Transport layer (4) QoS. The endpoint smartness of TCP makes it all work, filling and sharing the pipe, and backing off for datagram type of packets (e.g. UDP thus RTP). This is mostly often good enough – even for voice. However, in the process of sharing a filled pipe, even non TCP packets (e.g. UDP/RTP) are lost (and filling the whole pipe with such packets, is a catastrophe).
B. IP Layer (3) QoS (DSCP/TOS bits honored) is available in almost any IP network – just ignored on the Internet – and gives absolute priority. You simply don’t lose any packets unless the whole pipe is filled with your quality level packets (and higher). This is needed for critical real time applications, especially low delay, packet loss sensitive applications; obviously telepresence and sometimes even voice.
C. Giving IP Layer (3) QoS to the common Internet for free will of course not help! As soon as the first file sharer will select the highest quality, all users have to do the same to get their share and we are back to A. again. Thus, better IP Layer QoS has to bear a price – has to be charged!
D. Prioritization and traffic shaping in boxes like ours helps in case A.. However, that only works for traffic that is known or classified by the box, which typically is not the case for SIP using workaround methods like STUN/TURN/ICE or Far End NAT Traversal, Skype, Google Talks or the others and will remain in an environment with the lowest quality.
Give us a SIP address (same as ) for each phone number!
- A usable one like: (not
Let us have both: =
And why not the same and SIP address by default with the subscription? 32

33. Why Don’t Telcos Offer Global UC Communication?
IMS: The thought was good and promised all. But it is complex and so far only used for POTS replication
Soft Switches/SBCs : Building/continuing on PSTN/POTS-like structures on top of IP
One major problem:
No UC or Multimedia peering between the operators
A Voice minute is (maybe) a Voice minute
But what is a Video or UC minute? – Codec, Screen size? Will never happen!
And an even worse problem:
IMS and SIP do not reach the users on the LANs!
Instead FXS ports for analogue phones are still being deployed
And SIP trunking of PBXs is hopefully a step in that direction – although POTS connectivity is the current level

34. Our CPEs Can even be the PBX – With full UC!
Use with standard SIP clients/phones/terminals. “Federates” with all, globally!
A service provider can also offer the UC PBX (by enabling that functionality).
Allows an existing PBX installation to be expanded and updated with SIP clients/phones/terminals for UC communication and for remote users.
Remote Users
PBX with non-SIP phones
WiFi Mobile
Registrar
Numbers integrated
Soft Client 34

35. And There is Provisioning…
jkjjk

36. Element Management System – The iEMS
Functions for Provisioning, Monitoring, Reporting, Diagnostics, Logging, Debugging, Support, Configuration and Upgrade. Available now with basic functionality.
Will handle both Ingate and Intertex Firewalls and SIParators.
Highly scalable, runs on PC servers under the Linux OS.
HTTPS/SOAP interface to the IX78. Can read and write all configuration parameters, as well as asynchronous reporting by the device (like SNMP traps).
Web based secure access to the iEMS. Customized portals for operators, installers and customers, for the purpose of administration, management and usage.
The iEMS has northbound interfaces for integrating with the operator’s OSS and Fault Management systems, using XML-RPC and/or SOAP.