1. Beyond POTS Replacement
Is SIP Trunking a step on that route?
Prepared for: INTERNET TELEPHONY Conference
Ingate’s SIP Trunking Summit
Miami, January 2010
By: Karl Erik Ståhl
President & CEO Intertex Data AB
Chairman Ingate Systems AB
© 2010 Intertex Data AB 1

2. SIP Trunking: Can it be More Than a New Connection?
SIP Trunking Provider
IP Cloud
SIP Trunk GW
PSTN
SIP System
TDM Trunk GW
Ingate SIParator®
Firewall
IP-PBX
Data & VoIP LAN

3. and More Than About Interoperability
Service providers
IP-PBXs
360 Networks
Airespring
AT&T
BandTel
Bandwidth.com
Broadvox
Cbeyond
Cellip
Cordia Corporation
Excel Switching
Gamma
Global Crossing
IP-Only
Juma Networks
Level 3
Netlogic
Nexvortex
Nuvox O1
Paetec
Primus
RNK Telecom
TDC
Tele2
Telia
Toplink
VoEX
VoIP Unlimited
Voxbone
More in pipeline.....
3Com
Aastra
Digium/Asterisk
Avaya
Cisco
Dialogic
Ericsson MX-One
Fonality
Innovaphone
Interactive Intelligence
Iwatsu
Microsoft
Mitel
NEC / Sphere
Nortel
Objectworld
Panasonic
Pingtel
Samsung
SER
Shoretel
Siemens
SIP-Gear
Swyx
More in pipeline....
SIP Trunk
Ingate SIParator®
-or-
Ingate Firewall
As Ingate have interoped with many IP-PBX vendors like <Repeat the one in the list> we will also guarantee interoperability for the Service Provider. The Service Provider only have to ensure interop with Ingate and then a number of different IP-PBXs will automatically be supported.
<Next slide>
Compliant with
Carrier Equipment
Acme Packet
Broadsoft
NexPoint
Sonus
Sylantro
SER
See:

4. and More Than Easy Deployment
Update
Installation Wizard

5. Benefits of SIP Trunking
Monthly cost savings
Single network for all communications
Lower cost of Moves, Adds and Changes
Disaster Recovery / Business Continuity
User provisioning
Steps of going beyond POTS replacement – Unified Communication
Mobility – Remote workers
Multimedia - Video, IM, Presence, Real Time Text RFC 4103, etc.
Real SIP address – like address
WiFi mobile phone communication
Let’s talk about this now!

6. There is Potential to Go Beyond!
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
And there is more than Voice: Presence, IM, Video, etc.

7. But have We Seen Much More than POTSoIP?
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. And When will We See the Next Step of Internet Usage?
SMTP created
HTTP created the Web
SIP can create global Live IP Person-to-Person Communication!

9. There is a Severe Infrastructure Problem…
A common Network and common Protocols changed our lives:
SMTP gave us global !
HTTP gave us the Web!
NATs and Firewalls were designed to allow such protocols.
IMS
(SIP based)
What about SIP for Live Person-to-Person Communication?
Internet
web
SIP does not traverse the common NATs and firewalls protecting the LANs . FW FW FW FW
LAN
LAN

10. Why are NATs and Firewalls Such Obstacles
Typical Internet protocol (SMTP, HTTP…)
Internet
HOST
SERVER
SIP is the Protocol for IP Communication
Person-to-Person,
BUT IT DOES NOT REACH THE USER’s!
SIP (and H.323…) connects Person-to-Person
Internet
PERSON
Locate the person
Set up a session +
Open real time media streams

11. And that is a Main Problem when SIP Trunking IP-PBXs
SIP Trunking Provider
Public Internet GW
PSTN
SIP System
SIP Trunking does not pass a SIP unaware NAT/firewall!
…and the firewall cannot be opened enough to make it work because of NAT.
Firewall
IP-PBX
Data & VoIP LAN
Soft Clients and Multimedia Terminals

12. And Hosted VoIP Suffers from the Same Problem
Telephone ports (FXS) on the CPE is a popular way to deploy IP telephony. By logically placing the SIP clients on the outside of the NAT/Firewall, unreliable work-around methods like STUN, TURN and ICE become unnecessary. However, this only gives POTS replication, often even stopping general SIP based services!
Internet
The 5060 SIP-port is just grabbed on the outside to the FXS ports!
(And lower level SIP ALGs often cause problems and do not handle more than basic scenarios.)
FXS ports (for plugging in analog phones)
is really POTS replication!

13. Let’s Use Real SIP Capable NAT/Router/Firewalls
IMS
Internet
SIP
No battery draining of WiFi mobile phones, otherwise caused by keep-alive packets* inhibiting sleep mode.
* Work-around methods for SIP NAT-traversal like STUN, TURN, ICE and Far End NAT Traversal use frequent keep-alive packets to keep holes in the NAT/Firewall open.
Problems solved where they occur
No special requirements on the SIP Client – Just standard SIP
Wired or wireless SIP clients (phones, soft clients, PDAs)
Intertex and Ingate CPEs have a SIP Proxy based Firewall/NAT
General, can handle complex call scenarios and all SIP services
Additional functionality available (PBX like functionality, ENUM, etc.)

14. Let’s Fix the SIP Trunking and at the Same Time Enable Going Beyond POTS Replication
SIP Trunking Provider
Public Internet GW
PSTN
SIP System
Remote Users
Intertex IX78
IP-PBX
Demarcation point of service and bringing SIP communication to the LAN
Data & VoIP LAN
Soft Clients and Multimedia Terminals

15. And Step in to the World of Global Live IP Communication
Global IP Connectivity
VoIP++
All SIP Services
Fix the NATs and firewalls and there is no reason to be caught in POTSoIPs islands! SIP connects globally and has lots of applications. It’s not magic – It’s just the SIP standard!

16. Beyond POTS: Mobility, Multimedia and Numbers
Internet
THIS LAN, SIP Trunking Summit
US, Miami
gunnar.hellstrom
@omnitor.se
Sweden
ADSL
Omnitor Case Study:
Multimedia
Voice
Video
Real-time Text RFC4103

17. Gunnar Hellström, Omnitor, Presenting Live from Sweden
Using Omnitor application Allan eC:
Voice: G.722 wide band codec
Video: H kbps
Real-time text: RFC4103
Using standard SIP over the Internet.
See presentation: Omnitor-TotalConversation
Other Live Demos Follow!

18. Internet Sweden US, Miami Sweden PSTN PSTN SIP/PSTN Gateway ADSL
gunnar.hellstrom
@omnitor.se
Sweden
ADSL
Internet
PSTN
SIP/PSTN
Gateway
SIP Trunk Provider 2
US, Miami
INTERTEX LAN
intertex.se
Sweden
PSTN
SIP/PSTN
Gateway
SIP Trunk Provider 1 3G
INGATE LAN
ingate.com
CELL
PSTN
THIS LAN, SIP Trunking Summit

19. Beyond POTS: Mobility, Multimedia and Numbers
We certainly want our home workers connected to the company PBX
And the same goes for our road warriors
at the hotel
at public WiFi
All should have all PBX services
Reached by extension number or DID
Place PSTN calls (displaying correct CallerID)
Voice mail, conferencing etc.
Presence, IM, video if supported by the PBX

20. Internet Sweden US, Miami Sweden PSTN PSTN SIP/PSTN Gateway ADSL
gunnar.hellstrom
@omnitor.se
Sweden
ADSL
Internet
PSTN
SIP/PSTN
Gateway
SIP Trunk Provider 2
US, Miami
INTERTEX LAN
intertex.se
Sweden
PSTN
SIP/PSTN
Gateway
SIP Trunk Provider 1 3G
INGATE LAN
ingate.com
CELL
PSTN
THIS LAN, SIP Trunking Summit
PBX Mobility with SIP Trunking (demo)
PSTN  my direct number
steeg  29 = my extension number
calle  23 (steeg)
PSTN  Intertex main  ext 29, 25s leave Voice Mail
calle  mobile in the hall
Voice Mail comes via

21. Beyond POTS: Mobility, Multimedia and Numbers
Presence is really useful
So is IM (Instant Messaging)
Laptops have cameras and good screens, so why not video?
Video conferencing does not have to be complex with huge cost and for internal use only.
And voice can actually be better than 3kHz AM-radio quality!
Who said MOS score 5 was perfect? Hardly HiFi?

22. Internet Sweden US, Miami Sweden PSTN PSTN SIP/PSTN Gateway ADSL
gunnar.hellstrom
@omnitor.se
Sweden
ADSL
Internet
PSTN
SIP/PSTN
Gateway
SIP Trunk Provider 2
US, Miami
INTERTEX LAN
intertex.se
Sweden
PSTN
SIP/PSTN
Gateway
SIP Trunk Provider 1 3G
INGATE LAN
ingate.com
CELL
PSTN
THIS LAN, SIP Trunking Summit
…and other SIP based applications (demo)
Presence, Instant Messaging (Who is available?)
Not restricted to own domain intertex.se, here also ingate.com
 (listen + video)
Wide band codec: “S” is not “F” anymore!
Video
Media goes the shortest way (just to the local switch here)
and we saw global SIP calls – not restricted to own domain

23. Beyond POTS: Mobility, Multimedia and Numbers
Telephone numbers WILL be around for long
We are simple too used to E.164 numbers and everyone has one
But they are really not particularly user friendly…
Would have been a success if we had used our fax numbers?
Operators often provide SIP names like
Not user friendly at all. For internal use only.
We want a real SIP address:
Just like our addresses
Let us have both: =
Service providers can do it
Here the Intertex and Ingate products do it!

24. Internet Sweden US, Miami Sweden PSTN PSTN SIP/PSTN Gateway ADSL
gunnar.hellstrom
@omnitor.se
Sweden
ADSL
Internet
PSTN
SIP/PSTN
Gateway
SIP Trunk Provider 2
US, Miami
INTERTEX LAN
intertex.se
Sweden
PSTN
SIP/PSTN
Gateway
SIP Trunk Provider 1 3G
INGATE LAN
ingate.com
CELL
PSTN
THIS LAN, SIP Trunking Summit
Telephone numbers and SIP addresses (demo)
Can we do global SIP calls over the SIP trunk? It is up to the operators!
E.g. Telia routes real SIP calls and don’t steal the media
(even though they are on a managed VoIP cloud)
 calle using
(IP  PSTN > PSTN  IP  only POTS voice)
sophie  calle using
(ENUM: IP  IP  quick, wide band codec, video)

25. ENUM – Using Phone Numbers but Staying on IP
2) ENUM lookup: Is there a SIP address for ?
Ask DNS: e164.arpa
Yeah try
3) Place the call directly to: IP IP
1) Dial Phone Number
PSTN
Not only for PSTN by-pass, but also for better voice and multimedia
Clients, Intertexes/Ingates, or service providers can use ENUM

26. SIP Capable Firewalls See us at ITEXPO Room A108!
Intertex Data AB
Rissneleden 45
SE Sundbyberg
Sweden
Tel:
Ingate Systems Inc.
7 Farley Road Hollis
NH 03049
United States
Ph: +1 (603)
Ph Sweden:

27. What about STUN, TURN, ICE and Far End Nat Traversal (FENT)?
STUN, TURN, ICE (client based) and FENT (typically done by SBCs) are alternative methods for working around non SIP capable NATs and Firewalls
Use them if required, e.g. for road warriors behind well behaved NATs with a not too tight firewalls
Ingate and Intertex can enable FENT to help SIP remote clients behind non SIP aware NATs and firewalls, e.g. Remote Users
But for SIP trunking and global and general SIP communication, you need something reliable and secure that also handles real complex call scenarios

28. Workaround Methods have their Limitations…
And with general SIP on several WAN-pipes: No chance!
ISSUES:
IMS
VoIP
IMS
RELIABILITY: STUN, TURN, ICE and Far End NAT Traversal (FENT) rely on guesswork of NAT/Firewall behavior – Thus never fully reliable. Unsuccessful calls – especially in complex scenarios, one way media, timeout during calls etc. etc.. 
SECURITY AND STABILITY: STUN, TURN, ICE are Client based, FENT is operator based (part of SBC). Both rely on punching holes in the Firewall and keeping NAT bindings open.
Internet
Keep-alive packets inhibit sleep mode, thus draining batteries of WiFi devices. 
STUN TURN
No control of QoS– where it is most important! FW FW FW FW
SECURITY POLICY: These workarounds require Firewalls to have large port ranges open from inside. FW is no longer in control of what is allowed into the LAN! STUN, TURN and ICE delegate control to the Client and can also be used for evil protocols. FENT delegates control to the Operator.
LAN
LAN