beyond pots replacement is sip trunking a step on that route? © 2010 intertex data ab 1 prepared...

Beyond POTS Replacement Is SIP Trunking a step on that route?

© 2010 Intertex Data AB 1

Prepared for: INTERNET TELEPHONY ConferenceIngate’s SIP Trunking SummitMiami, January 2010

By: Karl Erik Ståhl President & CEO Intertex Data ABChairman Ingate Systems [email protected]

SIP Trunking: Can it be More Than a New Connection?

PSTNIP CloudSIP Trunking

Provider

IP-PBXFirewall

Ingate SIParator®

TDM Trunk

GW

Data & VoIP LAN

SIP System

SIP Trunk

GW

and More Than About Interoperability

SIP Trunk

Ingate SIParator®

-or-Ingate Firewall

3ComAastraDigium/AsteriskAvayaCiscoDialogicEricsson MX-OneFonalityInnovaphoneInteractive IntelligenceIwatsuMicrosoftMitelNEC / SphereNortelObjectworldPanasonicPingtelSamsungSERShoretelSiemensSIP-GearSwyxMore in pipeline....

360 NetworksAirespringAT&TBandTel Bandwidth.com BroadvoxCbeyond CellipCordia CorporationExcel SwitchingGammaGlobal Crossing IP-Only Juma NetworksLevel 3

NetlogicNexvortexNuvoxO1 PaetecPrimus RNK TelecomTDC Tele2TeliaToplinkVoEX VoIP UnlimitedVoxboneMore in pipeline.....

Carrier EquipmentAcme PacketBroadsoftNexPoint

SonusSylantroSER

Compliant with

Service providers IP-PBXs

See: www.siptrunk.org


Installation Wizard

and More Than Easy Deployment

Update


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 email address• WiFi mobile phone communication

Let’s talk about this now!


There is Potential to Go Beyond!

RJ45

LAN Intranet Internet

Now we have a new global network: The IP Networks

RJ11

POTS and PSTN have been there for 100 years

Black Phone

IP Phone

3.5 kHz isn’t HiFi, but MOS is 5!

Soft ClientWiFi Mobile

And we have a new standard: SIP

And there is more than Voice: Presence, IM, Video, etc.


Europe

US

VPNTunnel

IP PBX

PBX

But have We Seen Much More than POTSoIP?

PSTN

Gateway

Gateway

TollBypass

IP PBX

Gateway

SoftSwitch

Gateway

Voice overBroadband

Very seldom VoIP connectivity between the VoIP IP clouds!

Most broadband VoIP providers still run calls between each other over the PSTN!

Are we stuckwith old POTStelephony over new wires?


HTTP created the Web

SMTP created Email

SIP can create global Live IP Person-to-Person Communication!

And When will We See the Next Step of Internet Usage?


There is a Severe Infrastructure Problem…

LAN

LAN

FW FW

FWFW

InternetInternet

email web

SIP does not traverse the common NATs and firewalls protecting the LANs .

IMS

(SIP based)

IMS

(SIP based)

What about SIP for Live Person-to-Person Communication?

A common Network and common Protocols changed our lives:

SMTP gave us global email!

HTTP gave us the Web!

NATs and Firewalls were designed to allow such protocols.


Why are NATs and Firewalls Such Obstacles

Typical Internet protocol (SMTP, HTTP…)

Internet

HOSTSERVER

SIP (and H.323…) connects Person-to-Person

Internet

PERSONPERSON

SIP is the Protocol for IP Communication Person-to-Person,

BUT IT DOES NOT REACH THE USER’s!

Locate the person Set up a session+ Open real time media streams+

Data & VoIP LAN

Soft Clients and Multimedia Terminals

PSTNPublic

Internet

SIP Trunking Provider GW

IP-PBX Firewall

SIP Trunking does not pass a SIP unaware NAT/firewall!

…and the firewall cannot be opened enough to make it work because of NAT.

SIP System

And that is a Main Problem when SIP Trunking IP-PBXs


And Hosted VoIP Suffers from the Same Problem

InternetInternet

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.)

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!

FXS ports (for plugging in analog phones) is really POTS replication!


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.

Let’s Use Real SIP Capable NAT/Router/Firewalls

InternetInternet

Problems solved where they occur

No special requirements on the SIP Client – Just standard SIP

Wired or wireless SIP clients (phones, soft clients, PDAs)

SIP

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.)

IMSIMS


PSTNPublic

Internet

SIP Trunking Provider

GWSIP System

Data & VoIP LAN

IP-PBX

Demarcation point of service and bringing SIP communication to the LAN

Soft Clients and Multimedia Terminals

Intertex IX78

Remote Users

Let’s Fix the SIP Trunking and at the Same TimeEnable Going Beyond POTS Replication


And Step in to the World of Global Live IP Communication

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!

VoIP++

Global IP Connectivity

All SIP Services


Internet

THIS LAN, SIP Trunking Summit

[email protected]@ingate.com [email protected]

US, Miami

Multimedia• Voice• Video• Real-time Text RFC4103

Omnitor Case Study:

[email protected]

Beyond POTS: Mobility, Multimedia and Numbers

[email protected]

Sweden

ADSL

Gunnar Hellström, Omnitor, Presenting Live from Sweden

Using Omnitor application Allan eC:

Voice: G.722 wide band codec

Video: H.264 300kbps

Real-time text: RFC4103

Using standard SIP over the Internet.

See presentation: Omnitor-TotalConversation

Other Live Demos Follow!

INGATE LAN

ingate.com

InternetUS, Miami


[email protected]@ingate.com [email protected]

CELL

PSTN

INTERTEX LAN

intertex.se

Sweden

3G

[email protected]

PSTN

SIP/PSTNGateway

SIP Trunk Provider 1

PSTNSIP/PSTNGateway


[email protected]

Sweden

ADSL



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

INGATE LAN

ingate.com

InternetUS, Miami


([email protected]) [email protected]

CELL

PSTN

INTERTEX LAN

intertex.se

Sweden

3G

[email protected]

PSTN

SIP/PSTNGateway


PSTNSIP/PSTNGateway


PBX Mobility with SIP Trunking (demo)PSTN +46 8 12345629 my direct numbersteeg 29 = my extension numbercalle 23 (steeg)PSTN +46 8 12345600 Intertex main ext 29, 25s leave Voice Mailcalle mobile in the hallVoice Mail comes via email

[email protected]

[email protected]

Sweden

ADSL



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?

Presence is really useful

INGATE LAN

ingate.com

InternetUS, Miami


[email protected] ([email protected])

CELL

PSTN

INTERTEX LAN

intertex.se

Sweden

3G

[email protected]

PSTN

SIP/PSTNGateway


PSTNSIP/PSTNGateway


…and other SIP based applications (demo)• Presence, Instant Messaging (Who is available?)Not restricted to own domain intertex.se, here also ingate.com [email protected] [email protected] (listen + video)• Wide band codec: “S” is not “F” anymore!• VideoMedia goes the shortest way (just to the local switch here)and we saw global SIP calls – not restricted to own domain

[email protected]

[email protected]

Sweden

ADSL



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 email have been a success if we had used our fax numbers?

Operators often provide SIP names like [email protected] Not user friendly at all. For internal use only.

We want a real SIP address: [email protected] Just like our email addresses

Let us have both: +46 8 1234567 = [email protected]!- Service providers can do it- Here the Intertex and Ingate products do it!

INGATE LAN

ingate.com

InternetUS, Miami


[email protected] [email protected]

CELL

PSTN

INTERTEX LAN

intertex.se

Sweden

3G

[email protected]

PSTN

SIP/PSTNGateway


PSTNSIP/PSTNGateway


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)0850004195 calle using 08 12345629 (IP PSTN ------> PSTN IP only POTS voice)sophie calle using 08 12345629 (ENUM: IP IP quick, wide band codec, video)

[email protected]

[email protected]

Sweden

ADSL


IPIP

PSTN

ENUM – Using Phone Numbers but Staying on IP

IPIP

Not only for PSTN by-pass, but also for better voice and multimedia

Clients, Intertexes/Ingates, or service providers can use ENUM

+46 8 12345629 [email protected]

2) ENUM lookup: Is there a SIP address for +46812345629?Ask DNS: 9.2.6.5.4.3.2.1.8.6.4.e164.arpaYeah try sip:[email protected]

1) Dial Phone Number 08 12345629

3) Place the call directly to: sip:[email protected]


SIP Capable Firewalls

Ingate Systems [email protected] Farley Road HollisNH 03049United StatesPh: +1 (603) 883-6569Ph Sweden: +46 8 6007750

Intertex Data [email protected] 45 SE-174 44 SundbybergSwedensip:[email protected]: +46 8 6282828

See us at ITEXPO Room A108!


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

What about STUN, TURN, ICE and Far End Nat Traversal (FENT)?


Workaround Methods have their Limitations…

IMSIMS

VoIPVoIP

IMSIMS

LAN

LAN

FW FW

FWFW

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.. Internet Internet Keep-alive packets

inhibit sleep mode, thus draining batteries of WiFi devices.

STUN TURN

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.

No control of QoS– where it is most important!

No control of QoS– where it is most important!

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.

ISSUES:And with general SIP on several

WAN-pipes: No chance!

beyond pots replacement is sip trunking a step on that route? © 2010 intertex data ab 1 prepared...

Documents

email sip

ims sip

general sip

sip proxy

pdas sip intertex

person internet person

sip trunking ippbxs

real sip address