The VoIP Net: From POTS to Quality Unified Communications Globally © 2011 Intertex Data AB Prepared...

The VoIP Net: From POTS to Quality Unified Communications Globally

© 2011 Intertex Data AB

Prepared for: Ingate Systems 3 Day Seminar Unified Communications: SIP Trunking, Video, Collaboration and MoreITEXPO Conference, Austin, September 2011

By: Karl Erik Ståhl President Intertex Data ABCEO and Chairman Ingate Systems [email protected]

Also see Live Demo Presentation from ITEXPO SIP Trunking Summit Miami, February 2011! http://www.ingate.com/files/ITEXPO_Miami_2011_Presentations/Intertex%20-%20UC%20Across%20the%20Borders.pps

© 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


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?


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

Telcos can bring it together and offer better!

Why not better and beyond?We can Go Beyond POTS and Beyond Skype Now!


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 email) 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…


Shouldn’t We be Far Beyond PSTN and POTS?

RJ45

LAN Intranet Internet

Now we have a new global network: The IP Networks And we have a new standard: SIP

RJ11

POTS and PSTN have been there for 100 years

Black Phone

IP Phone

Presence Messaging VideoVoice

3.5 kHz isn’t HiFi, but MOS is 5!

Soft ClientWiFi Mobile


Europe

US

VPNTunnel

IP PBX

PBX

But IP Used to Replace Bits of the PSTN, keeping POTS

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?


SIP wasn’t Meant for Islands or Voice Only!

InternetInternet

CALL

RING!

Magic? – It’s just the SIP standard…

RING!RING!

CalleeProxy

CallerProxy

CalleeCaller

The SIP tapeziod

[email protected]

To initiate SIP calls:

- A proxy capable of routing (=DNS lookup!) - Add ENUM to use E.164 numbers

Outbound proxy for smartco.com

DNS partco.com

To receive SIP calls globally:

- A SIP server (Proxy Registrar)- SIP server domain published in DNS

[email protected]

Proxy Registrarfor partco.com


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…


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!


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).


But We Can Do It and We Are Close!

TeliaSonera Internet

SIParator IX78

ENUM

QoS IP Network QoS IP Network

CDR

CDR

MPLSMPLS

AT&TQwest Deutsche Telecom Internet

MPLS

Let’s see:- What is missing?- How can we do it?


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)

Soft Switch Soft Switch Soft SwitchSoft Switch

AT&TTeliaSonera Internet

Deutsche Telecom Internet

Qwest

IQ-Net = Quality bits (DSCP/TOS) honored

Multiple QoS Separated WAN Pipes are Common (Telia Network)

TR-069TR-069InternetInternet

IP-TVVoD

IP-TVVoD

IMSVoIP

IMSVoIP

PDA

All services must be available to multimedia terminals! – Over controlled high QoS pipes as well as the Internet.

VLANs or ADSL Virtual Circuits

The Multimedia LAN

The Multimedia LAN

WiFi InternetInternet

Application Innovation Requires it!

Telepresence

IP- PBX

But only the Internet is IP-peered globally. IP-peer the VoIP networks = IQ-Net!


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.


3-Party Video Conference with CDRs including Call Quality Metrics


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)


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)

FWPERSON FW PERSON

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

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

SERVER

FW

HTTP, SMTP, etc.

NAT/FW designed for this, but…

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


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.


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


A SIP Trunking E-SBC May Look Like a Specific “Gateway”!

PSTNGWSIP System

Data & VoIP LAN

IP-PBX

SIP Trunking Provider

SIParator®


…but can be an Enabler for SIP Services and Users Everywhere!

PSTNGWSIP System

Data & VoIP LAN

IP-PBX

UC Voice MailRemote

Users Ingate/Intertex E-SBCs enable SIP based Live UC Across the Borders! (SIP does not traverse ordinary NAT/Firewalls.)

SIP Trunking Provider

SIParator®


…and can Today Realize the Global UC Network!

TeliaSonera Internet

SIParator IX78

ENUM

QoS IP Network QoS IP Network

CDR

CDR

MPLSMPLS

AT&TQwest Deutsche Telecom Internet

MPLS


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


And What Is Required to “UC Communicate” Globally?

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.

ATA

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)

* See: http://www.ingate.com/files/ITEXPO_Miami_2011_Presentations/Intertex%20-%20UC%20Across%20the%20Borders.pps


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.


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 (email-like) and E.164 numbers


E-SBCs & SIP Capable Firewalls

Ingate Systems [email protected] Farley Road Hollis, NH 03049United StatesPh: +1 (603) 883-6569Tel sv: +46 8 6007750

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

See us at ITEXPO Room 9C!


E-SBCs & SIP Capable Firewalls

Ingate Systems [email protected] Farley Road Hollis, NH 03049United StatesPh: +1 (603) 883-6569Tel sv: +46 8 6007750

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

See us at ITEXPO Room 9C!

Some extra slides, with further details follows.


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: [email protected]) 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 simplificatio

n of

network architecture,

giving large savings for

the Telcos!


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 email) for each phone number!- A usable one like: sip:[email protected] (not [email protected])

Let us have both: +46 8 123456 = [email protected]

And why not the same email and SIP address by default with the subscription?


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


PBX with PBX with non-SIP non-SIP phonesphones

Our CPEs Can even be the PBX – With full UC!

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.

Registrar

Soft Client WiFi Mobile

Remote UsersNumbers integrated

Use with standard SIP clients/phones/terminals. “Federates” with all, globally!


And There is Provisioning…

jkjjk


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.

The VoIP Net: From POTS to Quality Unified Communications Globally © 2011 Intertex Data AB Prepared...

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