voip bypass

8/8/2019 Voip Bypass

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Alan Quayle Business and Service Development. All Rights Reserved.

MARKET ASSESSMENTOF VOIP BYPASSROAMING AND

OPERATOR IMPACT /

PROPOSITIONEVALUATING THE VOIP ROAMING BYPASS

MARKET: REGULATIONS, TARRIFS,

DEPLOYMENT OPTIONS / FEASIBILITY,

DIRECT MARKET FEEDBACK, OPERATOR

BUSINESS CASE, AND OPERATOR IMPACT

ANALYSIS / PROPOSITION WITH EMPHASIS

ON SUPPLIER PARTNER SOLUTION USING

HANDSET INTERCEPT OF ROAMING CALL

A L A N Q U A Y L E B U S I N E S S A N D S E R V I C ED E V E L O P M E N T


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Alan Quayle Business and Service Development, All Rights Reserved

CONTENTS

ABSTRACT 6

INTRODUCTION AND BACKGROUND 7

PURPOSE 7ROAMING DEFINITIONS AND TRANSFER ACCOUNT PROCEDURE (TAP) 10ROAMING DEFINITIONS 10TRANSFER ACCOUNT PROCEDURE 10TAP3 IN USE ERROR! BOOKMARK NOT DEFINED.WHAT DOES THE FUTURE HOLD FOR TAP? ERROR! BOOKMARK NOT DEFINED.TARRIFING BACKGROUNDER ERROR! BOOKMARK NOT DEFINED.MOBILE TERMINATION ERROR! BOOKMARK NOT DEFINED.MOBILE INTERNATIONAL ROAMING ERROR! BOOKMARK NOT DEFINED.EXPOSURE TO TARIFF PRESSURE FROM ROAMING AND TERMINATION ERROR! BOOKMARK NOT

DEFINED.COST DISTRIBUTION ON A MOBILE CALL 13CALLING SCENARIOS: WHEN ABROAD AND CALLING HOME 13CALLING SCENARIOS: WHEN ABROAD AND BEING CALLED FROM HOME 14

VOIP ROAMING BYPASS TECHNOLOGY DESCRIPTIONS 16

GATEWAY OPTION 16DESCRIPTION 16

OUTGOING CALL SCENARIOS 16INCOMING CALL SCENARIOS 18ISSUES 21HANDSET INTERCEPT OPTION 21DESCRIPTION 21ISSUES 25OTHER OPTIONS 25DUAL MODE (GSM WIFI) HANDSETS 25MOBILE CENTRIC START-UPS: JAJAH AND REBTEL 27JAJAH 27REBTEL 30INTERNET BRANDS: SKYPE, YAHOO! ERROR! BOOKMARK NOT DEFINED.

YAHOO! WORKING IN PARTNERSHIP WITH OPERATORS ERROR! BOOKMARK NOT DEFINED.SKYPE AND GOOGLE LIKELY HEAD-ON COMPETITION ERROR! BOOKMARK NOT DEFINED.CONSUMER BRANDS: NOKIA, APPLE, AND SONY ERROR! BOOKMARK NOT DEFINED.COMPARATIVE ANALYSIS ERROR! BOOKMARK NOT DEFINED.

INTERNATIONAL ROAMING REGULATION ERROR! BOOKMARK NOT DEFINED.


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Alan Quayle Business and Service Development. All Rights Reserved

EUROPEAN ROAMING REGULATION ERROR! BOOKMARK NOT DEFINED.CURRENT STATE OF EU REGULATION ERROR! BOOKMARK NOT DEFINED.MOBILE VIRTUAL NETWORK AND SMALL OPERATORS ERROR! BOOKMARK NOT DEFINED.OPERATOR REACTIONS TO EU INITIATIVE ERROR! BOOKMARK NOT DEFINED.IMPACT OF REGULATION ERROR! BOOKMARK NOT DEFINED.EUROPEAN MEMBER STATE POSITIONS ERROR! BOOKMARK NOT DEFINED.

ASIAN INTERNATIONAL ROAMING REGULATION ERROR! BOOKMARK NOT DEFINED.NORTH AMERICAN INTERNATIONAL ROAMING REGULATION ERROR! BOOKMARK NOT

DEFINED.

MARKET REACTION SUMMARY ERROR! BOOKMARK NOT DEFINED.

ASIA PACIFIC REGION RESULTS ERROR! BOOKMARK NOT DEFINED.EUROPE AND MIDDLE EAST RESULTS ERROR! BOOKMARK NOT DEFINED.NORTH AMERICA RESULTS ERROR! BOOKMARK NOT DEFINED.

MARKET SIZING ERROR! BOOKMARK NOT DEFINED.

ROAMING MARKET SIZE ERROR! BOOKMARK NOT DEFINED.OPERATOR EXPOSURE TO EU ROAMING REGULATION PROPOSALS ERROR! BOOKMARK NOT

DEFINED.THE PERCENTAGE OF REVENUES EXPOSED TO EU WIRELESS ERROR! BOOKMARK NOT

DEFINED.THE EXPOSURE OF WIRELESS REVENUES TO ROAMING ERROR! BOOKMARK NOT DEFINED.THE IMPACT ON 2008 FINANCIALS ERROR! BOOKMARK NOT DEFINED.ADDRESSABLE MARKET USING VOIP BYPASS ERROR! BOOKMARK NOT DEFINED.OPERATOR TARGETING ERROR! BOOKMARK NOT DEFINED.MARKET EVOLUTION ERROR! BOOKMARK NOT DEFINED.

OPERATOR IMPACT AND PROPOSITION ERROR! BOOKMARK NOT DEFINED.

OPERATOR SITUATION ERROR! BOOKMARK NOT DEFINED.IMPACT OF VOIP BYPASS TO OPERATOR'S BUSINESS ERROR! BOOKMARK NOT DEFINED.VOIP BYPASS PROPOSITION FOR OPERATOR ERROR! BOOKMARK NOT DEFINED.

CONCLUSIONS AND RECOMMENDATIONS ERROR! BOOKMARK NOT DEFINED.

APPENDIX 1 ACRONYMS ERROR! BOOKMARK NOT DEFINED.

APPENDIX 2 COMPANY DATA POINTS ERROR! BOOKMARK NOT DEFINED.

DEUTSCHE TELEKOM ERROR! BOOKMARK NOT DEFINED.


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FRANCE TELECOM ERROR! BOOKMARK NOT DEFINED.VODAFONE ERROR! BOOKMARK NOT DEFINED.TELECOM ITALIA ERROR! BOOKMARK NOT DEFINED.COSMOTE ERROR! BOOKMARK NOT DEFINED.TELEFONICA ERROR! BOOKMARK NOT DEFINED.BELGACOM ERROR! BOOKMARK NOT DEFINED.

MOBISTAR ERROR! BOOKMARK NOT DEFINED.

APPENDIX 3: INTERVIEW QUESTIONNAIRE ERROR! BOOKMARK NOT DEFINED.

APPENDIX 4: INTERVIEW RESULTS ERROR! BOOKMARK NOT DEFINED.

APPENDIX 5: JAJAH CALLING RATES ERROR! BOOKMARK NOT DEFINED.

APPENDIX 6: EUROPEAN MEMBER STATE REGULATORY POSITIONS ERROR!

BOOKMARK NOT DEFINED.

NATIONAL REGULATORS AND MINISTRIES ERROR! BOOKMARK NOT DEFINED.FINLAND: EFFECTIVE COMPETITION ERROR! BOOKMARK NOT DEFINED.FRANCE: IDENTIFICATION OF A RESTRICTED OLIGOPOLY ERROR! BOOKMARK NOT DEFINED.SPAIN: STRONG ADVOCATE OF NO EU-WIDE REGULATIONERROR! BOOKMARK NOT DEFINED.IRELAND: NO SINGLE OR JOINT DOMINANCE IDENTIFIED ERROR! BOOKMARK NOT DEFINED.ITALY: NO INDIVIDUAL OR COLLECTIVE MARKET POWER ERROR! BOOKMARK NOT DEFINED.NORWAY: MINISTERIAL SUPPORT OF EC REGULATION ERROR! BOOKMARK NOT DEFINED.OVERLAP WITH NRA MARKET 11 REVIEWS ERROR! BOOKMARK NOT DEFINED.


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TABLE O F FI GUR E S

Figure 1. International Calling from the Home Network Case_________________________________ 9Figure 2. International Roaming Case ____________________________________________________ 9Figure 3. Transferred Account Procedure (TAP) GSM Billing and Accounting ________________ 11

Figure 4. The collection and transfer of TAP information between the Home Public Land MobileNetwork and the Visited Public Land Mobile Network_______________________________________ 12Figure 5. Forecast Drag on EBITDA From Mobile Termination _______ Error! Bookmark not defined.Figure 6. Mobile Termination Rates and Announced Cuts across Europe Error! Bookmark not defined.Figure 7. Revenue and EBITDA Breakdown of Roaming Cut (source Lehman Brothers) _______Error!

Bookmark not defined.Figure 8. Proportion of Domestic Mobile Revenues from International Roaming _ Error! Bookmark not

defined.Figure 9. Exposure to Tariff Pressure from Roaming and Termination by Type of Operator _____Error!

Bookmark not defined.Figure 10. Mobile Revenue Pie and Drivers of Mobile Revenue Growth _ Error! Bookmark not defined.Figure 11 When Abroad and Calling Home _______________________________________________ 13Figure 12 When Abroad and being Called from Home ______________________________________ 14Figure 13. Classic GPRS Roaming ______________________________________________________ 16Figure 14. Option 1A, Gateway Centric: Circuit Switch far-end_______________________________ 17Figure 15. Option 1B, Gateway Centric: Packet Switch far-end _______________________________ 17Figure 16. Cisco Voice Toll Bypass Illustration ____________________________________________ 18Figure 17. Classic GSM Roaming Call Forward Call Model__________________________________ 19Figure 18. Roaming call forward Gateway Centric (Option 1) _______________________________ 19Figure 19. Roaming Call Forward (Option 2)______________________________________________ 20Figure 20. Roaming Call Forward (Option 3)______________________________________________ 20Figure 21. Handset Centric: Circuit Switched Far End _____________________________________ 23Figure 22. Handset Centric: Packet Switched Far End______________________________________ 23Figure 23. Handset Centric: Incoming Calls ______________________________________________ 24Figure 24. JAJAH Call Flow ___________________________________________________________ 29Figure 25. JAJAH Signalling__________________________________________________________ 29Figure 26. APAC Market Feedback ______________________________ Error! Bookmark not defined.

Figure 27. EMEA Market Feedback ______________________________ Error! Bookmark not defined.Figure 28. NAR Market Feedback________________________________ Error! Bookmark not defined.Figure 29. International Roaming Traffic and Revenues 2003-2010 ____ Error! Bookmark not defined.Figure 30. Roaming Contribution to Mobile Revenues and EU Regulatory Impact Error! Bookmark not

defined.Figure 31. Percentage of Group Revenues from EU Wireless __________ Error! Bookmark not defined.Figure 32. Percentage of Revenue from Roaming ___________________ Error! Bookmark not defined.Figure 33. Impact on Free Cash Flows ____________________________ Error! Bookmark not defined.Figure 34. Skype Out Rates from US All figures USD ______________ Error! Bookmark not defined.Figure 35. Example of Operator and Standard Roaming Rates All Figures GBP Error! Bookmark not

defined.


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ABSTRACT

There are emerging technologies that use functionality in the mobile phone to extend VoIP frombeing employed by the HPLMN to offer cheap international calls tariff rate to home subscribers. Tonow enable the HPLMN to employ VOIP roaming bypass, which enables HPLMN subscribers to

benefit from making cheap calls (local or international) when roaming abroad.. These technologiesare addressing the international roaming market, at $15B USD global business, and currently one ofthe mobile industries most lucrative revenue stream.


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INTRODUCTION AND BACKGROUND

PURPOSE

VoIP has created the conditions whereby valuable niches in voice communications are exploited.

For example, International Calling Cards, Skype, Yahoo!, Vonage, JaJah, RebTel, TruPhone,ConnectmeAnywhere, Hullo, etc. are all service providers offering voice services that bypassinternational call charges, or mobile network charges. Generally these services target a segment ofinternational calling / mobile customers. For example, international calling cards are usedextensively by foreign workers to call home. PC based solutions enable friends or small internationalbusinesses to hold calls at free / low cost.

Both JAJAH and REBTEL are focused on the mobile communication segment for internationalcalling from the users home network, see Figure 1. Breaking free of the PC and the internationalcalling card models. However, the JAJAH client solution could also be extended to the internationalroaming case as well, see Figure 2, though this has not yet been announced.

For the REBTEL RebIN service:

The User goes online to Rebtel website and sign up for the service. The user isasked to provide their mobile phone number and the mobile phone numbers of thefriends they want to call. The Rebtel system then generates a local number for theuser and a local number for each of his friends. Once the local numbers aregenerated, the Rebtel website sends a SMS and/or email to the user and to theirfriends. The SMS/email provides a local number for the user to dial into the PSTN VoIP gateway. When the user calls the local number, the call is routed by hisHPLMN to a local PSTN VoIP gateway. The VoIP Gateway will use his calling lineID to identify the user and the called party number to connect the call to his friendover a VoIP network. When his friend answers the call. The user asks his friend tohang up and dials the local number that was previously sent to him by email or SMS.

When his friend dials his local number, his mobile phone is connected to theRebtels local PSTN VoIP Gateway. The Rebtel system then joins the two local callstogether over a VOIP network. Rebtel has simply employed SMS and email as thebearer to inform the user the local phone number to dial to connect to Rebtels localPSTN VoIP Gateway.

For JAJAH it offers two methods.

Method 1: User goes onto their website and enters their mobile phone number andthe called mobile phone number. The JAJAH website then invokes their system tomake dial out to the user and the called number. Once the call is answered by bothparties, the JAJAH system connects the two call legs to together.

Method 2: To provide a Symbian OS based Java client to be installed on the usershandset. Once the JAJAH client is installed, the client will ask the user to nominatehis home country. The JAJAH client uses this information to select the local PSTN VoIP gateway number to use. When the user dials an international number, theJAJAH client intercepts this telephony request and makes a circuit switched voicecall to the local PSTN VOIP gateway. Once the call is connected to the PSTNVOIP Gateway, there are two possible methods for the JAJAH client to pass the


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original called party number to the local PSTN VOIP Gateway. One method is tosend the original called party number by SMS to the PSTN VOIP Gateway. Analternative method is to send the original called party number by using DTMF digitstransferred over the established circuit switched voice connection. When the PSTN VOIP gateway receives the incoming call, it waits for the original called partynumber to be sent by the client by DTMF. The PSTN VOIP Gateway simply

performs circuit switched voice TDM conversion to SIP VOIP connection. One ofthe most common PSTN VOIP Gateway in use is Cisco Access Server (e.g. CiscoAS53500, AS5350, AS5400).

In addition there are emerging technologies that use functionality in the mobile phone to extendVoIP bypass from international calling from the Home network, to international roaming, see Figure2, referred to as the Handset and Gateway options. These technologies are addressing theinternational roaming market, at $15B USD global business.

An example of the Gateway Option, Operator sites in Singapore a GGSN, noted as VFs GGSN(SG). The roaming UE sets up a PDP context to VF's GGSN in Singapore using VF's home APN.When the roaming UE initiates a PDP context activation request, the M1's SGSN interrogates itsmobile packet core DNS to resolve the APN to a GGSN IP address. Hence VoIP over GPRS to thelocal GGSN bypassing the VPLMN for incoming and outgoing calls.

An example of the Handset options is when the customer turns on their mobile phone inSingapore it goes through a two phase set-up. It registers with the VF home network then is ablegather local MGW and MS-ISDN/IMSI details, the phone then registers as a SG phone. Or possiblythe UEs roaming table is pre-programmed with the MGW and MS-ISDN/IMSI details. So theroaming UE masquerades as a local phone for incoming calls and uses the same procedure as JAJAHfor outgoing international calls.

This purpose of this market analysis is to:

Provide an introduction to international roaming;

Review the emerging VoIP bypass technologies;

Review the regulatory situations in the European, Asia-Pacific and North Americanregions;

Review the international roaming market, and assessing the impact thesetechnologies present;

Present the findings from a market survey of key operators from APAC, EMEAand NAR; to determine the issues, concerns and acceptability of the technologies;

Examine which market segments would be most likely to adopt these technologies,e.g. to MVNOs, MNOs and Internet Brands;

A specific assessment of the technologys impact upon Operator; and

Provide Supplier with as assessment on whether there is a market for thistechnology.


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Figure 1. International Calling from the Home Network Case

Figure 2. International Roaming Case

Home Country A Called Country B

HPLMNPLMN

Bypass VOIP network

Called Country C

Called UE ofCountry B

PLMN

Called UE of

Country C

Calling UE of

Country A

Operators subscribers pay local call tariffrate for international calls by routinginternational calls via bypass VOIPnetwork

Home Country ARoaming Country B

HPLMN

VPLMN

Bypass VOIP network

Called Country C

RoamingUE

Other PLMNsOperator subscribers pay local call tariffrate for international calls to home countryor to other countries by routinginternational calls made from VPLMN viabypass VOIP network


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ROAMING DEFINITIONS AND TRANSFER ACCOUNT PROCEDURE (TAP)

ROAMING DEFINITIONS

International Roaming Call

A call made by a subscriber roaming on a visited network to someone in anothercountry. The called person may be a subscriber on the same home mobile networkasthe calling person or a fixed network subscriber.

Local Roaming Call

A call made by a subscriber roaming on a visited network to another person in the country of the visited network.

The called person may be:

A mobile subscriber of the visited network

A mobile subscriber on another licensed mobile network in the same country

A fixed network subscriber in the same country

In this case, the person calls a party in the same country in which he or she isvisiting.

TRANSFER ACCOUNT PROCEDURE

GSM is founded on the concept of roaming - allowing customers from other networks andcountries to use their mobiles when they visit any country or network. Sounds simple. But with some

600+ GSM networks now operational, the GSM Association estimates that more than 20,000individual roaming agreements are in place between its operators, with more being added every day.So behind the simple objective of global roaming lies a complex process that gathers informationabout each call, about each caller and takes a standardized approach to the charges being incurred.

These individual roaming agreements, which change over time, and are subject to local regulatoryinfluences result in a complex ever changing patchwork quilt of termination changes. Forexample, in the UK VoIP call charges to a mobile phone are generally 10 times the charge to a fixedline, see Error! Reference source not found.. While say in Singapore, there is no difference inVoIP call charges to fixed or mobile.

Within the GSMA the Transferred Account Data Interchange Group (TADIG) is responsiblefor defining the interchange of billing data between different network operators by defining andimplementing the TAP protocol. TAP 3 is the version in use today, see Figure 3.

The Transferred Account Procedure is the mechanism by which operators exchange roamingbilling information. This is how roaming partners are able to bill each other for the use of networksand services through a standard process. Much of the traffic carried by a GSM Public MobileNetwork (PMN) either originates, or terminates in another network. The operator of the local fixednetwork charges the wireless operator for each call that terminates at one of its fixed subscribers.


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And likewise, the GSM operator will charge the fixed operator for each call made to a mobilenumber from a fixed line.

Therefore GSM network operators and their local fixed counterparts usually negotiate aninterconnect agreement to make charging as simple as possible. The other fixed internationaloperators have normally already negotiated similar agreements amongst themselves, see Figure X.

Therefore, in order to place a call from a German PMN to a Canadian fixed phone, it is notnecessary for the German PMN operator to negotiate a price with a Canadian fixed networkoperator.

The German PMN operator negotiates a price with the German fixed network operator. TheGerman fixed network operator then negotiates a price with the Canadian fixed network operator.So, the German fixed network operator passes this call cost back to the German PMN. This meansthat the German PMN has to recoup the cost of the call from its subscribers either directly (retailbilling), or via the appropriate Service Provider (wholesale billing). This form of inter-administrationaccounting covers the division of revenue between both fixed and mobile networks. It does not,however, cover the costs incurred by foreign subscribers whilst roaming in other networks.

Consider the case of a French subscriber calling a Canadian fixed phone from within a Germannetwork. The German fixed network will still charge the German PMN for the leg of the call placedto the Canadian number. In this case, the German PMN does not receive any revenue from its ownsubscriber. In order to recoup the costs incurred by the call, the German PMN must charge thehome mobile network operator, here the French PMN, to cover the costs incurred by the Frenchmobile subscriber.

Figure 3. Transferred Account Procedure (TAP) GSM Billing and Accounting

Figure 4 illustrates the collection and exchange of information required to support TAP. Thedetails of the calls made by a subscriber roaming in a visited network (VPLMN) are recorded by theserving switch, the Mobile Switching Centre or MSC listed above. Each call produces one or more


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call records. The GSM standard for these call records is defined in GSM 12.05, although many switchvendors use their own proprietary formats.

The call records produced by the MSC are transferred on a regular basis to the billing system ofthe VPLMN for pricing or rating. Those call records produced on behalf of roaming subscribers, willbe converted and grouped in files under the TAP format.

The TAP files are generated and sent, at the latest, 36 hours from call end time. This means thatoperators can send 1 or many TAP files per day. TAP files contain rated call information accordingto the operator's Inter Operator Tariff (IOT), plus any bilaterally agreed arrangements or discountingschemes.

The transfer of TAP records between the visited and the home mobile networks may beperformed directly, or more commonly, via a Clearinghouse. Invoicing between the operators thennormally happens once per month.

On reception by the HPMN, the TAP record is converted into an internal format and addedtogether with any call records produced by the subscriber whilst within the home network.

If a service provider serves the subscriber then the records will form the basis of the wholesalebilling between the HPMN and that Service Provider, an example is for a HPLMN operator toexchange TAP3 files with its MVNOs to settle wholesale billing agreements between an operator andits MVNOs. On receipt of the information from the HPMN, the Service Provider may re-rate thecalls according to its own tariff plans and produce an itemized bill, including call detail, for thesubscriber.

Figure 4. The collection and transfer of TAP information between the Home PublicLand Mobile Network and the Visited Public Land Mobile Network


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COST DISTRIBUTION ON A MOBILE CALL

CALLING SCENARIOS: WHEN ABROAD AND CALLING HOME

When abroad and Calling Home, the call is managed by the host operator (VPLMN), seeFigure 5. The host operator passes the call via 'international transit' to the home operator. Thehome operator (HPLMN) connects to the calling parties operator and establishes the call.

Cost Components:

Host Operator Origination Fee (Step 1)

International Transit Fee (Step 2)

Call Termination Fee (Step 3)

Home Operator Mark-up

Figure 5 When Abroad and Calling Home

Host Operator Origination Fee

Negotiated in the bi-lateral agreement, generally in the range of 8c-50c (Euro)

International Transit Fee

Due to the competition that exists in inter-country transport this fee is between2-10c (Euro) for Tier 1 countries. Note figure dependent upon volume androute - not published as highly competitive B2B business

Call Termination Fee

Charge Step 3


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As there is no competition for this fee, it has been the focus on regulations.Generally now in the range of 6-15c (Euro) See Error! Reference source notfound..


Covering operating costs and profit, Another area of regulator concern, mark-ups range of 25-200%.

Total Wholesale cost of 16c 75c

Total Retail cost of 20c 2.25 Euro

CALLING SCENARIOS: WHEN ABROAD AND BEING CALLED FROM HOME

A friend calls you on your mobile phone while you're roaming, see Figure 6. His operator routesthe call initially to your home operator (which may or may not be the same). Your home operatorforwards the call to the host operator you are currently roaming on in the destination country, via

'international transit.' The host operator receives the forwarded call, connects you using its networkand establishes your friends originated call.

Cost Components:

Step 1 Your friend will be charged a normal call by his home operator for callingyou.

Steps 2 and 3 Your home operator will charge you a tariff which includes inter aliathe international transit fees to forward the call to you in the destination countryand the cost for terminating the call on the host network.

International Transit Fee, Call Termination Fee, Home Operator Mark-up

Figure 6 When Abroad and being Called from Home


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International Transit Fee

Due to the competition that exists in inter-country transport this fee is between2-10c (Euro) for Tier 1 countries

Note figure dependent upon volume and route

Host Operator Call Termination Fee

As there is no competition for this fee, it has been the focus on regulations

Generally now in the range of 6-15c (Euro)

Note some countries (e.g. in the Middle East) have significantly highertermination fees


Covering operating costs and profit

Another area of regulator concern, mark-ups range of 25-200%

Total Wholesale cost of 8c 25c

Total Retail cost of 10c 75c Euro


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VOIP ROAMING BYPASS TECHNOLOGY DESCRIPTIONS

This section describes the two principle VoIP bypass options, as well as several other optionsavailable to customers, such as dual mode handsets, JAJAH, REBTEL and discusses the plans fromthe major Internet brands such as Skype and Yahoo!

GATEWAY OPTION

DESCRIPTION

Figure 7 shows the usual GPRS (General Packet Radio Service) roaming case. Lets take a real- world case and assume that Operator is the home operator, and a VF customer is roaming inSingapore on the M1 network. When the roaming Operator UE (User Equipment) requests a PDP(Packet Data Protocol) context, the M1 SGSN (Serving GPRS Service Node) interrogates its mobilepacket core DNS (Domain Name Server) to resolve the APN (Access Point Name) to a GGSN(Gateway GPRS Service Node) IP address in the Home Country. This sets out the default situation,from which we will explore the Gateway option.

Figure 7. Classic GPRS Roaming

OUTGOING CALL SCENARIOS

In the Gateway option, shown in Figure 8 and Figure 9, Operator sites in Singapore a GGSN,noted as VFs GGSN (SG). The roaming UE sets up a PDP context to VF's GGSN in Singaporeusing VF's home APN. When the roaming UE initiates a PDP context activation request, the M1's

SGSN interrogates its mobile packet core DNS to resolve the APN to a GGSN IP address.

In this case, the M1's DNS cannot directly resolve VF's APN. Instead, the M1 MPC (MobilePacket Core) DNS forwards the request to VF's MPC DNS for APN resolution. The VF MPC DNSwill resolve this APN to a VF GGSN deployed in Singapore and return the resolved IP address tothe M1's MPC DNS. M1 MPC DNS then returns the GGSN IP address to the M1 SGSN. The M1SGSN then sets up a PDP context to the VF GGSN deployed in Singapore.

Home Country Roaming Country

HPLMN PS Core VPLMN PS CoreGRX

SGSNSGSNGGSNGGSN

Home

Portal

UE established a PDP

context from VPLMN toHPLMN GGSN usingHome APN


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Figure 8. Option 1A, Gateway Centric: Circuit Switch far-end

Figure 9. Option 1B, Gateway Centric: Packet Switch far-end


GRX /

InternetHPLMN CS Core

VPLMN PS Core

GGSN

SIPVOIPGW

SIPVOIPGW

SGSNSGSN

SIPserver

SIPserver

Home operator hasGGSN and SIP serverin roaming countries

A SIP call session is set up betweenthe roaming UE and the SIP serverwhich functions as a SIP Redirectserver

Legend:

VOIP media transported as RTP/UDP/IP

SIP call session signalling path over GPRS packet data connection

Circuit switched voice call

Logical SIP call session association between roaming UE and HPLMN Operators SIP server

Roaming UEwith SIP client


GRX /Internet

HPLMN PS Core

VPLMN PS Core

GGSNIMS

SGSNSGSN

SIPserver

SIPserver

Home operator has

GGSN and SIP serverin roaming countries

A SIP call session is set upbetween the roaming UE andthe SIP server which functionsas a SIP Redirect server

Roaming UE

with SIP client

Called UE with SIPclient

Legend:


SIP call session signalling path over GPRS packet data connection

Logical SIP call session association between roaming UE and HPLMN Operators SIP server


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With the PDP context established to VFs GGSN (SG), Operator also has within the sameprivate IP space a SIP server. As they are both on the same private IP space, the MGW can directlyaddress the roaming mobile. This avoids the need to be part of M1s DMZ as M1 maybe unlikelyto co-operate through the potential loss in revenue.

With this configuration VoIP calls are set up over the PSDN (Packet Switched Data Network).

This is very similar to Cisco Voice Call Toll Bypass mode, see Figure 10. Cisco provides a SIP VOIPgateway which allows an incoming call to bypass the circuit switched toll connection and sends thecall over the IP WAN as VOIP call. However, there are several critical issues with the Gatewayoption, as described in the next section.

Figure 10. Cisco Voice Toll Bypass Illustration

INCOMING CALL SCENARIOS

Figure 11 shows the Classic GSM roaming call forward model, where a call for the roaming UEis forwarded from the Home gateway MSC to the Visited MSC, where the Visited PLMN terminatesthe call to the roaming UE. The purpose of the Gateway option is to bypass this method oftermination, to avoid roaming and termination charges placed by the Visited PLMN on the HomePLMN. This bypass of the existing bilateral roaming agreement is a critical issue that will bediscussed in the next section. The commercial / politic issues of implementing such as bypass on therelationship between the Home and Visited PLMN can not be underestimated.

Figure 12, Figure 13 and Figure 14 show three options the incoming call scenario of the Gatewayoption. Without IMS, with IMS on the Visited PLMN and with IMS in both the Home and VisitedPLMN, respectively. The Gateway MSC forwards the incoming call to the roaming UE onto theVoIP bypass nework, where the call is routed at the far end by the local SIP VoIP Gateway andterminated at the visited network, see Figure 13.

SIP VOIP

GatewaySystem

SIP VOIP

GatewaySystem

Voice over IPnetwork

CS Corenetwork

CS CoreNetworkUTRAN

UTRAN

Setup ISUP IAM SIP INVITE

ISUP IAM Setup

ISUP ACMISUP ACM

Call Proceed

Call Proceed Alerting

Connect

Connect Ack

ISUP ANMSIP 200 OKISUP ANMConnect

Connect Ack

AUDIO LOGICAL CHANNELS

SIP 100 Trying

ACK

SIP 183 Session P rogress

SIP 180 RingingISUP Facility

ISUP FacilityAlerting


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Figure 11. Classic GSM Roaming Call Forward Call Model

Figure 12. Roaming call forward Gateway Centric (Option 1)


HPLMN CS Core VPLMN CS CoreIntl ISDN Transit

Network

B-pty on internationalroaming pays for theroaming call forward

leg of the incoming call

GMSC

GMSC

VMSC

HLR

Originating Country

A-pty pays for theoriginating call legto the HPLMN


HPLMN CS Core VPLMN CS Core

VOIP network


leg of the incoming callat lower tariff rate

GMSC

GMSC

VMSC

HLR

Originating Country


SIPVOIPGW

SIPVOIPGW

SIPserver

SIPserver

Legend:


SIP call session signalling path



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Figure 13. Roaming Call Forward (Option 2)

Figure 14. Roaming Call Forward (Option 3)


HPLMN CS Core VPLMN PS Core

VOIP network



GMSC

GMSC

IMS

HLR

Originating Country


SIPVOIPGW

SIPserver

SIPserver

Legend:



Circuit Switched Voice Path


HPLMN PS Core VPLMN PS Core

VOIP network

IMS IMS

HSS

SIPserver

SIPserver

Legend:



IMS Call Session Path




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ISSUES

How will VF's MPC DNS know when to return a VF's GGSN in Singapore andwhen to return a VF's GGSN in HPLMN? Would this be implemented only forcustomers subscribing to the roaming bypass service?

Quality of service for voice over PSD (Packet Switched Data): VOIP over GPRSpacket data connection is most likely to be delivered with best effort QoS (Qualityof Service). Voice quality will suffer. In addition, if the VOIP is transported overGRPS radio access bearer and ATM is used as the transport interface between theRAN/UTRAN and the packet data core, VOIP delivery over ATM transport is aninefficient mechanism, depending upon roaming data charges could become asignificant cost for this service.

Legal intercept: There will need to be specific provisions in the SIP Server tosupport this capability. HPLMN Law Enforcement Agency (LEA) may require realtime interception of call content. Issue will be how the intercepted call content betransmitted from this MGW to the LEA monitoring centre in real-time?

Incoming calls over PSD require IMS: One of the IMS capabilities is to enablemobile management of a registered UE. It maintains a SIP session between the UEand S-CSCF. When a SIP call arrives at a S-CSCF, it will be able to set up the call tothe UE over PS data connection. So in the Gateway option, without an explicit IMSdefinition, is this capability proprietary?

Public IP addresses leave the handset open to attack, it is assumed only private IPaddresses are assigned. When a roaming UE uses the HPLMN APN to set up aPDP context with the HPLMN GGSN deployed in the roaming country, thisGGSN can assign a public IP address to this roaming UE. Since the UE is given apublic IP address, this UE can be reached directly from the internet. In order toprotect this UE from attack, a firewall is required on the Gi interface between the

GGSN and the local SIP server. A SIP proxy server may also be required to bedeployed on the Gi interface between the GGSN and the local SIP server. The SIPproxy server would be required to be deployed in an internet DMZ configuration.

If the VPLMN has CAMEL inter-working arrangement with HPLMN such that asubscriber roaming on VPLMN is given access to value added voice services e.g.dialling short code to home voicemail, roaming bypass proposal will take away allthese value added voice services from the subscriber as the application of CAMELcall control at VPLMN will be bypassed. How would those services beimplemented?

HANDSET INTERCEPT OPTION

DESCRIPTION

This description is an informed guess as to the operation, further information is required.

When the customer turns on their mobile phone in Singapore it goes through a two phase set-up. In its roaming stage it registers with the VF home network then is able gather local MGW andMS-ISDN/IMSI details, the phone then registers as a SG phone. Or the UEs roaming table is pre-


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programmed with the MGW and MS-ISDN/IMSI details. This assumes a dedicated MS-ISDN/IMSI being held in each country for each subscribed UE, which would increase the costs forthis option.

The only reason to allocate a local IMSI is to enable the roaming UE to register with the HLR ofthe VPLMN instead of the HLR of the HPLMN. However, what happens to calls to the home

number destined for the HPLMN? Does the home number remain active on the UEs HomeHLR, and they are diverted onto the VoIP network? Or can the UE cheat two registrations on thesame phone?

For outgoing calls:

The handset client intercepts all international outgoing calls while roaming, anddiverts the call to the local SIP VoIP MGW

The Handset must signal to the SIP VoIP MGW the destination the number to becalled this can be done in two ways:

Two stage dialling. With this option, the UE dials a local SIP VOIP gateway

access number to initiate a circuit switched voice call to the SIP VOIP gateway. The UE than use DTMF signalling to communicate the original called partynumber to the SIP VOIP gateway which then sets up the call to this called partynumber.

Packet switched data connection to communicate the original called partynumber to the SIP VOIP gateway. In this case, the SIP VOIP gateway will needto receive the called party number on the packet data interface and to receivethe incoming voice call on the circuit switched interface.

For incoming calls

As the proposal is to allocate a local IMSI to the roaming subscriber such that thesubscriber is now registered with the HLR of the VPLMN, incoming call to thissubscriber will somehow have to be routed directly to the VPLMN bypassing theHPLMN.

One option: The Home network registers the customer is roaming and subscribedto the flat-rate roaming service, the call goes to the MGW (UK) and routed to theMGW(SG)

Calls originate from VFs MGW(SG) go to the local number allocated to thatroaming mobile phone

As far as M1 (Roaming Network) is concerned this is a free (CPP) call.


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Figure 15. Handset Centric: Circuit Switched Far End

Figure 16. Handset Centric: Packet Switched Far End

Home Country

Roaming Country

GRX /

InternetHPLMN CS Core VPLMN CS core

SIPVOIPGW

SIP

VOIPGW

SIPServer

Legend:

Circuit switched voice call to local SIP VOIP GW by dialling local SIP VOIP GW access number

VOIP media transported as RTP/UDP/IP between SIP VOIP Gateways

SIP call session signalling

Roaming UEVPLMN PS core

SGSNSGSNGGSN

Passing of called party number over PS data connection

SIP server sets up call to HPLMNSIP VOIP GW based on calledparty number received fromroaming UE

Roaming UE pass called party number either us PS dataconnection or use DTMF signalling in a two stage diallingprocess

Home Country

Roaming Country

GRX /

InternetHPLMN PS Core VPLMN CS core

IMSSIP

VOIPGW

SIPServer

Legend:





SGSNSGSNGGSN

Passing of called party number over PS data connection

SIP server sets up call to HPLMNSIP VOIP GW based on calledparty number received fromroaming UE

Roaming UE pass called party number either us PS dataconnection or use DTMF signalling in a two stage diallingprocess

Called UE with SIPclient


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Figure 17. Handset Centric: Incoming Calls

JAJAH Mobile Solution can enable a user to benefit from paying local call tariff rate forinternational calls made from a country. It does not provide the solution for receiving incoming calls when roaming abroad. When a user is roaming abroad, the recipient pays for the roaming call

forward leg of the call as the call is forwarded from HPLMN to the VPLMN. Effectively, therecipient pays an international call rate for receiving incoming calls.

Many roamers are not aware that they will be paying expensive rates for receiving calls whenabroad. It will be a shock to their system when they receive a phone bill from the HPLMN operatorat the end of their month. In order to avoid this problem, they can activate Baring of Incoming callswhen roaming abroad. The challenge is to find a solution that enables a user to pay nothing or to paylocal call rate for receiving incoming calls when roaming abroad. This is a really hard problem tosolve. With GSM mobility management, any call to a subscriber's MSISDN will always be routed tothe HPLMN first.

HPLMN is responsible for locating the subscriber and routes the call to the subscriber. If thesubscriber is currently roaming abroad, the HPLMN will have to route the call to the foreign PLMN. A caller locating in the same country that the roamer is currently visiting will have to payinternational call rates to roamer's home country and the HPLMN will charge the roamer forforwarding the call from HPLMN to his visiting country. This is the famous GSM trombone callrouting model even if the caller and roamer are standing next to each other. The only way I can thinkof to bypass this trombone call routing is for the roamer to be allocated a local MSISDN from a localPLMN.

Home Country

Roaming Country

GRX /

Internet

HPLMN CS Core

VPLMN CS core

SIPVOIPGW

SIPVOIPGW

SIPServer

Legend:





SGSNSGSNGGSN

(2) SIP server sets up call to SIPVOIP GW in visited country using

local visited number

Roaming UE pass called party number either us PS data

connection or use DTMF signalling in a two stage diallingprocess

(1) Roaming UEs number called through PSTN/ PLMN (Home or International origin), HLR set-up with forwarding to VoIP bypass network withlocal visited number maybe this SIP serverdoes the translation not HLR?

HLR/HSS


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The roamer then gives out this local MSISDN to friends in the same country he is visiting. Butthis means that his mobile phone will have to support dual IMSIs such that friends outside hisvisiting country will reach him by calling his HPLMN MSISDN and friends in his visiting countrywill reach him by calling his VPLMN MSISDN. This seems to be a very clumsy solution. If the usergoes abroad, why doesn't he simply buy a global GSM phone? After he has landed in his visitingcountry, just go to buy a local prepaid SIM which comes with a temporary local phone number. He

will then have to let his friends know how to reach him by calling this new phone number (sendthem a SMS perhaps), and let his friends pay the international rates to reach him. However, if theirfriends are clever enough, install the JAJAH Mobile client on their handsets and call his new localphone number to reach him.

ISSUES

Legal intercept: There will need to be specific provisions in the SIP VoIP GW tosupport this capability. HPLMN Law Enforcement Agency (LEA) may require realtime interception of call content. Issue will be how the intercepted call content betransmitted from this GW to the LEA monitoring centre in real-time?

Would the terms and conditions of the Foreign MS-ISDNs bought by the Homeoperator restrict such usage for such a service?

Assumes the VPLMN supports GPRS roaming. This is not the case for the JAJAHMobile Solution; hence the Handset options may use DTMF in preference to GPRSsignalling.

OTHER OPTIONS

DUAL MODE (GSM WIFI) HANDSETS

Dual-mode handsets: These devices can connect to both Wi-Fi and cellular networks, but the

functionality and technology varies widely from handset to handset. Types of dual-mode solutionsinclude:

UMA, GAN, ASNAP handsets: These consumer devices allow roaming betweenWLANs and cellular, and are unable to connect to a business IP PBX.

Duct tape handsets: These devices include a Wi-Fi SIP phone and a cellularphone in one device, but there is no integration between the two. From a Wi-Fihotspot, the phone can function as an extension to an IP PBX. Otherwise, itfunctions as a normal cell phone.

Integrated dual-mode handsets: These allow users to be constantly connected to the

company VoIP system using Wi-Fi or cellular, with some solutions allowingseamless roaming. The main problem with this technology is that there are noindustry standards for call routing or handoff, leading to multiple proprietaryofferings. The Motorola solution includes the CN620 handset, Wireless ServicesManager and modified 802.11a access points. Alcatel has a solution for serviceproviders called Intelligent Mobile Redirect (IMR). At least four different groupsare working to establish standards (e.g., SCCAN, IEEE 802.21, Wireless WirelineConvergence Working Group and Mobile IGNITE).


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Technology Strengths

Easy integration with existing VoIP and Wi-Fi infrastructures: Wi-Fi handsets willwork with standard Wi-Fi networks and SIP VoIP systems. For initial functionality,it requires no investment beyond the handset itself.

Enables remote VoIP network access: This allows remote and mobile employees toconnect to the company VoIP network. It is much more convenient and portablethan a laptop or PDA-based softphone. Dual-mode phones can extend coveragebeyond hotspots to anywhere a cellular connection is available.

Converged cellular and VoIP: Dual-mode handsets merge cellular and VoIPtechnology in a single device, abstracting the service from the access technology.

Call savings by using the least expensive connection: Dual-mode phones canautomatically select the least expensive network for calls. Calls placed over Wi-Ficost the same as using the company VoIP network.

Technology Challenges

WLAN-cellular redirection and roaming: The lack of industry standards has led tomultiple proprietary implementations. There are no clear answers as to whatinfrastructure is required, what type of service from providers is needed and how itwould be billed. Motorola is the only vendor with an available, working solution, butit favours the enterprise more than SMBs.

Quality of service: Current wireless networks were not designed to handle voicedata. For optimal quality, voice traffic should be prioritized. Roaming betweenaccess points is an issue as call quality will dip due to the current handoff speed,which is too slow for voice. The upcoming 802.11e standard will bring QoS to Wi-

Fi networks and 802.11r will add fast roaming to address these issues.

Reduced battery life: Wi-Fi radios consume more power than cellular radios, leadingto reduced battery life or bulkier batteries. This will become less of an issue as Wi-Fichipset manufacturers continue to reduce power consumption and new Wi-Fistandards emerge for better power management.

Increased Wi-Fi bandwidth utilization: The majority of Wi-Fi networks still use theslower 802.11b standard, which may not have enough bandwidth to handleadditional traffic. Faster standards such as 802.11g and the upcoming 802.11nstandard can alleviate this bottleneck.

VoIP features not available on cellular networks: Only basic voice functionality isavailable when on a cellular network because users will not have full internetconnectivity to the IP PBX. Forthcoming wireless broadband services promise tochange this, providing an IP connection with the coverage of cellular.

VoIP and Wi-Fi security: VoIP and Wi-Fi attacks will increase significantly in thenear future, possibly disrupting communications, as vulnerabilities surface and are


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exploited. Security will become more of an issue as the popularity of these networksgrows, attracting the interest of hackers.

Handset costs may offset call savings: Companies must bear the full cost of a Wi-Fihandset, while cellular phones are carrier subsidized. This can reduce or eliminateany cost savings from using VoIP instead of cellular.

MOBILE CENTRIC START-UPS: JAJAH AND REBTEL

JAJAH

At the DEMO conference in San Diego (September 27th 2006), JAJAH unveiled their MobileSuite, that allows consumers to make free long-distance and global calls directly from their mobilephones. Appendix 5 includes the current calling plans.

The service is the latest step in JAJAHs stated mission to become the first true globalcommunications company by providing consumers with the smartest, cheapest and most innovativecommunication services around the world. Their view is service providers must be global in scope to

achieve economies of scale and price points unobtainable by locally focused access providers.

JAJAH it offers two methods to initiate its calls:

Method 1: User goes onto their website and enters his mobile phone number andthe called mobile phone number. The JAJAH website then invokes their system tomake dial out to the user and the called number. Once the call is answered by bothparties, the JAJAH system connects the two call legs to together.

Method 2: To provide a Symbian OS based Java client to be installed on the usershandset. Once the JAJAH client is installed, the client will ask the user to nominatehis home country. The JAJAH client uses this information to select the local PSTN VoIP gateway number to use. When the user dials an international number, the

JAJAH client intercepts this telephony request and makes a circuit switched voicecall to the local PSTN VOIP gateway. Once the call is connected to the PSTNVOIP Gateway, there are two possible methods for the JAJAH client to pass theoriginal called party number to the local PSTN VOIP Gateway. One method is tosend the original called party number by SMS to the PSTN VOIP Gateway. Analternative method is to send the original called party number by using DTMF digitstransferred over the established circuit switched voice connection. When the PSTN VOIP gateway receives the incoming call, it waits for the original called partynumber to be sent by the client by DTMF. The PSTN VOIP Gateway simplyperforms circuit switched voice TDM conversion to SIP VOIP connection. One ofthe most common PSTN VOIP Gateway in use is Cisco Access Server (e.g. CiscoAS53500, AS5350, AS5400). See Figure 18 and Figure 19.

The JAJAH Mobile client will first ask the user from which country they wish to call from. Oncethe user has selected the country from a list of countries presented by the client GUI, the JAJAHmobile client will simply use the local VOIP Gateway number of this country for dialling. When theuser decides to make an international call, the JAJAH mobile client will intercept this call request. Itreplaces the original called party number by the local VOIP Gateway number and sets up a voice callto this Gateway. Once the call is connected, the JAJAH Mobile Client will use DTMF to send the


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original called party number to the VOIP Gateway. When the VOIP Gateway receives the originalcalled party number, the VOIP Gateway will then initiate a call to the destination over the global VOIP network. If the user subsequently decides to visit another country, and wish to make aninternational call from this country. The user will have to activate his JAJAH Mobile client and selecta new country from which they wish to make international calls. Once the user has selected the newcountry, the JAJAH Mobile client will use the local VOIP Gateway of current visiting country to dial

out.

The JAJAH Mobile Client will be able to distinguish whether the user is dialling an internationalcall or a local call from the fact that GSM dictates that international call must be prefixed by the +sign. If the user dials a local number when roaming, the JAJAH Mobile client will not intercept thiscall and let the handset makes the call to the dialled local number. JAJAH Mobile Solution does notrequire any co-operation from a mobile operator. A mobile subscriber owning a Symbian handsetcan simply navigate to JAJAH website and download the application onto their handset and they canuse the application straight away whether from home country or when roaming abroad. The onlylimitation is that the user is limited to those countries where JAJAH has a local PSTN VOIPGateway deployed.

In fact, the user can even benefit from accessing his HPLMN voicemail when roaming abroad.For example, he accesses his voicemail by dialling a short code when registered on HPLMN. Whenthe user visits another country and registered on a roaming partner's PLMN, there are two ways forthe user to access his voicemail. If the roaming partner has CAMEL inter-working with his HPLMNoperator, he can dial the normal voicemail short code, and the VPLMN will route the call to theHPLMN voicemail system. Alternatively, the user can dial the long access number of his voicemailbox. He will be able to find out this number from his HPLMN website or just by calling theservice desk of his HPLMN operator. When he dials the long access number, he is effectively makinga long distance call back to his home country. If he chooses to dial his voicemail short code(assuming the VPLMN has CAMEL inter-working arrangement with HPLMN), he will still becharged international call tariff rate.

If the user has the JAJAH Mobile client installed, the user can dial the long voicemail access

number. The JAJAH Mobile client will intercept this call as described earlier and diverts the call tothe local VOIP Gateway. The call is then routed to the HPLMN's voicemail system via Rajahs globalVOIP network. When the call is delivered by Jajahs global VOIP network to the home country'sVOIP Gateway, the VOIP Gateway can set up a local circuit switched voice call to the HPLMN 'svoicemail system. Since the HPLMN sees the call is originated from the home country, the HPLMNcan only charge a termination rate corresponding to a call originated from a local PSTN.

As a result, the JAJAH Mobile Solution is a significant threat to all mobile operators who will seetheir international voice and voicemail revenue cannibalised by the VOIP roaming bypass solution.The problem is that there is not much a mobile operator can do about this as the solution does notdemand any co-operation from the mobile operators.

Roman Scharf and Daniel Mattes founded JAJAH in 2005. JAJAH has offices in Mountain View,CA and Luxembourg.


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Figure 18. JAJAH Call Flow

Figure 19. JAJAH Signalling

Home Country Called Country

OriginatingHPLMN

Terminating

PLMN

GGSN

SIPVOIP

GW

SIPVOIP

GW

SIPserver

SIPserver

Legend:


SIPVOIPGW

SIPVOIPGW

Global VOIP network

Terminating

PSTN

Calling party A

Called Party B

Called Party CSIP signalling


Calling party pays local call rate forinternational calls to called party B and C.Jajah client intercepts the call request anddirects the call to a local PSTN VOIPgateway,

Jajah mobile application client based solution call signalling illustration

MS A Orig.PLMN

OrigVOIP

GW

Term.

VOIP

GW

TermPLMN

MS BSIP

Proxy

SetupIAM

ACM

Proceed

ANMConnect

CS Voice call to local VOIP GW established

Pass original called party number asDTMF digits to local VOIP GW

INVITEINVITE IAM

Setup

ACM100 Trying100 TryingProceed

AlertingCPG183 SessionProgress

183 Session

Progress ConnectANM200 OK

200 OK

ACK ACK

Established end to end voice connection


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REBTEL

To get started, people sign up for a Rebtel account at www.rebtel.com, where they enter theirmobile phone number and the mobile phone numbers of global friends. Rebtel then instantly createspairs of local numbers and sends them in text messages (sms) so they can be saved in the friendsphone address books and used to call each other from then on.

For example, a person in San Francisco gets a local San Francisco number for calling a friend inLondon, and their friend in London gets a local London number for reaching them in San Francisco.The local calls are connected using Voice over Internet Protocol (VoIP) technology.

Once set up, Rebtel charges $1 USD per week for use of two services: REBin and REBout.

With REBout, people use local numbers where they live to call anywhere in the world and onlypay for the local call, plus a small per-minute fee to Rebtel, from 2c per min.

Rebtel also offers a REBin, where the users local call is connected with their global friends localcall in a virtual room on the Internet, called a REBroom. In the REBroom all calls are free nomatter how many, how often or how long. No additional charges over the $1 USD per week fee and

the cost of the local calls that most consumers have already paid for with their mobile carrier.

To get to the REBroom, when friends phone, instead of answering, the user simply hangs up,and while the friend hangs on, the user calls the friends local number. The two calls are thenautomatically connected, and the friends can hang out and talk for as long as they like, and not worryabout the cost.

With Rebtel, consumers can use the mobile phones they own today, and dont have to buyanything else, download software, get a new SIM card, use a headset connected to a computer, orworry about confusing additional charges.

Consumers need to sign up to create local numbers, but to call a local number created by a

Rebtel user, no sign-up is necessary. And, users will only be charged Rebtels $1 USD per weekservice fee if they actually make calls. If no calls are made during a week there are no charges.

Current country coverage: Argentina, Australia, Belgium, Brazil, Bulgaria, Canada, Chile, Cyprus,Czech Republic, Denmark, Estonia, Finland, France, Germany, Hungary, Ireland, Israel, Italy, Japan,Latvia, Lithuania, Luxembourg, Mexico, Netherlands, New Zealand, Norway, Peru, Poland, Portugal,Romania, Singapore, Spain, Sweden, Switzerland, Turkey, United Kingdom, and United States.

Rebtel was founded in 2005, backed by Index Ventures and Benchmark Capital (backers ofSkype), it recently raised $20m in September 2006. Given the additional configuration, and theadditional cost compared to JAJAH, it is the authors opinion that REBTEL will need to modify itsproposition to survive.

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