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ALAN QUAYLE BUSINESS AND SERVICE

DEVELOPMENT

MARKET ASSESSMENT
O F VO I P B Y PA S S
ROA M I N G A N D
O P E R AT O R I M PAC T /
P RO P O S I T I O N
EVALUATING 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

© Alan Quayle Business and Service Development. All Rights Reserved.


CONTENTS

ABSTRACT 6

INTRODUCTION AND BACKGROUND 7

PURPOSE 7
ROAMING DEFINITIONS AND TRANSFER ACCOUNT PROCEDURE (TAP) 10
ROAMING DEFINITIONS 10
TRANSFER ACCOUNT PROCEDURE 10
TAP3 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 13
CALLING SCENARIOS: WHEN ABROAD AND “CALLING HOME” 13
CALLING SCENARIOS: WHEN ABROAD AND BEING CALLED FROM HOME 14

VOIP ROAMING BYPASS TECHNOLOGY DESCRIPTIONS 16

GATEWAY OPTION 16
DESCRIPTION 16
OUTGOING CALL SCENARIOS 16
INCOMING CALL SCENARIOS 18
ISSUES 21
HANDSET INTERCEPT OPTION 21
DESCRIPTION 21
ISSUES 25
OTHER OPTIONS 25
DUAL MODE (GSM WIFI) HANDSETS 25
MOBILE CENTRIC START-UPS: JAJAH AND REBTEL 27
JAJAH 27
REBTEL 30
INTERNET 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.

© 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.

© Alan Quayle Business and Service Development. All Rights Reserved


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 REGULATION ERROR! 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.

© Alan Quayle Business and Service Development, All Rights Reserved


TABLE OF FIGURES
Figure 1. International Calling from the Home Network Case _________________________________ 9
Figure 2. International Roaming Case ____________________________________________________ 9
Figure 3. Transferred Account Procedure (TAP) – GSM Billing and Accounting ________________ 11
Figure 4. The collection and transfer of TAP information between the Home Public Land Mobile
Network and the Visited Public Land Mobile Network _______________________________________ 12
Figure 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 _______________________________________________ 13
Figure 12 When Abroad and being Called from Home ______________________________________ 14
Figure 13. Classic GPRS Roaming ______________________________________________________ 16
Figure 14. Option 1A, Gateway Centric: Circuit Switch far-end _______________________________ 17
Figure 15. Option 1B, Gateway Centric: Packet Switch far-end _______________________________ 17
Figure 16. Cisco Voice Toll Bypass Illustration ____________________________________________ 18
Figure 17. Classic GSM Roaming Call Forward Call Model __________________________________ 19
Figure 18. Roaming call forward – Gateway Centric (Option 1) _______________________________ 19
Figure 19. Roaming Call Forward (Option 2)______________________________________________ 20
Figure 20. Roaming Call Forward (Option 3)______________________________________________ 20
Figure 21. Handset Centric: Circuit Switched Far End _____________________________________ 23
Figure 22. Handset Centric: Packet Switched Far End ______________________________________ 23
Figure 23. Handset Centric: Incoming Calls ______________________________________________ 24
Figure 24. JAJAH Call Flow ___________________________________________________________ 29
Figure 25. JAJAH Signalling __________________________________________________________ 29
Figure 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.

© Alan Quayle Business and Service Development. All Rights Reserved


ABSTRACT

There are emerging technologies that use functionality in the mobile phone to extend VoIP from
being employed by the HPLMN to offer cheap international calls tariff rate to home subscribers. To
now 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 technologies
are addressing the international roaming market, at $15B USD global business, and currently one of
the mobile industries most lucrative revenue stream.

© Alan Quayle Business and Service Development, All Rights Reserved


I N T RO D U C T I O N A N D B A C K G RO U N D

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 bypass
international call charges, or mobile network charges. Generally these services target a segment of
international calling / mobile customers. For example, international calling cards are used
extensively by foreign workers to call home. PC based solutions enable friends or small international
businesses to hold calls at free / low cost.

Both JAJAH and REBTEL are focused on the mobile communication segment for international
calling from the user’s home network, see Figure 1. Breaking free of the PC and the international
calling card models. However, the JAJAH client solution could also be extended to the international
roaming 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 is
asked to provide their mobile phone number and the mobile phone numbers of the
friends they want to call. The Rebtel system then generates a local number for the
user and a local number for each of his friends. Once the local numbers are
generated, the Rebtel website sends a SMS and/or email to the user and to their
friends. 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 his
HPLMN to a local PSTN VoIP gateway. The VoIP Gateway will use his calling line
ID to identify the user and the called party number to connect the call to his friend
over a VoIP network. When his friend answers the call. The user asks his friend to
hang 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 the
Rebtel’s local PSTN VoIP Gateway. The Rebtel system then joins the two local calls
together over a VOIP network. Rebtel has simply employed SMS and email as the
bearer to inform the user the local phone number to dial to connect to Rebtel’s local
PSTN VoIP Gateway.

For JAJAH it offers two methods.

Method 1: User goes onto their website and enters their mobile phone number and
the called mobile phone number. The JAJAH website then invokes their system to
make dial out to the user and the called number. Once the call is answered by both
parties, 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 user’s


handset. Once the JAJAH client is installed, the client will ask the user to nominate
his 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 voice
call to the local PSTN VOIP gateway. Once the call is connected to the PSTN
VOIP Gateway, there are two possible methods for the JAJAH client to pass the

© Alan Quayle Business and Service Development. All Rights Reserved


original called party number to the local PSTN VOIP Gateway. One method is to
send the original called party number by SMS to the PSTN VOIP Gateway. An
alternative method is to send the original called party number by using DTMF digits
transferred over the established circuit switched voice connection. When the PSTN
VOIP gateway receives the incoming call, it waits for the original called party
number to be sent by the client by DTMF. The PSTN VOIP Gateway simply
performs circuit switched voice TDM conversion to SIP VOIP connection. One of
the most common PSTN VOIP Gateway in use is Cisco Access Server (e.g. Cisco
AS53500, AS5350, AS5400).

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

An example of the Gateway Option, Operator sites in Singapore a GGSN, noted as VF’s 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 its
mobile packet core DNS to resolve the APN to a GGSN IP address. Hence VoIP over GPRS to the
local GGSN bypassing the VPLMN for incoming and outgoing calls.

An example of the Handset options is when the customer turns on their mobile phone in
Singapore it goes through a two phase set-up. It registers with the VF home network then is able
gather local MGW and MS-ISDN/IMSI details, the phone then registers as a SG phone. Or possibly
the UE’s roaming table is pre-programmed with the MGW and MS-ISDN/IMSI details. So the
roaming UE masquerades as a local phone for incoming calls and uses the same procedure as JAJAH
for 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 American
regions;

Review the international roaming market, and assessing the impact these
technologies present;

Present the findings from a market survey of key operators from APAC, EMEA
and 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 technology’s impact upon Operator; and

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


technology.

© Alan Quayle Business and Service Development, All Rights Reserved


Called Country C
Operator’s subscribers pay local call tariff PLMN
rate for international calls by routing
international calls via bypass VOIP
network
Called UE of
Country C

Bypass VOIP network

PLMN Called UE of
HPLMN
Calling UE of Country B
Country A Called Country B
Home Country A

Figure 1. International Calling from the Home Network Case

Called Country C
Operator subscribers pay local call tariff Other PLMNs
rate for international calls to home country
or to other countries by routing
international calls made from VPLMN via
bypass VOIP network

Bypass VOIP network

VPLMN Roaming
HPLMN
UE
Roaming Country B
Home Country A

Figure 2. International Roaming Case

© Alan Quayle Business and Service Development. All Rights Reserved


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 another


country. The called person may be a subscriber on the same home mobile network
as the 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 is
visiting.

TRANSFER ACCOUNT PROCEDURE

GSM is founded on the concept of roaming - allowing customers from other networks and
countries 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 20,000
individual 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 information
about 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 regulatory
influences result in a complex ever changing “patchwork quilt” of termination changes. For
example, in the UK VoIP call charges to a mobile phone are generally 10 times the charge to a fixed
line, see Error! Reference source not found.. While say in Singapore, there is no difference in
VoIP call charges to fixed or mobile.

Within the GSMA the Transferred Account Data Interchange Group (TADIG) is responsible
for defining the interchange of billing data between different network operators by defining and
implementing 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 roaming
billing information. This is how roaming partners are able to bill each other for the use of networks
and services through a standard process. Much of the traffic carried by a GSM Public Mobile
Network (PMN) either originates, or terminates in another network. The operator of the local fixed
network charges the wireless operator for each call that terminates at one of its fixed subscribers.

© Alan Quayle Business and Service Development, All Rights Reserved


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

Therefore GSM network operators and their local fixed counterparts usually negotiate an
interconnect agreement to make charging as simple as possible. The other fixed international
operators 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 not
necessary for the German PMN operator to negotiate a price with a Canadian fixed network
operator.

The German PMN operator negotiates a price with the German fixed network operator. The
German 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 means
that the German PMN has to recoup the cost of the call from its subscribers either directly (retail
billing), or via the appropriate Service Provider (wholesale billing). This form of inter-administration
accounting 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 German
network. The German fixed network will still charge the German PMN for the leg of the call placed
to the Canadian number. In this case, the German PMN does not receive any revenue from its own
subscriber. In order to recoup the costs incurred by the call, the German PMN must charge the
home mobile network operator, here the French PMN, to cover the costs incurred by the French
mobile subscriber.

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

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

© Alan Quayle Business and Service Development. All Rights Reserved


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

The call records produced by the MSC are transferred on a regular basis to the billing system of
the VPLMN for pricing or rating. Those call records produced on behalf of roaming subscribers, will
be 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 that
operators can send 1 or many TAP files per day. TAP files contain rated call information according
to the operator's Inter Operator Tariff (IOT), plus any bilaterally agreed arrangements or discounting
schemes.

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

On reception by the HPMN, the TAP record is converted into an internal format and added
together 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 wholesale
billing between the HPMN and that Service Provider, an example is for a HPLMN operator to
exchange TAP3 files with its MVNOs to settle wholesale billing agreements between an operator and
its MVNOs. On receipt of the information from the HPMN, the Service Provider may re-rate the
calls according to its own tariff plans and produce an itemized bill, including call detail, for the
subscriber.

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

© Alan Quayle Business and Service Development, All Rights Reserved


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), see
Figure 5. The host operator passes the call via 'international transit' to the home operator. The
home 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

Charge Step 3

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 between
2-10c (Euro) – for Tier 1 countries. Note figure dependent upon volume and
route - not published as highly competitive B2B business

Call Termination Fee

© Alan Quayle Business and Service Development. All Rights Reserved


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 not
found..

Home Operator Mark-up

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 routes
the call initially to your home operator (which may or may not be the same). Your home operator
forwards 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 network
and establishes your friend’s originated call.

Cost Components:

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

Steps 2 and 3 Your home operator will charge you a tariff which includes inter alia
the international transit fees to forward the call to you in the destination country
and 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

© Alan Quayle Business and Service Development, All Rights Reserved


International Transit Fee

Due to the competition that exists in inter-country transport this fee is between
2-10c (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 higher
termination fees

Home Operator Mark-up

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

© Alan Quayle Business and Service Development. All Rights Reserved


VO I P ROA M I N G B Y PA S S T E C H N O L O G Y D E S C R I P T I O N S

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

GATEWAY OPTION

DESCRIPTION

Figure 7 shows the usual GPRS (General Packet Radio Service) roaming case. Let’s take a real-
world case and assume that Operator is the home operator, and a VF customer is roaming in
Singapore 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 mobile
packet 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.

HPLMN PS Core GRX VPLMN PS Core


Home
Portal
GGSN SGSN

Home Country Roaming Country

UE established a PDP
context from VPLMN to
HPLMN GGSN using
Home APN

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 VF’s 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 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 (Mobile
Packet Core) DNS forwards the request to VF's MPC DNS for APN resolution. The VF MPC DNS
will resolve this APN to a VF GGSN deployed in Singapore and return the resolved IP address to
the M1's MPC DNS. M1 MPC DNS then returns the GGSN IP address to the M1 SGSN. The M1
SGSN then sets up a PDP context to the VF GGSN deployed in Singapore.

© Alan Quayle Business and Service Development, All Rights Reserved


A SIP call session is set up between
the roaming UE and the SIP server
which functions as a SIP Redirect
server
SIP
server

VPLMN PS Core
SIP
VOIP GRX /
HPLMN CS Core GW GGSN
SGSN
Internet

Home Country Roaming Country


Home operator has
GGSN and SIP server
Legend: Roaming UE
in roaming countries
with SIP client
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 Operator’s SIP server

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

A SIP call session is set up


between the roaming UE and
the SIP server which functions
as a SIP Redirect server
SIP
server

VPLMN PS Core
IMS GRX /
HPLMN PS Core GGSN
SGSN
Internet

Home Country Roaming Country


Home operator has
Called UE with SIP GGSN and SIP server
client Roaming UE
in roaming countries with SIP client

Legend:

VOIP media transported as RTP/UDP/IP


SIP call session signalling path over GPRS packet data connection

Logical SIP call session association between roaming UE and HPLMN Operator’s SIP server

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

© Alan Quayle Business and Service Development. All Rights Reserved


With the PDP context established to VF’s GGSN (SG), Operator also has within the same
private IP space a SIP server. As they are both on the same private IP space, the MGW can directly
address the roaming mobile. This avoids the need to be part of M1’s DMZ – as M1 maybe unlikely
to 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 VOIP
gateway which allows an incoming call to bypass the circuit switched toll connection and sends the
call over the IP WAN as VOIP call. However, there are several critical issues with the Gateway
option, as described in the next section.

SIP VOIP Voice over IP SIP VOIP


CS Core CS Core
UTRAN Gateway network Gateway UTRAN
network Network
System System

Setup ISUP IAM SIP INVITE


SIP 100 Trying ISUP IAM Setup
Call Proceed
SIP 183 Session Progress ISUP ACM
ISUP ACM
Call Proceed Alerting
ISUP Facility
ISUP Facility SIP 180 Ringing
Alerting
ISUP ANM Connect
SIP 200 OK
Connect ISUP ANM
Connect Ack
Connect Ack ACK

AUDIO LOGICAL CHANNELS

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 UE
is forwarded from the Home gateway MSC to the Visited MSC, where the Visited PLMN terminates
the call to the roaming UE. The purpose of the Gateway option is to bypass this method of
termination, to avoid roaming and termination charges placed by the Visited PLMN on the Home
PLMN. This bypass of the existing bilateral roaming agreement is a critical issue that will be
discussed in the next section. The commercial / politic issues of implementing such as bypass on the
relationship 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 Gateway
option. Without IMS, with IMS on the Visited PLMN and with IMS in both the Home and Visited
PLMN, respectively. The Gateway MSC forwards the incoming call to the roaming UE onto the
VoIP bypass nework, where the call is routed at the far end by the local SIP VoIP Gateway and
terminated at the visited network, see Figure 13.

© Alan Quayle Business and Service Development, All Rights Reserved


HPLMN CS Core Intl ISDN Transit VPLMN CS Core
HLR Network
GMSC VMSC

Home Country Roaming Country

B-pty on international
A-pty pays for the
roaming pays for the
GMSC originating call leg
roaming call forward
to the HPLMN
leg of the incoming call

Originating Country

Figure 11. Classic GSM Roaming Call Forward Call Model

SIP
server

HPLMN CS Core VPLMN CS Core


HLR SIP SIP
GMSC VOIP VOIP VMSC
GW GW
VOIP network
Home Country Roaming Country

B-pty on international
A-pty pays for the
roaming pays for the
GMSC originating call leg
roaming call forward
to the HPLMN
leg of the incoming call
at lower tariff rate

Originating Country
Legend:

VOIP media transported as RTP/UDP/IP


SIP call session signalling path

Circuit switched voice call

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

© Alan Quayle Business and Service Development. All Rights Reserved


SIP
server

HPLMN CS Core VPLMN PS Core


HLR SIP
GMSC VOIP IMS
GW
VOIP network
Home Country Roaming Country

B-pty on international
A-pty pays for the
roaming pays for the
GMSC originating call leg
roaming call forward
to the HPLMN
leg of the incoming call
at lower tariff rate

Originating Country
Legend:

VOIP media transported as RTP/UDP/IP


SIP call session signalling path

Circuit Switched Voice Path

Figure 13. Roaming Call Forward (Option 2)

SIP
server

HPLMN PS Core VPLMN PS Core


HSS
IMS IMS

VOIP network
Home Country Roaming Country

B-pty on international
roaming pays for the
roaming call forward
Legend: leg of the incoming call
at lower tariff rate
VOIP media transported as RTP/UDP/IP
SIP call session signalling path

IMS Call Session Path

Figure 14. Roaming Call Forward (Option 3)

© Alan Quayle Business and Service Development, All Rights Reserved


ISSUES

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

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

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

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

Public IP addresses leave the handset open to attack, it is assumed only private IP
addresses are assigned. When a roaming UE uses the HPLMN APN to set up a
PDP context with the HPLMN GGSN deployed in the roaming country, this
GGSN can assign a public IP address to this roaming UE. Since the UE is given a
public IP address, this UE can be reached directly from the internet. In order to
protect 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 be
deployed on the Gi interface between the GGSN and the local SIP server. The SIP
proxy server would be required to be deployed in an internet DMZ configuration.

If the VPLMN has CAMEL inter-working arrangement with HPLMN such that a
subscriber 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 all
these value added voice services from the subscriber as the application of CAMEL
call control at VPLMN will be bypassed. How would those services be
implemented?

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 and
MS-ISDN/IMSI details, the phone then registers as a SG phone. Or the UE’s roaming table is pre-

© Alan Quayle Business and Service Development. All Rights Reserved


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 for
this option.

The only reason to allocate a local IMSI is to enable the roaming UE to register with the HLR of
the 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 UE’s Home
HLR, and they are diverted onto the VoIP network? Or can the UE cheat two registrations on the
same phone?

For outgoing calls:

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

The Handset must signal to the SIP VoIP MGW the destination the number to be
called 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 party
number to the SIP VOIP gateway which then sets up the call to this called party
number.

Packet switched data connection to communicate the original called party


number to the SIP VOIP gateway. In this case, the SIP VOIP gateway will need
to receive the called party number on the packet data interface and to receive
the 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 the
subscriber is now registered with the HLR of the VPLMN, incoming call to this
subscriber will somehow have to be routed directly to the VPLMN bypassing the
HPLMN.

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

Calls originate from VF’s MGW(SG) go to the local number allocated to that
roaming mobile phone

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

© Alan Quayle Business and Service Development, All Rights Reserved


SIP server sets up call to HPLMN
SIP VOIP GW based on called
party number received from
roaming UE
SIP
Server Roaming Country

SIP
VOIP GRX / SIP
VOIP
HPLMN CS Core GW GW VPLMN CS core
Internet

Home Country
GGSN SGSN
Roaming UE pass called party number either us PS data
connection or use DTMF signalling in a two stage dialling Roaming UE
process VPLMN PS core

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


Passing of called party number over PS data connection

Figure 15. Handset Centric: Circuit Switched Far End

SIP server sets up call to HPLMN


Roaming UE pass called party number either us PS data SIP VOIP GW based on called
connection or use DTMF signalling in a two stage dialling party number received from
process roaming UE
SIP
Server Roaming Country

IMS GRX / SIP


VOIP
HPLMN PS Core GW VPLMN CS core
Internet

Home Country
Called UE with SIP GGSN SGSN
client
Roaming UE
VPLMN PS core

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

Passing of called party number over PS data connection

Figure 16. Handset Centric: Packet Switched Far End

© Alan Quayle Business and Service Development. All Rights Reserved


(1) Roaming UE’s number called through PSTN
/ PLMN (Home or International origin), HLR set-
up with forwarding to VoIP bypass network with (2) SIP server sets up call to SIP
‘local visited number’ – maybe this SIP server VOIP GW in visited country using
‘local visited number’
does the translation not HLR?
SIP
Server Roaming Country

HPLMN CS Core
SIP
VOIP GRX / SIP
VOIP
GW GW VPLMN CS core
Internet
HLR/
HSS

Home Country

Roaming UE pass called party number either us PS data GGSN SGSN


connection or use DTMF signalling in a two stage dialling Roaming UE
process VPLMN PS core

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

Figure 17. Handset Centric: Incoming Calls

JAJAH Mobile Solution can enable a user to benefit from paying local call tariff rate for
international 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, the
recipient pays an international call rate for receiving incoming calls.

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

HPLMN is responsible for locating the subscriber and routes the call to the subscriber. If the
subscriber 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 pay
international call rates to roamer's home country and the HPLMN will charge the roamer for
forwarding the call from HPLMN to his visiting country. This is the famous GSM trombone call
routing model even if the caller and roamer are standing next to each other. The only way I can think
of to bypass this trombone call routing is for the roamer to be allocated a local MSISDN from a local
PLMN.

© Alan Quayle Business and Service Development, All Rights Reserved


The roamer then gives out this local MSISDN to friends in the same country he is visiting. But
this means that his mobile phone will have to support dual IMSIs such that friends outside his
visiting country will reach him by calling his HPLMN MSISDN and friends in his visiting country
will reach him by calling his VPLMN MSISDN. This seems to be a very clumsy solution. If the user
goes abroad, why doesn't he simply buy a global GSM phone? After he has landed in his visiting
country, 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 (send
them a SMS perhaps), and let his friends pay the international rates to reach him. However, if their
friends are clever enough, install the JAJAH Mobile client on their handsets and call his new local
phone number to reach him.

ISSUES

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

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

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

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 solutions
include:

UMA, GAN, ASNAP handsets: These consumer devices allow roaming between
WLANs 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 cellular
phone in one device, but there is no integration between the two. From a Wi-Fi
hotspot, the phone can function as an extension to an IP PBX. Otherwise, it
functions 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 allowing
seamless roaming. The main problem with this technology is that there are no
industry standards for call routing or handoff, leading to multiple proprietary
offerings. The Motorola solution includes the CN620 handset, Wireless Services
Manager and modified 802.11a access points. Alcatel has a solution for service
providers called Intelligent Mobile Redirect (IMR). At least four different groups
are working to establish standards (e.g., SCCAN, IEEE 802.21, Wireless Wireline
Convergence Working Group and Mobile IGNITE).

© Alan Quayle Business and Service Development. All Rights Reserved


Technology Strengths

Easy integration with existing VoIP and Wi-Fi infrastructures: Wi-Fi handsets will
work 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 to
connect to the company VoIP network. It is much more convenient and portable
than a laptop or PDA-based softphone. Dual-mode phones can extend coverage
beyond hotspots to anywhere a cellular connection is available.

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

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

Technology Challenges

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

Quality of service: Current wireless networks were not designed to handle voice
data. For optimal quality, voice traffic should be prioritized. Roaming between
access 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, leading
to reduced battery life or bulkier batteries. This will become less of an issue as Wi-Fi
chipset manufacturers continue to reduce power consumption and new Wi-Fi
standards emerge for better power management.

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

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

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

© Alan Quayle Business and Service Development, All Rights Reserved


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

Handset costs may offset call savings: Companies must bear the full cost of a Wi-Fi
handset, while cellular phones are carrier subsidized. This can reduce or eliminate
any 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 Mobile
Suite, that allows consumers to make free long-distance and global calls directly from their mobile
phones. Appendix 5 includes the current calling plans.

The service is the latest step in JAJAH’s stated mission to become the first true global
communications company by providing consumers with the smartest, cheapest and most innovative
communication 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 and
the called mobile phone number. The JAJAH website then invokes their system to
make dial out to the user and the called number. Once the call is answered by both
parties, 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 user’s


handset. Once the JAJAH client is installed, the client will ask the user to nominate
his 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 voice
call to the local PSTN VOIP gateway. Once the call is connected to the PSTN
VOIP Gateway, there are two possible methods for the JAJAH client to pass the
original called party number to the local PSTN VOIP Gateway. One method is to
send the original called party number by SMS to the PSTN VOIP Gateway. An
alternative method is to send the original called party number by using DTMF digits
transferred over the established circuit switched voice connection. When the PSTN
VOIP gateway receives the incoming call, it waits for the original called party
number to be sent by the client by DTMF. The PSTN VOIP Gateway simply
performs circuit switched voice TDM conversion to SIP VOIP connection. One of
the most common PSTN VOIP Gateway in use is Cisco Access Server (e.g. Cisco
AS53500, 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. Once
the user has selected the country from a list of countries presented by the client GUI, the JAJAH
mobile client will simply use the local VOIP Gateway number of this country for dialling. When the
user decides to make an international call, the JAJAH mobile client will intercept this call request. It
replaces the original called party number by the local VOIP Gateway number and sets up a voice call
to this Gateway. Once the call is connected, the JAJAH Mobile Client will use DTMF to send the

© Alan Quayle Business and Service Development. All Rights Reserved


original called party number to the VOIP Gateway. When the VOIP Gateway receives the original
called 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 an
international call from this country. The user will have to activate his JAJAH Mobile client and select
a new country from which they wish to make international calls. Once the user has selected the new
country, 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 international
call 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 this
call and let the handset makes the call to the dialled local number. JAJAH Mobile Solution does not
require any co-operation from a mobile operator. A mobile subscriber owning a Symbian handset
can simply navigate to JAJAH website and download the application onto their handset and they can
use the application straight away whether from home country or when roaming abroad. The only
limitation is that the user is limited to those countries where JAJAH has a local PSTN VOIP
Gateway 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. When
the user visits another country and registered on a roaming partner's PLMN, there are two ways for
the user to access his voicemail. If the roaming partner has CAMEL inter-working with his HPLMN
operator, he can dial the normal voicemail short code, and the VPLMN will route the call to the
HPLMN voicemail system. Alternatively, the user can dial the long access number of his voice
mailbox. He will be able to find out this number from his HPLMN website or just by calling the
service desk of his HPLMN operator. When he dials the long access number, he is effectively making
a 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 be
charged 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 to
the local VOIP Gateway. The call is then routed to the HPLMN's voicemail system via Rajah’s global
VOIP network. When the call is delivered by Jajah’s global VOIP network to the home country's
VOIP Gateway, the VOIP Gateway can set up a local circuit switched voice call to the HPLMN 's
voicemail system. Since the HPLMN sees the call is originated from the home country, the HPLMN
can 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 see
their 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 not
demand 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.

© Alan Quayle Business and Service Development, All Rights Reserved


SIP
server

Terminating
Calling party A
SIP PLMN
VOIP
Originating GW GGSN
HPLMN
Called Party B
Home Country Global VOIP network SIP
VOIP
Called Country
GW

Calling party pays local call rate for


international calls to called party B and C.
Jajah client intercepts the call request and
directs the call to a local PSTN VOIP
gateway,
Terminating
Legend:
PSTN
Circuit switched voice call

VOIP media transported as RTP/UDP/IP


Called Party C
SIP signalling

Figure 18. JAJAH Call Flow


Jajah mobile application client based solution call signalling illustration

Orig SIP Term. Term MS B


MS A Orig.
VOIP Proxy VOIP PLMN
PLMN
GW GW

Setup
IAM

ACM
Proceed

ANM
Connect

CS Voice call to local VOIP GW established

Pass original called party number as


DTMF digits to local VOIP GW
INVITE
INVITE IAM
Setup

ACM Proceed
100 Trying 100 Trying
CPG Alerting
183 Session 183 Session
Progress Progress ANM Connect
200 OK
200 OK
ACK ACK

Established end to end voice connection

Figure 19. JAJAH Signalling

© Alan Quayle Business and Service Development. All Rights Reserved


REBTEL

To get started, people sign up for a Rebtel account at www.rebtel.com, where they enter their
mobile phone number and the mobile phone numbers of global friends. Rebtel then instantly creates
pairs of local numbers and sends them in text messages (sms) so they can be saved in the friends’
phone 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 in
London, 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 only
pay for the local call, plus a small per-minute fee to Rebtel, from 2c per min.

Rebtel also offers a REBin, where the user’s local call is connected with their global friend’s local
call in a virtual room on the Internet, called a REBroom. In the REBroom all calls are free – no
matter 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 friend’s local number. The two calls are then
automatically connected, and the friends can hang out and talk for as long as they like, and not worry
about the cost.

With Rebtel, consumers can use the mobile phones they own today, and don’t have to buy
anything else, download software, get a new SIM card, use a headset connected to a computer, or
worry 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 Rebtel’s $1 USD per week
service 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 of
Skype), it recently raised $20m in September 2006. Given the additional configuration, and the
additional cost compared to JAJAH, it is the author’s opinion that REBTEL will need to modify its
proposition to survive.

© Alan Quayle Business and Service Development, All Rights Reserved

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