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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 BY MARKET: REGULATIONS, TARRIFS, DEPLOYMENT OPTIONS / FEASIBILITY, DIRECT MARKET , 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 PROCEDURE (TAP) 10 ROAMING DEFINITIONS 10 TRANSFER 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 BY TECHNOLOGY DESCRIPTIONS GATEWAY OPTION DESCRIPTION OUTGOING CALL SCENARIOS INCOMING CALL SCENARIOS ISSUES HANDSET INTERCEPT OPTION DESCRIPTION ISSUES OTHER OPTIONS DUAL MODE (GSM WIFI) HANDSETS MOBILE CENTRIC START-UPS: JAJAH AND REBTEL JAJAH REBTEL INTERNET BRANDS: SKYPE, YAHOO! YAHOO! – WORKING IN PARTNERSHIP WITH OPERATORS SKYPE AND GOOGLE – LIKELY HEAD-ON COMPETITION CONSUMER BRANDS: NOKIA, APPLE, AND SONY COMPARATIVE ANALYSIS INTERNATIONAL ROAMING REGULATION
16
16 16 16 18 21 21 21 25 25 25 27 27 30 ERROR! BOOKMARK NOT DEFINED. ERROR! BOOKMARK NOT DEFINED. ERROR! BOOKMARK NOT DEFINED. ERROR! BOOKMARK NOT DEFINED. 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
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ASIA PACIFIC REGION RESULTS EUROPE AND MIDDLE EAST RESULTS NORTH AMERICA RESULTS
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MARKET SIZING
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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 BY ERROR! BOOKMARK NOT DEFINED. OPERATOR TARGETING ERROR! BOOKMARK NOT DEFINED. MARKET EVOLUTION ERROR! BOOKMARK NOT DEFINED. OPERATOR IMPACT AND PROPOSITION OPERATOR SITUATION IMPACT OF VOIP BY TO OPERATOR'S BUSINESS VOIP BY PROPOSITION FOR OPERATOR
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CONCLUSIONS AND RECOMMENDATIONS
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APPENDIX 1 – ACRONYMS
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APPENDIX 2 – COMPANY DATA POINTS
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DEUTSCHE TELEKOM
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© Alan Quayle Business and Service Development. All Rights Reserved
TELECOM VODAFONE TELECOM ITALIA COSMOTE TELEFONICA BELGACOM MOBISTAR
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APPENDIX 3: INTERVIEW QUESTIONNAIRE
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APPENDIX 4: INTERVIEW RESULTS
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APPENDIX 5: JAJAH CALLING RATES
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APPENDIX 6: EUROPEAN MEMBER STATE REGULATORY POSITIONS BOOKMARK NOT DEFINED.
ERROR!
NATIONAL REGULATORS AND MINISTRIES ERROR! BOOKMARK NOT DEFINED. FINLAND: EFFECTIVE COMPETITION ERROR! BOOKMARK NOT DEFINED. : 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 T DOMINANCE IDENTIFIED ERROR! BOOKMARK NOT DEFINED. ITALY: NO INDIVIDUAL OR COLLECTIVE MARKET POWER ERROR! BOOKMARK NOT DEFINED. NORWAY: MINISTERIAL 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 Procedure (TAP) – GSM Billing and ing ________________ 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 By 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 ______________________________ Error! Bookmark not defined. Figure 27. EMEA Market ______________________________ Error! Bookmark not defined. Figure 28. NAR Market ________________________________ 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 by, 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 by 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 ’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 goes online to Rebtel website and sign up for the service. The 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 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 and to their friends. The SMS/email provides a local number for the to dial into the PSTN VoIP gateway. When the 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 and the called party number to connect the call to his friend over a VoIP network. When his friend answers the call. The 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 s the two local calls together over a VOIP network. Rebtel has simply employed SMS and email as the bearer to inform the the local phone number to dial to connect to Rebtel’s local PSTN VoIP Gateway. For JAJAH it offers two methods. Method 1: 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 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 ’s handset. Once the JAJAH client is installed, the client will ask the to nominate his home country. The JAJAH client uses this information to select the local PSTN VoIP gateway number to use. When the 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 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 by 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 bying 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 s with the VF home network then is able gather local MGW and MS-ISDN/IMSI details, the phone then s 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 by 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 rate for international calls by routing international calls via by VOIP network
PLMN
Called UE of Country C
By VOIP network
Calling UE of Country A
PLMN
HPLMN
Called UE of Country B
Called Country B
Home Country A
Figure 1. International Calling from the Home Network Case Called Country C Operator subscribers pay local call tariff rate for international calls to home country or to other countries by routing international calls made from VPLMN via by VOIP network
Other PLMNs
By VOIP network
VPLMN
HPLMN
Home Country A
Roaming Country B
Figure 2. International Roaming Case
© Alan Quayle Business and Service Development. All Rights Reserved
Roaming UE
ROAMING DEFINITIONS AND TRANSFER 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 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 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 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 es 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-istration ing 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 Procedure (TAP) – GSM Billing and ing
Figure 4 illustrates the collection and exchange of information required to 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 es 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, markups 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 by 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 realworld 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 ( 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.
Home Portal
HPLMN PS Core
GRX
VPLMN PS Core
GGSN
SGSN
Roaming Country
Home 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 HPLMN CS Core
Home Country Legend:
SIP VOIP GW
GRX / Internet
Home operator has GGSN and SIP server in roaming countries
GGSN
SGSN
Roaming Country Roaming UE 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
HPLMN PS Core
Home Country Called UE with SIP client
GRX / Internet
Home operator has GGSN and SIP server in roaming countries
GGSN
SGSN
Roaming Country Roaming UE 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 By mode, see Figure 10. Cisco provides a SIP VOIP gateway which allows an incoming call to by 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.
UTRAN
Setup
CS Core network
ISUP IAM
SIP VOIP Gateway System
Voice over IP network
Alerting Connect
ISUP ACM
SIP 183 Session Progress
ISUP Facility
SIP 180 Ringing
ISUP ANM
CS Core Network
SIP 200 OK
ISUP IAM ISUP ACM ISUP Facility ISUP ANM
ACK
Connect Ack
UTRAN
SIP INVITE SIP 100 Trying
Call Proceed
SIP VOIP Gateway System
Setup Call Proceed Alerting Connect Connect Ack
AUDIO LOGICAL CHANNELS
Figure 10. Cisco Voice Toll By 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 by this method of termination, to avoid roaming and termination charges placed by the Visited PLMN on the Home PLMN. This by 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 by 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 by 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
VPLMN CS Core
Intl ISDN Transit Network
HLR GMSC
VMSC
Roaming Country
Home Country
B-pty on international roaming pays for the roaming call forward leg of the incoming call
A-pty pays for the originating call leg to the HPLMN
GMSC
Originating Country
Figure 11. Classic GSM Roaming Call Forward Call Model
SIP server
HPLMN CS Core HLR GMSC
VPLMN CS Core SIP VOIP GW
SIP VOIP GW
VMSC
VOIP network
Roaming Country
Home Country
GMSC
Legend:
A-pty pays for the originating call leg to the HPLMN
B-pty on international roaming pays for the roaming call forward leg of the incoming call at lower tariff rate
Originating Country 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)
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SIP server
HPLMN CS Core HLR GMSC
VPLMN PS Core SIP VOIP GW
IMS
VOIP network
Roaming Country
Home Country
GMSC
Legend:
B-pty on international roaming pays for the roaming call forward leg of the incoming call at lower tariff rate
A-pty pays for the originating call leg to the HPLMN
Originating Country 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
Legend: VOIP media transported as RTP/UDP/IP
Roaming Country
B-pty on international roaming pays for the roaming call forward leg of the incoming call at lower tariff rate
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 by 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 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 ed 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 by proposal will take away all these value added voice services from the subscriber as the application of CAMEL call control at VPLMN will be byed. 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 setup. In its roaming stage it s with the VF home network then is able gather local MGW and MS-ISDN/IMSI details, the phone then s 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 MSISDN/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 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 ed with the HLR of the VPLMN, incoming call to this subscriber will somehow have to be routed directly to the VPLMN bying the HPLMN. One option: The Home network s 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 (P) call.
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SIP server sets up call to HPLMN SIP VOIP GW based on called party number received from roaming UE SIP Server
HPLMN CS Core
SIP VOIP GW
Roaming Country
GRX / Internet
SIP VOIP GW
VPLMN CS core
Home Country GGSN
Roaming UE called party number either us PS data connection or use DTMF signalling in a two stage dialling process
SGSN
VPLMN PS core
Roaming UE
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 ing of called party number over PS data connection
Figure 15. Handset Centric: Circuit Switched Far End
SIP server sets up call to HPLMN SIP VOIP GW based on called party number received from roaming UE
Roaming UE called party number either us PS data connection or use DTMF signalling in a two stage dialling process SIP Server
HPLMN PS Core
IMS
GRX / Internet
Roaming Country SIP VOIP GW
VPLMN CS core
Home Country GGSN
Called UE with SIP client
SGSN
VPLMN PS core
Roaming UE
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 ing 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 setup with forwarding to VoIP by network with ‘local visited number’ – maybe this SIP server does the translation not HLR?
(2) SIP server sets up call to SIP VOIP GW in visited country using ‘local visited number’ SIP Server
HPLMN CS Core
SIP VOIP GW
GRX / Internet
Roaming Country SIP VOIP GW
VPLMN CS core
HLR/ HSS
Home Country Roaming UE called party number either us PS data connection or use DTMF signalling in a two stage dialling process
GGSN
SGSN
VPLMN PS core
Roaming UE
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 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 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 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 by this trombone call routing is for the roamer to be allocated a local MSISDN from a local PLMN.
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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 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 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 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 and conditions of the Foreign MS-ISDNs bought by the Home operator restrict such usage for such a service? Assumes the VPLMN s 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 s 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).
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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 802.11e standard will bring QoS to WiFi 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 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 s 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
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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: 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 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 ’s handset. Once the JAJAH client is installed, the client will ask the to nominate his home country. The JAJAH client uses this information to select the local PSTN VoIP gateway number to use. When the 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 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 from which country they wish to call from. Once the 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 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 subsequently decides to visit another country, and wish to make an international call from this country. The will have to activate his JAJAH Mobile client and select a new country from which they wish to make international calls. Once the 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 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 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 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 is limited to those countries where JAJAH has a local PSTN VOIP Gateway deployed. In fact, the can even benefit from accessing his HPLMN voicemail when roaming abroad. For example, he accesses his voicemail by dialling a short code when ed on HPLMN. When the visits another country and ed on a roaming partner's PLMN, there are two ways for the 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 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 has the JAJAH Mobile client installed, the 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 by 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
Calling party A
Terminating PLMN
SIP VOIP GW
Originating HPLMN
GGSN
Called Party B Global VOIP network
Home Country
SIP VOIP 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,
Called Country
Terminating PSTN
Legend: 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 Term PLMN
Term. VOIP GW
SIP Proxy
Orig VOIP GW
Orig. PLMN
MS A
MS B
Setup IAM Proceed Connect
ACM ANM
CS Voice call to local VOIP GW established original called party number as DTMF digits to local VOIP GW
INVITE
INVITE
IAM Setup Proceed
100 Trying
100 Trying
ACM G
Alerting
183 Session Progress
183 Session Progress 200 OK
ANM
Connect
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 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 ’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 simply hangs up, and while the friend hangs on, the 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, 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 , no is necessary. And, s 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, , , 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.
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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 BY MARKET: REGULATIONS, TARRIFS, DEPLOYMENT OPTIONS / FEASIBILITY, DIRECT MARKET , 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 PROCEDURE (TAP) 10 ROAMING DEFINITIONS 10 TRANSFER 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 BY TECHNOLOGY DESCRIPTIONS GATEWAY OPTION DESCRIPTION OUTGOING CALL SCENARIOS INCOMING CALL SCENARIOS ISSUES HANDSET INTERCEPT OPTION DESCRIPTION ISSUES OTHER OPTIONS DUAL MODE (GSM WIFI) HANDSETS MOBILE CENTRIC START-UPS: JAJAH AND REBTEL JAJAH REBTEL INTERNET BRANDS: SKYPE, YAHOO! YAHOO! – WORKING IN PARTNERSHIP WITH OPERATORS SKYPE AND GOOGLE – LIKELY HEAD-ON COMPETITION CONSUMER BRANDS: NOKIA, APPLE, AND SONY COMPARATIVE ANALYSIS INTERNATIONAL ROAMING REGULATION
16
16 16 16 18 21 21 21 25 25 25 27 27 30 ERROR! BOOKMARK NOT DEFINED. ERROR! BOOKMARK NOT DEFINED. ERROR! BOOKMARK NOT DEFINED. ERROR! BOOKMARK NOT DEFINED. 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
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ASIA PACIFIC REGION RESULTS EUROPE AND MIDDLE EAST RESULTS NORTH AMERICA RESULTS
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MARKET SIZING
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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 BY ERROR! BOOKMARK NOT DEFINED. OPERATOR TARGETING ERROR! BOOKMARK NOT DEFINED. MARKET EVOLUTION ERROR! BOOKMARK NOT DEFINED. OPERATOR IMPACT AND PROPOSITION OPERATOR SITUATION IMPACT OF VOIP BY TO OPERATOR'S BUSINESS VOIP BY PROPOSITION FOR OPERATOR
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CONCLUSIONS AND RECOMMENDATIONS
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APPENDIX 1 – ACRONYMS
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APPENDIX 2 – COMPANY DATA POINTS
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DEUTSCHE TELEKOM
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TELECOM VODAFONE TELECOM ITALIA COSMOTE TELEFONICA BELGACOM MOBISTAR
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APPENDIX 3: INTERVIEW QUESTIONNAIRE
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APPENDIX 4: INTERVIEW RESULTS
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APPENDIX 5: JAJAH CALLING RATES
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APPENDIX 6: EUROPEAN MEMBER STATE REGULATORY POSITIONS BOOKMARK NOT DEFINED.
ERROR!
NATIONAL REGULATORS AND MINISTRIES ERROR! BOOKMARK NOT DEFINED. FINLAND: EFFECTIVE COMPETITION ERROR! BOOKMARK NOT DEFINED. : 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 T DOMINANCE IDENTIFIED ERROR! BOOKMARK NOT DEFINED. ITALY: NO INDIVIDUAL OR COLLECTIVE MARKET POWER ERROR! BOOKMARK NOT DEFINED. NORWAY: MINISTERIAL OF EC REGULATION ERROR! BOOKMARK NOT DEFINED. OVERLAP WITH NRA MARKET 11 REVIEWS ERROR! BOOKMARK NOT DEFINED.
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TABLE OF FIGURES Figure 1. International Calling from the Home Network Case _________________________________ 9 Figure 2. International Roaming Case ____________________________________________________ 9 Figure 3. Transferred Procedure (TAP) – GSM Billing and ing ________________ 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 By 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 ______________________________ Error! Bookmark not defined. Figure 27. EMEA Market ______________________________ Error! Bookmark not defined. Figure 28. NAR Market ________________________________ 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 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 by, 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.
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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 by 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 ’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 goes online to Rebtel website and sign up for the service. The 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 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 and to their friends. The SMS/email provides a local number for the to dial into the PSTN VoIP gateway. When the 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 and the called party number to connect the call to his friend over a VoIP network. When his friend answers the call. The 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 s the two local calls together over a VOIP network. Rebtel has simply employed SMS and email as the bearer to inform the the local phone number to dial to connect to Rebtel’s local PSTN VoIP Gateway. For JAJAH it offers two methods. Method 1: 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 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 ’s handset. Once the JAJAH client is installed, the client will ask the to nominate his home country. The JAJAH client uses this information to select the local PSTN VoIP gateway number to use. When the 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 the
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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 by 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 bying 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 s with the VF home network then is able gather local MGW and MS-ISDN/IMSI details, the phone then s 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 by 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.
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Called Country C Operator’s subscribers pay local call tariff rate for international calls by routing international calls via by VOIP network
PLMN
Called UE of Country C
By VOIP network
Calling UE of Country A
PLMN
HPLMN
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Figure 1. International Calling from the Home Network Case Called Country C Operator subscribers pay local call tariff rate for international calls to home country or to other countries by routing international calls made from VPLMN via by VOIP network
Other PLMNs
By VOIP network
VPLMN
HPLMN
Home Country A
Roaming Country B
Figure 2. International Roaming Case
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Roaming UE
ROAMING DEFINITIONS AND TRANSFER 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 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 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 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.
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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 es 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-istration ing 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 Procedure (TAP) – GSM Billing and ing
Figure 4 illustrates the collection and exchange of information required to 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
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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
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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), see Figure 5. The host operator es 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
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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 not found.. Home Operator Mark-up Covering operating costs and profit, Another area of regulator concern, markups 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
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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
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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 by 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 realworld 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 ( 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.
Home Portal
HPLMN PS Core
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GGSN
SGSN
Roaming Country
Home 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.
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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 HPLMN CS Core
Home Country Legend:
SIP VOIP GW
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Home operator has GGSN and SIP server in roaming countries
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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
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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
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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 By mode, see Figure 10. Cisco provides a SIP VOIP gateway which allows an incoming call to by 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.
UTRAN
Setup
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ACK
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AUDIO LOGICAL CHANNELS
Figure 10. Cisco Voice Toll By 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 by this method of termination, to avoid roaming and termination charges placed by the Visited PLMN on the Home PLMN. This by 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 by 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 by 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.
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HPLMN CS Core
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B-pty on international roaming pays for the roaming call forward leg of the incoming call
A-pty pays for the originating call leg to the HPLMN
GMSC
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Figure 11. Classic GSM Roaming Call Forward Call Model
SIP server
HPLMN CS Core HLR GMSC
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Legend:
A-pty pays for the originating call leg to the HPLMN
B-pty on international roaming pays for the roaming call forward leg of the incoming call at lower tariff rate
Originating Country 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)
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SIP server
HPLMN CS Core HLR GMSC
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Legend:
B-pty on international roaming pays for the roaming call forward leg of the incoming call at lower tariff rate
A-pty pays for the originating call leg to the HPLMN
Originating Country 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
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Legend: VOIP media transported as RTP/UDP/IP
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B-pty on international roaming pays for the roaming call forward leg of the incoming call at lower tariff rate
SIP call session signalling path IMS Call Session Path
Figure 14. Roaming Call Forward (Option 3)
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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 by 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 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 ed 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 by proposal will take away all these value added voice services from the subscriber as the application of CAMEL call control at VPLMN will be byed. 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 setup. In its roaming stage it s with the VF home network then is able gather local MGW and MS-ISDN/IMSI details, the phone then s as a SG phone. Or the UE’s roaming table is pre-
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programmed with the MGW and MS-ISDN/IMSI details. This assumes a dedicated MSISDN/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 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 ed with the HLR of the VPLMN, incoming call to this subscriber will somehow have to be routed directly to the VPLMN bying the HPLMN. One option: The Home network s 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 (P) call.
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SIP server sets up call to HPLMN SIP VOIP GW based on called party number received from roaming UE SIP Server
HPLMN CS Core
SIP VOIP GW
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Roaming UE called party number either us PS data connection or use DTMF signalling in a two stage dialling process
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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 ing of called party number over PS data connection
Figure 15. Handset Centric: Circuit Switched Far End
SIP server sets up call to HPLMN SIP VOIP GW based on called party number received from roaming UE
Roaming UE called party number either us PS data connection or use DTMF signalling in a two stage dialling process SIP Server
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Roaming UE
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 ing 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 setup with forwarding to VoIP by network with ‘local visited number’ – maybe this SIP server does the translation not HLR?
(2) SIP server sets up call to SIP VOIP GW in visited country using ‘local visited number’ SIP Server
HPLMN CS Core
SIP VOIP GW
GRX / Internet
Roaming Country SIP VOIP GW
VPLMN CS core
HLR/ HSS
Home Country Roaming UE called party number either us PS data connection or use DTMF signalling in a two stage dialling process
GGSN
SGSN
VPLMN PS core
Roaming UE
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 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 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 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 by 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 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 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 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 and conditions of the Foreign MS-ISDNs bought by the Home operator restrict such usage for such a service? Assumes the VPLMN s 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 s 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 802.11e standard will bring QoS to WiFi 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 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 s 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: 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 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 ’s handset. Once the JAJAH client is installed, the client will ask the to nominate his home country. The JAJAH client uses this information to select the local PSTN VoIP gateway number to use. When the 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 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 from which country they wish to call from. Once the 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 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 subsequently decides to visit another country, and wish to make an international call from this country. The will have to activate his JAJAH Mobile client and select a new country from which they wish to make international calls. Once the 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 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 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 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 is limited to those countries where JAJAH has a local PSTN VOIP Gateway deployed. In fact, the can even benefit from accessing his HPLMN voicemail when roaming abroad. For example, he accesses his voicemail by dialling a short code when ed on HPLMN. When the visits another country and ed on a roaming partner's PLMN, there are two ways for the 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 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 has the JAJAH Mobile client installed, the 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 by 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
Calling party A
Terminating PLMN
SIP VOIP GW
Originating HPLMN
GGSN
Called Party B Global VOIP network
Home Country
SIP VOIP 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,
Called Country
Terminating PSTN
Legend: 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 Term PLMN
Term. VOIP GW
SIP Proxy
Orig VOIP GW
Orig. PLMN
MS A
MS B
Setup IAM Proceed Connect
ACM ANM
CS Voice call to local VOIP GW established original called party number as DTMF digits to local VOIP GW
INVITE
INVITE
IAM Setup Proceed
100 Trying
100 Trying
ACM G
Alerting
183 Session Progress
183 Session Progress 200 OK
ANM
Connect
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 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 ’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 simply hangs up, and while the friend hangs on, the 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, 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 , no is necessary. And, s 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, , , 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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