28 Converged service for fixed and mobile telephony In coming years, virtually every operator will need to update the fixed and mobile network infrastructure. But when, where, how, and to what extent? While the options are truly numerous the best alternatives are not always obvious. P e t e r Gr a n s t röm, L e n na rt Nor e l l a n d S v e n Å k e s s on Even though the evolutionary paths of transformation differ for fixed and mobile operators, both groups target the same long-term objective: converged multimedia-capable person-toperson service. When planning for network evolution, operators should take fixed-mobile convergence, the positioning of the Rich Communication Suite with, and the various alternatives for introducing mobile telephony into consideration. The 3rd Generation Partnership Project (PP) has defined the IP Multimedia Subsystem () as the service engine for next-generation IP networks. ETSI TISPAN, PP2 and CableLabs, in turn, have adopted the PP standards, thereby extending their validity to networks. In addition, the industry has identified a need for detailed standardization of key -based services, in particular BOX A PP ARPU CAPEX DSL DTM FTTH FWT I for person-to-person communication. Standardizing such services facilitates the multivendor interoperability and operator interconnects that are needed for global mass-market uptake. ETSI TISPAN has thus initiated the standardization of -based multimedia telephony services ().1 enables people to communicate using any combination of voice, video, video-sharing and real-time text. It also enables integration with a presence-enhanced phonebook and defines a set of supplementary and regulatory services. The specifications are referenced by PP and extended to operate on PP accesses. thus plays a fundamental part in enabling fixedmobile convergence in networks and services. Background The situation for telephony services in fixed networks is quite different from that for mobile networks. networks are experiencing growth in number of users and traffic volumes whereas traditional fixed telephony service is faced with ongoing loss of users and traffic volumes. Therefore, in the near term, one can anticipate BOX b softswitch A softswitch is a central device in a telecommunications network that connects calls from one phone line to another entirely by means of software running on a computer system. This work was formerly carried out by hardware that employed physical switchboards to route calls. A softswitch separates network hardware from network software. Terms and abbreviations 3rd Generation Partnership Project average revenue per user capital expenditures digital subscriber line dual-transfer mode fiber to the home fixed-wireless terminal high-speed packet access centralized services IP Multimedia Subsystem NNI OMA OPEX TN R SR-VCC VoIP PP Long Term Evolution multimedia telephony services network-to-network interface Open Alliance operating expenses public switched telephone network Rich Communication Suite single radio voice call continuity voice over IP that telephony in fixed and mobile networks will evolve along different paths (Figure 1). Both areas, though, target the same long-term objective: converged multimedia-capable personto-person service. The expansion of access, which opens up traditional wireline telephony service for competition from voice over IP (VoIP) over access, is driving the evolution of wireline telephony. VoIP is typically offered as part of a service bundle with access and possibly IPTV. Over time, access will evolve from digital subscriber line (DSL) technology to optical fiber and mobile access technologies that do not support legacy voice service. As a consequence, voice service must either evolve toward a converged -based service or fragment into stovepipe voice solutions for each access technology. At present, the main emphasis in mobile telephony is on managing subscriber growth and lowering costs in order to serve low average revenue per user (ARPU) subscriber groups. This calls for a rationalization of the circuit-switched network by introducing mobile softswitch on IP. services are introduced alongside circuit-switched telephony to provide an enhanced user experience, but VoIP will not replace legacy telephony service in the near future. Wireline telephony evolution The evolution of wireline telephony service is characterized by a migration of users from traditional wireline service to mobile telephony and VoIP services, which are typically bundled together with internet access. The challenge for operators is to balance E r i c s s o n r e v i e w 2 2009 C_209a.indd 28 09-09-15 15.18.31
29 investments in legacy telephone networks with investments to reduce operating expences in new -based telephony services. The aim of the new services is to retain subscribers who give up traditional wireline telephony for offerings. enables operators to offer converged service for migrating TN from traditional copper lines to new access types, such as fiber to the home (FTTH) and fixed-wireless access. Figure 2 portrays as a common service engine for all fixed accesses. The initial driver is the rapid growth of wireline access (DSL, fiber, etc.) where provides telephony services (Figure 2, household at bottom left). is not fully equivalent to legacy TN service, but to maximize synergies, many operators, especially in markets with a high penetration of VoIP, are considering for TN access (Figure 2, middle household). The household at the bottom right of Figure 2 portrays as the service engine for fixed-wireless terminals FIGUrE 2 (FWT). The FWT segment is an attractive alternative to DSL that is expected to grow. FWT deployments with might constitute an initial evolutionary step toward introducing for mobile access (,, etc.) including mobility support. One advantage of is that it provides a standardized and access-agnostic service definition that works with any access, such as copper loop (0.3-3.4kHz voice or VoIP/DSL), FTTH, mobile, and cable. It thus opens telephone networks to an optimal mix of access technologies, which is ideal given that, not voice, is driving the evolution. In all likelihood the wireline service will first be used for plain VoIP, but over time it will migrate to multimedia services. This transformation will be influenced by the service evolution of the Rich Communication Suite (R) on the mobile side. is also a key enabler of new converged multimedia service offerings that combine fixed and mobile terminals. As such it gives operators new opportunities to generate revenue and fight churn. FIGUrE 1 Evolution fixed and mobile telephony exhibit different transformation paths. R Fixed Converged service. MGW DSL, Fiber TN IP network Fixed SIP SIP, HTTP SIP, HTTP AGW multimedia POTS Fixed-wireless terminal (Broadband + mutimedia) Telephony in mobile terminals Broadband multimedia VoIP telephony evolution Several parallel trends are driving network modernization in mobile networks. Chief among these are continued growth in /GSM and the rise of standardized multimedia services. Traffic volumes are still growing substantially, driven by reduced tariffs, the popularity of flat rates, and bundled offerings. However, the ARPU generated by legacy telephony and SMS is flattening out or declining. Accordingly, operators are seeking out new revenue streams to offset dwindling revenues from traditional services. In addition, to stay competitive and secure good margins, operators must reduce their operating expenses (OPEX). At present, traditional voice telephony and SMS account for the lion s share of operator revenues. A multimedia service offering gives operators a way of seeking new revenue streams and of reducing churn and decline in revenues from traditional services. Multimedia services might also enrich traditional telephony by adding presence and other compelling network-based content. For end-users, multimedia services are about personalization that is, having access to individual data via a device of choice whenever and wherever one wants it. Migration to multimedia-enriched telephony operators have a strong position and platform on which to build enriched communications service offerings. The GSMA R initiative is a joint effort between leading industry stakeholders to facilitate the adoption of applications and services that provide an interope r i c s s o n r E v i E w 2 2009 C_209a.indd 29 09-09-15 15.18.32
30 FIGUrE 3 GSMA r r1. Presence Chat Multimedia messaging Video share Image share File transfer Voice erable, converged, rich communication experience that is based on.2 Initially, R focuses on time to market, interoperability and an enhanced user experience through integration and aggregation of a multitude of existing service components with established inter-operator network-to-network interfaces (NNI). It offers a client-based service architecture that is centered around a presenceenabled address book. Figure 3 illustrates the initial (R R1) high-level architecture, where legacy mobile circuit-switched services (voice, video, etc.) are combined and enriched with for other media types and service components, such as an active FIGUrE 4 devices. presence-enabled phonebook, video sharing, image sharing and OMA instant messaging for chat and file sharing. It should be noted that is not used in the initial R R1 initiative. However, the R R1-based solutions should be introduced at around the same time as is introduced for wireline access. Therefore, to achieve the vision of fixed and mobile convergence, the industry should join forces in evolving toward a combined R and track for operators with a converged fixed-mobile offering. The purpose of R R2, as illustrated in Figure 4, is to take the first step GSMA r r2 adding support for fixed Presence Chat Multimedia messaging Video share Image share File transfer Voice BB toward such convergence by introducing support for wireline devices (including FWT devices) as a complement to R R1 mobile service. Doing so gives operators the ability to bundle services and thereby become a fixed/mobile community alternative to over-the-top internet services. R R2 supports the concept of a common mobile identity for all devices and will introduce for voice. The decision to include the video part of in R R3 has not yet been made. After - and -based mobile has been introduced, users will in all likelihood repeat the behavior seen in fixed networks. That is, they will start to adopt internet-centric communication styles, which increasingly revolve around the sharing of everyday life experiences anywhere, anytime and on any device. The evolution of R to gradually include for mobile packet access will provide a true converged multimedia solution (Figure 2). Initially, will be introduced for wireline access and devices. Over time, R over and will become more radio-efficient than present-day circuit-switched voice technology. At that point, R (using for voice, video, video-sharing and real-time text) over and can fully replace circuit-switched access while also reducing operators capital expenditures (CAPEX) and OPEX. telephony As outlined in Figure 5, the industry is discussing a variety of alternatives for migrating legacy circuit-switched telephony to mobile packet access. For, which is specified as packet-only access, the obvious alternative is. For / phones, the default telephony service is circuit-switched service. When is introduced for, then over would be a clear complement providing transparently over high-speed packet technologies. Compared with circuit-switched voice in /, packet-based voice service (such as ) also has the potential to lower power consumption in mobile devices. These savings can also be achieved with the circuit- E r i c s s o n r E v i E w 2 2009 C_209a.indd 30 09-09-15 15.18.33
31 FIGUrE 5 Telephony alternatives for mobile. SM /GSM / Packet core GSM el MMT PA S oh switched-over- method when is used as a bearer for legacy circuit-switched telephony service. However, this method has no synergies with. Many GSM/-based operators will have an aggressive deployment strategy prior to introducing. Initially, they will deploy to boost data capacity for laptops. Later, when handhelds become available, will be a long-term solution for telephony service. As an intermediate step, they can use the circuit-switched fallback (FB) mechanism that is part of the PP spcifications. This way, users of who receive or place a call temporarily fall back (for the duration of the call) to the circuit-switched network. Circuit-switched fallback is associated with minor call setup delay but should provide acceptable service when good / coverage overlaps coverage. The advantage of circuit-switched fallback is that it decouples the introduction of and in the networks. It will also be an intermediate solution for inbound roamers, since it can use existing roaming agreements for circuit-switched telephony. If is used for telephony, the sensible mechanism when leaving coverage is packet handover to /. This handover, which is transparent for, is handled on the packet core level. If the combined highspeed packet coverage from and deployments is spotty when operators introduce, then they will also have to provide a handover mechanism to circuit-switched telephony. The PP standardization of SR-VCC (single radio voice call continuity) and I ( centralized services) address this contingency by anchoring calls in the domain and providing services via. /GSM access is used when the caller (terminal) is outside high-speed packet coverage. The situation for operators who migrate CDMA to is similar to that for /. When CDMA operators deploy after having introduced high-speed packet enhancements to CDMA (ehrpd), the next logical telephony solution is. To provide uninterrupted calls to subscribers who leave areas with high-speed pack- GSM et access, the system must hand over from to ehrpd and possibly to circuit-switched CDMA service. Some operators will transition directly from GSM to. The introduction of will thus also be their first deployment of high-speed packet data access. These operators will typically introduce as a mobile service for laptops. However, they will also have to support telephony in -capable handhelds. A drawback to the circuitswitched GSM fallback mechanism is that data services will be lost or suspended if the GSM network does not support dual-transfer mode (DTM). Moreover, even where DTM is available, the supported data rates will be very low compared with those of. Consequently it is not possible to decouple deployment from providing a packet telephony service. A sensible choice would be to deploy, but some early adopters of have raised concerns about the dependency between and deployments. Consequently, alternative solutions have been suggested that reuse GSM control for VoIP calls over. In this case, the SR-VCC-like mechanism is needed to allow handover to GSM access when a terminal leaves coverage. A main concern with this approach is that there are already too many alternative solutions. Therefore, to avoid fragmentation in the terminal market, the industry needs to agree on a select few of these. Conclusion While fixed and mobile operators follow different evolutionary paths, the end goal is the same: a converged multimedia service offering that meets the competition from communication clients and counters the price pressure and churn from traditional voice service. A combined Rich Communication Suite (R) and Multimedia Telephony () track is a key aspect for reaching that goal. Initially R will be the starting point for mobile networks, serving to enrich legacy circuitswitched service; likewise, will be the starting point in wireline for -based voice over IP (VoIP). Over time will be introduced within R, initially for wireline accesses and devices (including fixed-wireless terminals, FWT) and later also for mobile. A fully converged all-ip R (including ) gives a rich multimedia E r i c s s o n r E v i E w 2 2009 C_209a.indd 31 09-09-15 15.18.34
32 service that is interoperable across the entire telecommunications community regardless of access or device; that is, its services are valid across PCs, mobile terminals, IPTV sets, and so forth. Operators with a converged fixedmobile offering benefit from reduced CAPEX and OPEX because they can use the same core network and service engine to serve all fixed and mobile users. Changes and extensions of the service set become valid for all access types as soon as they are introduced there is no need for dual development or synchronization between different service engines. Peter Granström Lennart Norell joined Ericsson in 1985 to work with the AXE IOG11 system. Over the years he has gained global experience in a variety of markets (USA, UK and Sweden) and has held various positions in the areas of strategic product management, technical solutions management, systems management and R&D. He has been a key contributor in defining and communicating strategies since the inception of. At present, he is an Expert in product management and strategies. Peter holds an M.Sc. in engineering physics from the Uppsala University of Technology, Sweden. joined Ellemtel in 1977 to work in the development of the AXE system. He moved to Ericsson in 1982 and has since held various management and technical expert positions within system and product management in the telecom and datacom product areas. He has been active in the design of where he has been leading the strategic systems management unit. He currently works with system management as an Expert in systems and technology at Business Unit Networks. He holds an M.Sc. in electrical engineering from the Chalmers University of Technology, Sweden. Sven Åkesson joined Ericsson in 1989. He has held various management positions within system management in core and E2E networks where he was instrumental in the definition and startup of GPRS, MSS and. At present, he leads the Strategy & Business Plan unit at Business Unit Networks, where among other things, he has led the telephony evolution work. Sven holds an M.Sc. in electrical engineering from the Chalmers University of Technology, Sweden. References 1. Enström, D., Nohlgren, A., Olofsson, H., Peisa, J. and Synnergren, P.: Multimedia Telephony for Interoperable VoIP with Multimedia support. Ericsson Review, Vol. 84(2007):2 pp. 44-47 2. Badulescu, C., Greene, N., Gustavsson, Å., Jarmillo, C., Leclerc, M., Postmus, P., Saavedra, G. and Servant, M.: Delivering the optimal enduser experience: Ericsson Multimedia Communication Suite. Ericsson Review, Vol. 85(2008): 2 pp. 52-57 E r i c s s o n r e v i e w 2 2009 C_209a.indd 32 09-09-15 15.18.35