EXPLOITING SIMILARITIES BETWEEN SIP AND RAS: THE ROLE OF THE RAS PROVIDER IN INTERNET TELEPHONY. Nick Marly, Dominique Chantrain, Jurgen Hofkens

Transcription

1 Nick Marly, Dominique Chantrain, Jurgen Hofkens Alcatel Francis Wellesplein 1 B-2018 Antwerp Belgium Key Theme T3 Tel : (+32) Fax : (+32) Nick.Marly@alcatel.be Tel : (+32) Fax : (+32) Dominique.Chantrain@alcatel.be Tel : (+32) Fax : (+32) Jurgen.Hofkens@alcatel.be

2 Abstract Most of the attention on Internet telephony has concentrated on interworking between the Internet and the traditional telephone network. However, this paper considers the use of Voice over IP services by remote access users. It focuses on the service scenarios and protocols involved in initiating and receiving calls from a host connected to the Internet, typically via a narrowband dial-up service or an Asymmetric Digital Subscriber Line (ADSL) connection. The paper investigates the similarities between the scenarios and protocols for remote access services and for Internet telephony, and explores opportunities for value-added services associated with connection establishment. It then takes a closer look at the business environment for these two service categories. Based on the results of two studies, a number of business and technical opportunities are identified for integrating Voice over IP services with the overall service offer of a Remote Access Service provider. Because they have a first hand, trustworthy relationship with users, access providers have a significant advantage in deploying and commercializing these new services. The paper highlights the new functions required to support this service, and describes how intelligence can be distributed over the various network elements involved. At the end, some conclusions are formulated.

3 1. Introduction Before a PC user can perform any actions related to Internet telephony (e.g. set up or receive a call), a connection has to be established with the network (Internet, corporate network or virtual private network). This service, which is typically supplied by the Access Provider, is generally referred to as the Remote Access Service (RAS). This service covers aspects of IP-address assignment (e.g. Dynamic Host Configuration Protocol; DHCP), as well as authentication and connection parameter negotiations (e.g. Point-to-Point Protocol; PPP). Many aspects of Internet telephony from/to a host, start with an established Internet Protocol (IP) connection to the network. There is a focus on the signaling protocols used to set up or tear down the call, the establishment of a data path (routing and bandwidth reservation) and interworking between the Public Switched Telephone Network (PSTN) and Voice over Internet Protocol (VoIP). This paper addresses some of the numerous issues and opportunities for offering value-added services and features associated with establishing a connection (e.g. user location, private or public IP address assignment, auto-registration, dial-out features, authentication and authorization, accounting, billing). Examining these opportunities is a legitimate goal as VoIP will (probably) be one of the first and most widely deployed services. Furthermore, Internet telephony is a major opportunity for access providers to create value-added services [1] by offering a variety of Quality of Service (QoS) levels, a choice of long distance carriers, directory services (White/Yellow Pages), Web-based access to voice mail, and a Unified Messaging System (UMS). These services should be integrated in the overall service portfolio of the RAS provider. The following three sections discuss scenarios for remote access and Internet telephony, together with their business opportunities. Section 5 examines a service based on RAS and Internet telephony integration, while Section 6 looks at how to add distributed intelligence to this integrated service. The final section summarizes the various business opportunities and technologies. 2. Scenarios for remote access services During connection/session setup, certain configuration parameters are negotiated between the host Network Termination (NT) and the Network Access Server (NAS), as shown in Figure 1, and at several layers in the protocol hierarchy. At the lower layers, in the case of a narrowband connection this typically involves negotiating the modem parameters. If IP is used, an IP address has to be assigned and additional information (netmask, Domain Name Server address) has to be provided, when applicable. Authentication and authorization have to be carried out when requested by the NAS. 1

4 Corporate NW User Access Network NAS Telco owned Data NW Figure 1 Remote access architecture Internet Currently the main protocols used are PPP [2] and DHCP [3]. The connection between host and NAS and the service selection mechanism can be realized in several ways. In the permanent connection scenario, the host has a permanent logical link with a fixed Virtual Private Network (VPN) and receives permanent configuration parameters from the pool of configuration parameters for that VPN (typically using static allocation). An alternative approach, sometimes referred to as the direct connect scenario, also establishes a connection to a fixed VPN, but in this case the resources (e.g. IP address) are allocated dynamically. Whereas these two methods provide a connection with a predefined VPN, a third approach, known as the "service select" scenario, allows users to choose the VPN from a list of VPNs or services to which they have subscribed. Some NASs realize this function by automatically uploading a Web page to the user showing which VPNs (services) can be selected. Service selection can be compared to the telephone directory service. The service select scenario relies on various features (connection parameter negotiation, authentication, selection of the destination, etc) which are common to Internet telephony. This similarity will be considered later. The NAS is an ideal point to offer value-added services to the user. During connection establishment, the user must first connect to the NAS. Owning the customer profiles at this point gives the access provider a competitive advantage when deploying customized value-added services. Many users will take up these services in view of their trusted relationship with the Telco. 3. Internet telephony scenarios In the PSTN, the voice path and signaling path (Signaling System No 7) use the same routing tables and the voice path is established simultaneously with signal propagation and before ringing is enabled. However, this is not the case in VoIP or Internet telephony. Signaling is independent of data path reservation and follows an independent route. Signaling in the Session Initiation Protocol (SIP) [4] is mainly concerned with locating the called party and alerting him or her of the incoming phone call, negotiating common media parameters, and 2

5 letting the caller know whether or not the call could be established. In the PSTN, the signal passes through several Signal Transfer Points (STP) which are linked to local or transit exchanges (providing the trunk to carry the voice signal). In the architecture proposed for SIP, the signals traverse state-aware (edge of the network) and/or state-unaware (core network) Proxy Servers (PS), but no routers have to be associated with these servers. Furthermore, users can register their names and domains with a registrar (possibly collocated with a proxy server), which can be contacted to determine the current location of the user (inbound calls). PS PS LS SIP SIP SIP Internet UAS RTP UAC Figure 2 Host-to-host signaling (SIP) and data (RTP) paths for VoIP SIP signaling uses the architecture shown in Figure 2. To initiate a call, the caller s User Agent Client (UAC) sends a request to establish a call (called party and domain are provided) to a proxy server which handles the request on behalf of the user. The proxy server contacts a Location Server (LS) to determine which proxy server to contact to reach the called party. If the called party is using the PSTN, a call agent, which interfaces with the PSTN signaling network and controls a media gateway, handles the call. However, if the called party is connected to the Internet, the proxy server handling calls in the called party's domain contacts the called party's User Agent Server (UAS). Eventually the call is accepted by the UAS and the response propagates back to the UAC following the same path as the original request. The voice stream is carried on top of the Real-time Transport Protocol (RTP) and the route is established using, for example, the Border Gateway Protocol (BGP). A certain QoS can be guaranteed using IntServ (access network) or DiffServ (core network). 4. Business aspects The technologies and scenarios used for the registrar and the location server largely determine the commercial opportunities for VoIP. The registrar is equipped with a valuable database 3

6 containing information about all the VoIP service subscribers in a domain. As competition in value-added services and in the local loop increases, the combination of RASs and VoIP services could turn out to be a winning package. Connectivity providers are already terminating remote access sessions and know their customers' behavior, locations and IP addresses. If the provider also offers VoIP services, a variety of value-added services becomes possible (see Sections 5 and 6). Such services are attractive to consumers and service providers. Consumers are looking for turnkey solutions, while service providers are looking for connectivity providers that offer their subscribers an integrated connectivity solution and a large customer base. The originating domain uses a location server to search for a proxy server that can terminate the call. Proxy servers will only terminate calls that come from known, and thus billable, domains. This represents an opportunity for intermediaries the so-called Internet Telephony Exchange Carriers (ITXC). Ideally ITXCs have agreements with several terminating parties in each country. They can also promise a certain QoS to the originator. Smaller access providers, which don't have the power to negotiate attractive agreements worldwide, can use the ITXC's services. By listing the ITXC's proxy server in the Access Provider's location server, the Access Provider can offer its customers a worldwide Internet telephony service. 5. Integrated service scenario Nowadays PSTN connectivity is indispensable in our personal and professional lives, and data connectivity (Internet or VPN) is swiftly achieving this status. However, dialing in with a modem results in a loss of PSTN connectivity. Basic rate ISDN lines allow simultaneous connectivity, and even permanent connectivity using the "always on dynamic ISDN principle". ADSL and cable modems offer full PSTN and broadband data connectivity. Nevertheless, at some point data and voice still follow different paths and use different transport mechanisms (packet switched vs circuit switched). Also, the two kinds of connectivity are generally provided by different suppliers. Only recently has it become possible to integrate both at the service and connectivity levels. Now that it is feasible to transport voice traffic over packet networks (VoIP), and the necessary protocols and architectures are being developed, the next logical step is to integrate voice (multimedia) and data connectivity and services. It is necessary to provide integration architectures and scenarios with features such as extensibility and reusability, combining them with Intelligent Network concepts. Figure 3 illustrates the concepts of such an integrated system. During the service select scenario, RAS authentication and authorization can be extended to allow phone calls (after a one-time subscription), and to provide automatic registration of the current location with a registrar in the specified domain. Even when the user is connected via a RAS to a specific VPN, setting up or receiving a phone call (within or external to this VPN) can be only one click away. To realize this, a SIP User Agent (Client + Server) has to be enabled or downloaded 4

7 automatically, allowing outgoing and incoming phone calls, and the UAC has to be configured to contact a proxy server in the domain. Naturally, charging for the RAS and the telephone service should be seamlessly integrated and transparent to the user. AAA = Authentication, Authorization & Accounting Server UMS = Unified Messaging System Server PS = Proxy Server LS = Location Server W/Y = White/Yellow Pages UP = User Profile Database AAA PS LS UMS W/Y UP Corporate NW Access NW Telco owned Data NW Internet User NAS Figure 3 SIP and RAS integration Starting from these basic features, a wide range of value-added services could be introduced. A UMS (fax, voice mail, etc) requires a permanent registration (one-time subscription, fixed user and domain name). Registration during connection setup could inform the UMS as to where and whether the user can be reached, allowing the proxy server to redirect incoming calls to the user's current location. Redirection could also be possible when the user connects to a location other than his or her usual domain; this is a typical value-added service which supports personal mobility. Several types of directory and database services should be provided in parallel with the Authentication, Authorization and Accounting (AAA) server. A user profile database should store all user information deemed necessary for seamless operation. A typical example is that the user profile contains information about when, where and how (voice mail, phone, fax, etc) the user can be reached. This information should be easily controlled and updated from the RAS via a user-friendly interface, such as a personalized Web page. Another service could be that if the user sets up a session with a VPN, his or her Internet telephony settings are automatically adjusted for calls within that VPN. 6. Service intelligence in the access point and user terminal Data networks are ideally suited to distributed service intelligence. Consequently, intelligence can be located where it is most needed, namely at the network edge. Furthermore, intelligent nodes can easily be linked and resources shared. Several functional blocks containing valuable user information are already present at the edge. The RAS together with a Service Management Center (SMC) can provide user authentication and authorization, user service subscriptions, and adjustment of the QoS level according to the user s demands. An Access 5

8 Application Server (AAS) or, rather, a set of servers, can be introduced at the network edge to realize various services and applications, such as personalized Web pages for service selection and service management by the end user, directory services, Internet telephony applications, and others. All these services should work seamlessly together, providing a transparent, easy to use and easily maintained system for the user. It should also be easy to deploy new services. During connection/session setup, after authentication the user could automatically be authorized to use his or her subscribed services, some of which (e.g. Internet telephony) could be enabled automatically. 7. Conclusions Pure connectivity is migrating towards a full service offering at the network edge, bringing new opportunities for access providers to create value-added services [1]. One of these services will definitely be Internet telephony, with its inherent business opportunities. Because of their close and trustworthy relationship with users, access providers have a considerable advantage when it comes to deploying and commercializing these new services. Integration of RAS and SIP is one of the key enablers in the move to create service environments offering seamless interoperability and advanced features. This, in turn, can be an important differentiator for an access provider in the "battle for the end user". 8. References [1] J. Pirot: "Implementation of Value Added Services in an Internet Environment", Alcatel Telecommunications Review, 4th Quarter, [2] B. Lloyd, W. Simpson: "PPP Authentication Protocols", IETF- RFC 1334, L&A, DayDreamer, October [3] R. Droms: "DHCP: Dynamic Host Configuration Protocol", IETF-RFC 2131, Bucknell University, March [4] M. Handley, H. Schulzrinne, E. Schooler, J. Rosenberg: "SIP: Session Initiation Protocol", IETF-RFC 2543, March