In order to make use of Internet telephony, a signaling protocol is needed to setup calls. Two signaling protocols have emerged to fill this need: the ITU-T H.323 suite of protocols [84] and the Session Initiation Protocol (SIP) [159] developed by the IETF. A comparison of the two protocols can be found in [170].
SIP is a textual client-server protocol, modeled after the Simple Mail Transfer Protocol (SMTP) [99, 151] and the Hypertext Transfer Protocol (HTTP) [59]. SIP is used to establish, change, and tear down calls between one or more endpoints in an IP-based network. The protocol reuses much of the syntax and semantics of HTTP, including the response code architecture, many message headers, and the overall operation. SIP can be run on top of either TCP or UDP; TCP allows many requests to be sent over the same connection; UDP facilitates fast operation and group communications through multicast. A SIP address is similar to an E-mail address;user name@dns-nameorphone number@IP address, for example.
A SIP system is built of three types of components. The client application is called a User Agent and allows the user to initiate and receive calls. Two different servers handle the call of a user: SIP proxies transparently determine the proper server to be contacted for the specified recipient, and redirect servers enable the forwarding of the call to the current whereabout of the recipient. Any number of these servers may be involved before the final recipient is found. In practice, the operation of these servers are analogous to recursive (proxies) and iterative (redirect servers) searches in the Domain Name System (DNS).
The SIP protocol includes six message types. The INVITE and REGISTER messages are the core messages. INVITE is used to invite a user to a call. The message also includes the intended recipient, information about the codecs, ports and protocols to be used for sending the media stream, for example, parameters that can be used with RTP. The REGISTER message is used by the user to register her current whereabout to a SIP server. In addition, SIP uses ACK messages to provide reliable message exchanges, a BYE message to terminate a call, and CANCEL messages to terminate the latest message exchange without terminating
3.4 User Mobility with the Session Initiation Protocol 45
Table 3.1: Example of SIP mobility support
User Name Terminal Identifier Terminal Location [email protected] 128.214.9.198 4.17.168.56 [email protected] 214.10.11.39 207.46.230.218
an ongoing call. The OPTIONS message is used to solicit information about the capabilities of the recipient.
The current version of SIP supports user mobility readily by proxying and redirecting requests to the current location of the user. A recipient may also change end hosts over time, even during a session. These locations can be dy-namically registered with the SIP server; the REGISTER request allows a client to let a proxy or redirect server know at which address it can be reached. However, this support requires several messages to be sent end-to-end and, therefore, may not give very good support to users moving very fast.
To provide smoother mobility, SIP could use Mobile IP registration services for location update [124]. SIP could run over Mobile IP and take use of the IPsec mechanisms used with Mobile IP updates together with authentication. This so-lution does not add anything to SIP, other than make SIP run on top of Mobile IP.
Another solution would be to enhance the SIP location servers to handle entries presented in Table 3.1.
This scheme would require additional software in the SIP location servers but would allow a more scalable location management service, as it is external. A hierarchy of location servers would be needed to cover larger networks. However, this solution creates additional overhead since two location management schemes are working over each other, SIP for personal and MIP for host mobility. In addi-tion there might be some redundancy in the updating traffic [124].
This user mobility is not in any way tied to the mobility of a terminal, since a user may log on a foreign host and then register her location to her own SIP server. This is different than using the same terminal on the move and register the current location of the physical terminal with a Mobile IP Home Agent.
Chapter 4
Mobile QoS Architectures
This chapter takes a look at key network architectures that provide support for both mobility and service differentiation. The main focus is on mobile networks that provide IP-based data services. Two of the most important telecommunica-tions industry specificatelecommunica-tions are the General Packet Radio Service (GPRS) and the Universal Mobile Telephone System (UMTS). In addition, this chapter intro-duces some interesting architectures studied in the academic research community, namely INSIGNIA, Mobile RSVP and ITSUMO.
4.1 Overview of Mobile Telecommunications
The first wireless networks that provide data services include the Global System for Mobile Telecommunication (GSM) [153], the Cellular Digital Packet Data (CDPD) [161] and the Trans-European Trunked Radio (TETRA) [133, 194].
GSM was originally designed for voice communications and, therefore, the packet data capabilities were limited to a circuit-switched 9.6 kbps connection.
This service has been enhanced with a new coding scheme that increased the bandwidth to 14.4 kbps [44]. The High Speed Circuit Switched Data (HSCSD) [46, 47, 185] can potentially provide a circuit-switched connection bandwidth up to 115 kbps [164], currently 43.2 kbps is available. However, although the speeds seem potentially sufficient for most people, the service is still circuit-switched and, therefore, not the most suitable for bursty data communication, and rather expensive, too.
The CDPD approach is a mobile data technology that permits subordinate packet data operation on the spectrum assigned to a telephone cellular network.
TETRA comprises a comprehensive suite of standards for digital private mo-bile radio (PMR). The system is targeted at government and official use and can transport both voice and packet data at up to 28.8 kbps with some QoS for
multi-47
media services [172].
The Wireless Application Protocol (WAP) [193] has enabled the first ”wireless Internet” services for the GSM architecture. WAP is a protocol specification for any IP-based transport that is meant to provide a data service that can adapt to the limited capabilities of mobile terminals. However, so far WAP services have not been as successful as initially was estimated when the service was launched.
This has been partly due to charging issues and because the services have not been differentiated enough from earlier services based on the GSM Short Message Service (SMS) [49]. To be commercially feasible, WAP requires a GPRS network as the network architecture.
On the other hand, the Japanese I-Mode [43, 104, 149] has been a tremen-dously successful packet data service deployed by NTT DoCoMo [214]. The service is a combination of a sluggish 9.6 kbps packet data connection and a stripped-down version of HTML called compact HTML (cHTML) developed by the Japanese embedded software producer Access [207]. In January 2001 Do-CoMo added Java support to I-Mode allowing applets to be downloaded to mo-bile phones. The example of I-Mode shows that the speed of the connection is not solely the main driver for consumers adopting a new service. It is rather the combination of the properties of the connection, the services that can be provided and the pricing of such services. The European General Packet Radio Service to-gether with the Wireless Application Protocol and Java-applications can provide the same success for European operators.
All these networks and services have been the first step towards mobile data services. The rest of this chapter will look at the main new mobile packet data services, the General Packet Radio Network and Universal Mobile Telecommu-nications System. The end of this chapter takes a look at schemes that can be seen as variations of existing IETF protocols aiming to provide mobility and QoS support.