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Trunk Signalling

Dans le document SIP DEMYSTIFIED (Page 38-44)

Interexchange signalling in circuit-switched networks is referred to as trunk signalling. Two types exist:Channel Associated Signalling (CAS) and Common Channel Signalling(CCS). CAS systems preceded CCS systems, which offer better capacity, reliability, and flexibility, and at present are pro-gressively being replaced by them.

Exchange Local Terminal

(1) SET UP Call attempt is received.

Channel Associated Signalling In CAS systems, the signalling and the voice associated with a certain call are transferred along the same path. In-band signalling, which was described previously, is an example of CAS.

CAS can be used in both analog and digital systems. Digital systems that use CAS implement a specific channel for line signalling. Line signalling controls the establishment and release of voice channels (see Figure 1-15).

The line signalling associated with all voice channels in a trunk is trans-ported in a single channel. European systems, for instance, reserve time slot number 16 in each PCM link. Thus, the sixteenth time slot carries line sig-nalling associated with channels 1 through 15 and 17 through 31 (time slot 0 is used for frame synchronization). Some examples of line signalling are:

idle line, seizure of line, answer, and charging pulses.

On the other hand, register signalling, which handles addressing infor-mation, is transported through the voice path. Some examples of register signals are: callee’s number, callee’s status, and caller’s number. Since reg-ister signals are sent only during call set-up, when there is still no voice transfer, they do not interfere with the user plane.

Exchange Exchange

Voice channel 31 Voice channel 30 Voice channel 18 Voice channel 17 Voice channel 15 Voice channel 14 Voice channel 2 Voice channel 1 Synchronization channel 0

SIGNALLING CHANNEL 16 Figure 1-15

Channels in a PCM link.

Common Channel Signalling (CCS) In CCS, voice and signalling, as a rule, traverse different paths through the network (although their paths are still related to a certain extent). All the nodes handling media also han-dle signalling, so that, if a voice path traverses two exchanges, these two exchanges will also receive the associated signalling. How signalling is routed between them, however, is different in CCS than other systems.

After leaving one exchange in the voice path, signalling can traverse a set of intermediate nodes, none of which handle voice before reaching the next one. We call these intermediate nodes Signalling Transfer Points(STPs).

STPs are a special feature of CSS, necessitating the implementation of a dedicated network optimized for signalling transport, and thus capable of better performance and reliability than shared voice/signal networks.

Dedicated Signalling Network Signalling traffic associated with a call is bursty in nature. It is very intense in certain phases of the call, like call establishment or call release, but otherwise quite low once the call is estab-lished. Bursty is good in that it enables a single signalling channel to con-trol several voice channels. In practice, one signalling channel can handle several thousands of voice time slots.

Implementing a signalling network independent from its voice counter-part enables service creation and makes interworking different networks easier. SS7 is the most widespread CCS system at present, and it is SS7 in particular that allows the creation of services that go beyond basic CAS

STP STP

Voice and signalling Voice and

signalling Voice and

signalling

Signalling path Voice path Figure 1-16

Signalling Transfer Points.

connections. The capability to signal between dedicated network nodes that do not handle voice is in fact the basis of all the advanced services provided by the current PSTN. By implementing service logic and switching logic separately, we encourage service creation and user plane manipulation to evolve independently.

For instance, when someone dials a toll-free number, the system must be able to look up the phone number dialed by the user (it’ll be a 1-800 num-ber in the United States) and translate it into the (geographically deter-mined) number of the company the user wants to reach. This database look up is performed entirely through signalling messages (see Figure 1-17).

Toll-free and other services can be implemented thanks to the separation between the signalling and the voice path. Otherwise, it would be necessary to route the voice path towards the database and then back from it in order to perform the number look-up, leading to a severe decrease in voice qual-ity experienced by the caller.

Today, two main CCS systems are in use: SS6 and SS7. SS6 was stan-dardized in 1968 and was meant to be used in international connections.

SS7 was standardized in 1980 and remains the dominant signalling system at present.

Database (just signalling)

Voice and signalling Voice and

signalling Voice and

signalling

Signalling path 1-800-55-5555

1-212-555-5544

Voice path Figure 1-17

User calling a toll free number.

SS7

SS7 provides both circuit-related and non-circuit-related signalling.

Circuit-related signalling is, as its name suggests, associated with the establishment of a voice channel. But frequently no voice channel has to be established—as when the system consults a routing table—and this kind of query can be handled by circuit-related signalling. As an enabler, non-circuit signalling was a vital step toward mobility.

Let’s first look at circuit related signalling.ISDN User Part(ISUP) and Telephone User Part(TUP) are the relevant SS7 protocols in this regard. In the last specifications, ISUP included all the functionality supported by TUP, plus more features such as the capability to transfer user-to-user information, so ISUP is sufficient for our discussion here. ISUP is actually a generic name for a protocol which has many flavors in many different countries. When carriers want to implement a new feature, they often use non-compatible extensions that cause interoperability problems, and in so doing, create a variety of local ISUP flavors. Naturally, national variants of ISUP are incompatible, so an international version of ISUP is used to inter-connect networks with ISUP variants. Gateway switches between them understand/support both international ISUP and the specific ISUP imple-mented in the network and perform translations between them.

The drawback of this arrangement is that international ISUP is com-paratively feature-poor, limiting network services to users connected to the same network. For example, users in Spain must choose among an avail-able set of services for calls within Spain, because the signalling protocol used for those calls is the Spanish ISUP. Users in Brazil have another set of services available for their internal calls. But services can’t extend to calls between both countries, because there’s no feature transparency between both flavors.

Supplementary Services Functionality provided by ISUP can be divided into three groups: basic services, circuit management, and supple-mentary services. Basic services include the basic call establishment to pro-vide voice service.

ISUP circuit management functions include blocking and unblocking cir-cuits and establishing, releasing, and testing circuit-switched paths. The same protocol used for establishing calls is also used for managing network resources.

Supplementary services are more varied; examples are call waiting, call transfer, and conferencing. These supplementary services are undeniably useful and many users take advantage of them. However, gaining access to them is pretty complicated. Because the interface used for this purpose is usually the keypad of a telephone set, the user has to learn different codes for different services. If the service is not accessed very often, remembering codes becomes difficult. A non-atypical example of service invocation is typ-ing “*22telephone number ” in order to turn on call forwarding, compli-cated enough that it will probably require the user to look up procedures before nearly every forward. We will see that the introduction of better user interfaces not only makes for more frequent usage, but also makes protocols evolve more rapidly, since the users can use newly introduced features in a more friendly way.

ISUP supplementary services are implemented in a distributed manner.

Users gain access to new services when the local exchange to which they are connected is upgraded to support new features. Therefore, service avail-ability depends on the local exchange and its business plan. Moreover, not all users of the network have access to the same services since all the local exchanges of the network cannot be upgraded at the same time and with the same version of software and the same hardware. Even users accessing the same service might expect different performance behavior, depending on which switch they are requesting the service from.

Intelligent Network (IN) Services These limitations of ISUP with respect to service provision prompted implementation of the Intelligent Net-work(IN) services, which utilizes Intelligent Network Application Protocol (INAP). Although ISUP services were distributed, IN services are clustered

Spanish ISUP Brazilian ISUP

in a centrally located node. This node, called Service Control Point(SCP), can be accessed by any user in the network, so providers can deliver new services simply by upgrading the node—a big advance in service manage-ment. Services are implemented in the SCP with scripts—a set of rules and instructions that contain the particular service logic. Scripts come in two types: system and group. The system script performs number analysis in response to the number entered by a user, and then calls the proper group script to execute the actual service logic.

Distributed throughout the network are a number of IN service triggers called Service Switching Points (SSPs). An SSP analyzes the number entered by the user and decides whether intelligent network services are needed. It then contacts the SCP, which in turn determines the actions required to provision the service invoked.

Dans le document SIP DEMYSTIFIED (Page 38-44)