Signalling System #7 (SS7)
Signalling System #7 (SS7)
There are two essential components to all telephone calls. The first, and most obvious, is the actual contentour voices, faxes, modem data, etc. The second is the information that instructs telephone exchanges to establish connections and route the content to an appropriate destination. Telephony signaling is concerned with the creation of standards for the latter to achieve the former. These standards are known as protocols. SS7 or Signaling System Number 7 is simply another set of protocols that describe a means of communication between telephone switches in public telephone networks. They have been created and controlled by various bodies around the world, which leads to some specific local variations, but the principal organization with responsibility for their administration is the International Telecommunications Union or ITU-T. Signalling System Number 7 (SS#7 or C7) is the protocol used by the telephone companies for interoffice signalling. In the past, in-band signalling techniques were used on interoffice trunks. This method of signalling used the same physical path for both the call-control signalling and the actual connected call. This method of signalling is inefficient and is rapidly being replaced by out-of-band or commonchannel signalling techniques. To understand SS7 we must first understand something of the basic inefficiency of previous signaling methods utilized in the Public Switched Telephone Network (PSTN). Until relatively recently, all telephone connections were managed by a variety of techniques centered on in band signaling. A network utilizing common-channel signalling is actually two networks in one: 1. First there is the circuit-switched "user" network which actually carries the user voice and data traffic. It provides a physical path between the source and destination. 2. The second is the signalling network which carries the call control traffic. It is a packet-switched network using a common channel switching protocol.
The original common channel interoffice signalling protocols were based on Signalling System Number 6 (SS#6). Today SS#7 is being used in new installations worldwide. SS#7 is the defined interoffice signalling protocol for ISDN. It is also in common use today outside of the ISDN environment. The primary function of SS#7 is to provide call control, remote network management, and maintenance capabilities for the interoffice telephone network. SS#7 performs these functions by exchanging control messages between SS#7 telephone exchanges (signalling points or SPs) and SS#7 signalling transfer points (STPs). The switching offices (SPs) handle the SS#7 control network as well as the user circuit-switched network. Basically, the SS#7 control network tells the switching office which paths to establish over the circuitswitched network. The STPs route SS#7 control packets across the signalling network. A switching office may or may not be an STP. SS7 Protocol layers: The SS7 network is an interconnected set of network elements that is used to exchange messages in support of telecommunications functions. The SS7 protocol is designed to both facilitate these functions and to maintain the network over which they are provided. Like most modern protocols, the SS7 protocol is layered.
Physical Layer (MTP-1) This defines the physical and electrical characteristics of the signaling links of the SS7 network. Signaling links utilize DS0 channels and carry raw signaling data at a rate of 56 kbps or 64 kbps (56 kbps is the more common implementation). Message Transfer PartLevel 2 (MTP-2) The level 2 portion of the message transfer part (MTP Level 2) provides link-layer functionality. It ensures that the two end points of a signaling link can reliably exchange signaling messages. It incorporates such capabilities as error checking, flow control, and sequence checking. Message Transfer PartLevel 3 (MTP-3) The level 3 portion of the message transfer part (MTP Level 3) extends the functionality provided by MTP level 2 to provide network layer functionality. It ensures that messages can be delivered between signaling points across the SS7 network regardless of whether they are directly connected. It includes such capabilities as node addressing, routing, alternate routing, and congestion control. Signaling Connection Control Part (SCCP) The signaling connection control part (SCCP) provides two major functions that are lacking in the MTP. The first of these is the capability to address applications within a signaling point. The MTP can only receive and deliver messages from a node as a whole; it does not deal with software applications within a node. While MTP network-management messages and basic call-setup messages are addressed to a node as a whole, other messages are used by separate applications (referred to as subsystems) within a node. Examples of subsystems are 800 call processing, calling-card processing, advanced intelligent network (AIN), and custom local-area signaling services (CLASS) services (e.g., repeat dialing and call return). The SCCP allows these subsystems to be addressed explicitly.
ISDN User Part (ISUP) ISUP user part defines the messages and protocol used in the establishment and tear down of voice and data calls over the public switched network (PSN), and to manage the trunk network on which they rely. Despite its name, ISUP is used for both ISDN and nonISDN calls. In the North American version of SS7, ISUP messages rely exclusively on MTP to transport messages between concerned nodes. Transaction Capabilities Application Part (TCAP)
TCAP defines the messages and protocol used to communicate between applications (deployed as subsystems) in nodes. It is used for database services such as calling card, 800, and AIN as well as switchto-switch services including repeat dialing and call return. Because TCAP messages must be delivered to individual applications within the nodes they address, they use the SCCP for transport. Operations, Maintenance, and Administration Part (OMAP) OMAP defines messages and protocol designed to assist administrators of the SS7 network. To date, the most fully developed and deployed of these capabilities are procedures for validating network routing tables and for diagnosing link troubles. OMAP includes messages that use both the MTP and SCCP for routing.
Application Part, Location Services Extension (BSSAP-LE). In addition, SCCP provides Global Title Translation (GTT) functionality. The signaling connection control part (SCCP) provides two major functions that are lacking in the MTP. The first of these is the capability to address applications within a signaling point. The MTP can only receive and deliver messages from a node as a whole; it does not deal with software applications within a node. While MTP network-management messages and basic call-setup messages are addressed to a node as a whole, other messages are used by separate applications (referred to as subsystems) within a node. Examples of subsystems are 800 call processing, calling-card processing, advanced intelligent network (AIN), and custom local-area signaling services (CLASS) services (e.g., repeat dialing and call return). The SCCP allows these subsystems to be addressed explicitly. The signaling connection control part (SCCP) provides two major functions that are lacking in the MTP. The first of these is the capability to address applications within a signaling point. The MTP can only receive and deliver messages from a node as a whole; it does not deal with software applications within a node. While MTP network-management messages and basic call-setup messages are addressed to a node as a whole, other messages are used by separate applications (referred to as subsystems) within a node. Examples of subsystems are 800 call processing, calling-card processing, advanced intelligent network (AIN), and custom local-area signaling services (CLASS) services (e.g., repeat dialing and call return). The SCCP allows these subsystems to be addressed explicitly. The second function provided by the SCCP is Global Title translation, the ability to perform incremental routing using a capability called global title translation (GTT). GTT frees originating signaling points from the burden of having to know every potential destination to which they might have to route a message. A switch can originate a query, for example, and address it to an STP along with a request for GTT. The receiving STP can then examine a portion of the message, make a determination as to where the message should be routed, and then route it. For example, calling-card queries (used to verify that a call can be properly billed to a calling card) must be routed to an SCP designated by the company that issued the calling card. Rather than maintaining a nationwide database of where such queries should be routed (based on the calling-card number), switches generate queries addressed to their local STPs, which, using GTT, select the correct destination to which the message should be routed. Note that there is no magic here; STPs must maintain a database that enables them to determine where a query should be routed. GTT effectively centralizes the problem and places it in a node (the STP) that has been designed to perform this function.
In performing GTT, an STP does not need to know the exact final destination of a message. It can, instead, perform intermediate GTT, in which it uses its tables to find another STP further along the route to the destination. That STP, in turn, can perform final GTT, routing the message to its actual destination. Intermediate GTT minimizes the need for STPs to maintain extensive information about nodes that are far removed from them. GTT also is used at the STP to share load among mated SCPs in both normal and failure scenarios. In these instances, when messages arrive at an STP for final GTT and routing to a database, the STP can select from among available redundant SCPs. It can select an SCP on either a priority basis (referred to as primary backup) or so as to equalize the load across all available SCPs (referred to as load sharing).
Call connection: When the called party picks up the phone, the destination switch terminates the ringing tone and transmits an ISUP answer message (ANM) to the originating switch via its home STP. The STP routes the ANM to the originating switch which verifies that the calling party's line is connected to the reserved trunk and, if so, initiates billing. Call tear down: If the calling party hangs-up first, the originating switch sends an ISUP release message (REL) to release the trunk circuit between the switches. The STP routes the REL to the destination switch. If the called party hangs up first, or if the line is busy, the destination switch sends an REL to the originating switch indicating the release cause (e.g., normal release or busy). Upon receiving the REL, the destination switch disconnects the trunk from the called party's line, sets the trunk state to idle, and transmits an ISUP release complete message (RLC) to the originating switch to acknowledge the release of the remote end of the trunk circuit. When the originating switch receives (or generates) the RLC, it terminates the billing cycle and sets the trunk state to idle in preparation for the next call.