CCS Has Evolved To Address The Limitations of The CAS Signaling Method. CCS Has The Following Advantages Over CAS
CCS Has Evolved To Address The Limitations of The CAS Signaling Method. CCS Has The Following Advantages Over CAS
The exchange of information (other than by speech) specifically concerned with the establishment,
release and other control of calls, and network management, in automatic telecommunications
operation
CCS has evolved to address the limitations of the CAS signaling method. CCS has the following
advantages over CAS:
Greater flexibility
Capacity to evolve
Most CCS calls can be set up in half the time it takes to set up CAS calls. CCS achieves greater call
control because no contention exists between signaling and user traffic as it does with in-band CAS.
Because the subscriber cannot generate particular signals intended for inter-switch (core network)
signaling, CCS offers a greater degree of protection against fraud than analog CAS methods.
CCS links can be a single point of failure—a single link can control thousands of voice circuits, so if a
link fails and no alternative routes are found, thousands of calls could be lost.
There is no inherent testing of speech path by call set-up signaling, so elaborate Continuity Test
procedures are required.
SS7 network nodes are called signaling points (SPs). Each SP is addressed by an integer called a point
code (PC).
SPs are connected to each other by signaling links over which signaling takes place. The bandwidth
of a signaling link is normally 64 kilobits per second (kbps).
. In recent years, high-speed links have been introduced that use an entire 1.544 Mbps T1 carrier for
signaling. Links are typically engineered to carry only 25 to 40 percent of their capacity so that in
case of a failure, one link can carry the load of two.
To provide more bandwidth and/or for redundancy, up to 16 links between two SPs can be used.
Links between two SPs are logically grouped for administrative and load-sharing reasons. A logical
group of links between two SP is called a linkset.
A Signal Transfer Point (STP) is responsible for the transfer of SS7 messages between other SS7
nodes, acting somewhat like a router in an IP network.
An STP is neither the ultimate source nor the destination for most signaling messages. Generally,
messages are received on one signaling link and are transferred out another. The only messages that
are not simply transferred are related to network management and global title translation.
· Standalone STP
Integrated STPs combine the functionality of an SSP and an STP. They are both the source and
destination for MTP user traffic. They also can transfer incoming messages to other nodes.
A Service Switching Point (SSP) is a voice switch that incorporates SS7 functionality. It processes
voice-band traffic (voice, fax, modem, and so forth) and performs SS7 signaling. All switches with SS7
functionality are considered SSPs regardless of whether they are local switches (known in North
America as an end office) or tandem switches.
An SSP can originate and terminate messages, but it cannot transfer them. If a message is received
All nodes in the SS7 network are called signaling points. A signaling point has the capability to perform
message discrimination (read the address and determine if the message is for that node) and to route SS7
messages to another signaling point. Every signaling point has a unique address called a point code
The SSP function uses the information provided by the calling party (such as dialed digits) to determine how to connect a
call. A routing table in the switch itself will identify which trunk circuit or Transmission Control Protocol (TCP) socket to
use to connect the call and at which switch this trunk terminates. An SS7 message must be sent to this adjacent switch
requesting a circuit connection on the specified trunk or socket. The circuit identification [referred to as the circuit
identification code (CIC)], the calling and called telephone numbers, and information about the voice transmission method
used are in this SS7 message. There is also information about the type of call, the type of decoding used in the voice
transmission, and possibly any switch features needed during the call.
The adjacent switch grants permission to connect this trunk or socket by sending back an acknowledgment to the
originating switch. Using the called-party information in the setup message, the adjacent switch then can determine
how to connect the call to its final destination. The same process is followed using a setup message to any adjacent
switches and circuits connecting those switches. The entire call may require several connections between several
switches. The SSP function in each switch manages these connections but really has no knowledge of the status of
remote connections (nonadjacent connections). The SSP only has visibility of its own connections and does not
maintain the status of all the connections needed to connect and maintain a call.
Very few SS7 features are required of an SSP. The capability to send messages using the ISDN User Part (ISUP)
protocol and the Transaction Capabilities Application Part (TCAP) protocol is the only requirement besides the
network management functions. SSPs are responsible for the management of ISUP messages specifically,
which may include different variants of ISUP standards.
The STP only processes the transport layers [Message Transfer Part (MTP) or TCP/IP-based protocols]. The STP
routes SS7 messages as received from the various SSPs throughout the network to their appropriate
destinations
Quasi-associated signaling (Figure 2.6) uses a minimal number of nodes to reach the final
destination. This is the most favorable method of signaling because each node introduces additional delays in
signaling delivery. For this reason, SS7 networks favor quasi-associated signaling.
Links are placed into groups called linksets. All the links in a linkset must have the same adjacent node. The
switching equipment alternates transmission across all the links in a linkset to ensure equal use of all facilities.
Up to 16 links can be assigned to one linkset.
In addition to linksets, a signaling point must define routes. A route is a collection of linksets used to reach a
particular destination. A linkset can belong to more than one route. A collection of routes is known as a routeset.
A routeset is assigned to a destination. Routesets are necessary because if only a single route existed and that
route became unavailable, an alternate route would not be defined, and no signaling could be sent to that
destination. A routeset provides alternate routes to the same destination in the event that any one route
becomes unavailable.
C-links (Figure 2.14) connect an STP to its mate STP. C-links are always deployed in pairs to maintain
redundancy on the network. Normal SS7 traffic is not routed over these links, except in congestion conditions.
The only messages to travel between mated STPs during normal conditions are network management messages.
If a node becomes isolated and the only available path is over the C-links, then normal SS7 messages can be
routed over these links. A maximum of eight C-links can be deployed between STP pairs.
D-links (Figure 2.15) are used to connect mated STP pairs at a primary hierarchical level to another STP mated
pair at a secondary hierarchical level. For example, a carrier may have STPs deployed in every Local Access
Transport Area (LATA). A maximum of eight D-links can be used between two mated STP pairs.
(Figure 2.16) are used to connect to remote STP pairs from an SSP. The SSP connects to its home STP
pair but, for diversity, also may be connected to a remote STP pair using E-links. E-links then become the
alternate route for SS7 messages in the event that congestion occurs within the home STP pairs. A
(Figure 2.17) are used when a large amount of traffic exists between two SSPs or when an SSP cannot be
connected directly to an STP. F-links enable SSPs to use the SS7 protocol and access SS7 databases even
TUP and ISUP both perform the signaling required to set up and tear down telephone calls. As such,
both are circuit-related signaling protocols. TUP was the first call control protocol specified. It could
support only plain old telephone service (POTS) calls. Most countries are replacing TUP with ISUP.
ISUP supports both POTS and ISDN calls. It also has more flexibility and features than TUP.
1.1.1 MTP
MTP levels 1 through 3 are collectively referred to as the MTP. The MTP comprises the functions to
transport information from one SP to another.
The MTP transfers the signaling message, in the correct sequence, without loss or duplication,
between the SPs that make up the SS7 network. The MTP provides reliable transfer and delivery of
signaling messages. The MTP was originally designed to transfer circuit-related signaling because no
noncircuit-related protocol was defined at the time.
The recommendations refer to MTP1, MTP2, and MTP3 as the physical layer, data link layer, and
network layer, respectively.