Signaling System No.7 Protocol Architecture And Sevices part 14 potx

The core network interface connections for VoIP into the PSTN are the trunk facilities that carry the voice channels and the signaling links that carry SS7 signaling.. MTP2 performs the

Integration of SS7 into the PSTN

This section provides a brief overview of how the SS7 architecture is applied to the PSTN Since SS7 has not been presented in great detail, the examples and

information are brief and discussed only in the context of the network nodes

presented in this section

The PSTN existed long before SS7 The network's general structure was already in place, and it represented a substantial investment The performance requirements mandated by the 800 portability act of 1993 was one of the primary drivers for the initial deployment of SS7 by ILECs in the United States IXCs embraced SS7 early

to cut down on post-dial delay which translated into significant savings on

access/egress charges Federal regulation, cost savings, and the opportunity to provide new revenue generating services created a need to deploy SS7 into the existing PSTN

SS7 was designed to integrate easily into the existing PSTN, to preserve the

investment and provide minimal disruption to the network During SS7's initial deployment, additional hardware was added and digital switches received software upgrades to add SS7 capability to existing PSTN nodes In the SS7 network, a digital switch with SS7 capabilities is referred to as a Service Switching Point (SSP) When looking at the SS7 network topologies in later chapters, it is

important to realize that the SSP is not a new node in the network

Instead, it describes an existing switching node, to which SS7 capabilities have been added Similarly, SS7 did not introduce new facilities for signaling links, but used timeslots on existing trunk facilities PSTN diagrams containing End Offices and tandems connected by trunks represent the same physical facilities as those of SS7 diagrams that show SSP nodes with interconnecting links The introduction of SS7 added new nodes, such as the STP and SCP; however, all of the switching nodes and facilities that existed before SS7 was introduced are still in place Figure 5-12 shows a simple view of the PSTN, overlaid with SS7-associated signaling capabilities

Figure 5-12 SS7 Overlaid onto the PSTN

[View full size image]

View a in the previous figure shows that trunk facilities provide the path for voice and in-band signaling View b shows the SS7 topology using simple associated signaling for all nodes View c shows the actual SS7-enabled PSTN topology The existing switching nodes and facilities are enhanced to provide basic SS7 call

processing functionality Although this associated signaling architecture is still quite common in Europe, the United States primarily uses a quasi-associated

signaling architecture

SS7 Link Interface

The most common method for deploying SS7 links is for each link to occupy a timeslot, such as a T1 or E1, on a digital trunk As shown in Figure 5-12, the

signaling links actually travel on the digital trunk transmission medium throughout the network At each node, the SS7 interface equipment must extract the link

timeslot from the digital trunk for processing This process is typically performed using a channel bank, or a Digital Access and Cross-Connect (DAC), which

demultiplexes the TDM timeslot from the digital trunk The channel bank, or DAC, can extract each of the timeslots from the digital stream, allowing them to be

processed individually The individual SS7 link provides the SS7 messages to the digital switch for processing While implementations vary, dedicated peripheral processors usually process the lower levels of the SS7 protocol (Level 1, Level 2, and possibly a portion of Level 3); call- and service-related information is passed

on to the central processor, or to other peripheral processors that are designed for handling call processing–related messages Of course, this process varies based on the actual equipment vendor

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Evolving the PSTN to the Next Generation

The expansion of the Internet continues to drive multiple changes in the PSTN environment First, more network capacity is used to transport data over the PSTN Dial-up Internet services use data connections that are set up over the PSTN to carry voice-band data over circuit-switched connections This is a much different situation than sending data over a data network Data networks use packet

switching, in which many data transactions share the same facilities

Circuit-switched connections are dedicated connections, which occupy a circuit for the duration of a call The phone networks were originally engineered for the three-minute call, which was the average length used for calculations when engineering the voice network Of course, Internet connections tend to be much more lengthy, meaning that more network capacity is needed The changes driven by the Internet, however, reach much further than simply an increase in network traffic Phone traffic is being moved to both private packet-based networks and the public

Internet, thereby providing an alternative to sending calls over the PSTN Several different architectures and protocols are competing in the VoIP market to establish alternatives to the traditional circuit-switched network presented in this chapter The technologies are not necessarily exclusive; some solutions combine the

various technologies Among the current leading VoIP technologies are:

• Soft switches

• H.323

• Session Initiation Protocol (SIP)

Each of these VoIP architectures use VoIP-PSTN gateways to provide some means

of communication between the traditional PSTN networks and VoIP networks These gateways provide access points for interconnecting the two networks,

thereby creating a migration path from PSTN-based phone service to VoIP phone service The core network interface connections for VoIP into the PSTN are the trunk facilities that carry the voice channels and the signaling links that carry SS7 signaling PRI is also commonly used for business to network access Figure 5-13 shows the interconnection of VoIP architectures to the PSTN using signaling

gateways and trunking gateways Chapter 14, "SS7 in the Converged World," discusses these VoIP technologies in more detail

Figure 5-13 VoIP Gateways to the PSTN

[View full size image]

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Summary

This chapter provides an overview of the PSTN, as it existed before VoIP

technologies emerged The majority of the PSTN still appears as this chapter

presents it Many of the diagrams in telecommunications literature illustrating next generation technologies—such as soft switches, H.323, and Session Initial Protocol (SIP)—show interfaces to the PSTN The diagrams refer to the PSTN discussed here, dominated by large, digital switches The technologies introduced often

replace some portion of the existing PSTN; however, they must also remain

connected to the existing PSTN to communicate with the rest of the world The VoIP-PSTN gateways provide this transition point, thus enabling a migration path from the traditional PSTN to the next generation architecture

While the PSTN varies in its implementation from country to country, a number of common denominators exist The PSTN is a collection of digital switching nodes that are interconnected by trunks The network topology is usually a hierarchical structure, but it often incorporates some degree of mesh topology The topology provides network access to residential and business subscribers for voice and data services VoIP began another evolution of the PSTN architecture The PSTN is a large infrastructure that will likely take some time to completely migrate to the next generation of technologies; but this migration process is underway

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Chapter 6 Message Transfer Part 2 (MTP2)

This chapter is the first in a series of chapters that examine a specific SS7/C7

protocol layer This chapter details the Layer 2 protocol, which is known as

Message Transfer Part 2 (MTP2) MTP2 corresponds to OSI Layer 2 (the data link layer) and as such is the lowest protocol in the stack Sitting on the physical layer,

it provides a reliable means of transfer for signaling information between two directly connected signaling points (SPs), ensuring that the signaling information is delivered in sequence and error-free

MTP2 performs the following functions:

• Delimitation of signal units

• Alignment of signal units

• Signaling link error detection

• Signaling link error correction by retransmission

• Signaling link initial alignment

• Error monitoring and reporting

• Link flow control

The signaling information is transmitted in frames called signal units (SUs) SUs are of variable length, thereby requiring the start and end of each SU to be flagged

in the data stream MTP2 performs this function, which is called signal unit

delimitation The ability to correctly recognize signal units is achieved through signal unit alignment

Error correction is implemented by retransmitting the signal unit(s) received in error The link is also continuously monitored to ensure that error rates are within permissible limits If the error rate becomes greater than predefined limits, MTP2 reports the failure to Message Transfer Part 3 (MTP3), which subsequently orders MTP2 to remove the link from service Conversely, initial alignment procedures are used to bring links into service

Link flow control procedures are provided to resolve congestion at the MTP2

layer Congestion occurs if MTP3 falls behind in processing SUs from the MTP2 buffer

This chapter describes each of the previously outlined functional areas of MTP2

It is important to understand that the MTP2 protocol does not work end to end Rather, it operates on a link-by-link basis (known in datacoms as point to point) between two SPs Therefore, each signaling data link has an associated MTP2 at each end

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Signal Unit Formats

SUs transfer information, which originates from higher layers (MTP3, ISUP,

SCCP, TUP, and so on) in the form of messages, over the signaling link MTP2 is similar to data network bit-oriented link protocols such as HDLC, SDLC, and LAPB The primary difference with these protocols comes from the performance requirements in terms of lost and out-of-sequence messages and delay

There are three types of SUs, each with its own format: the fill-in signal unit

(FISU), the link status signal unit (LSSU), and the Message Signal Unit (MSU)

An in-service signaling link carries a continuous SU stream in each direction

FISUs and LSSUs are used only for MTP2 functions MSUs also contain the same MTP2 fields, but they have two additional fields filled with information from MTP3 and Level 4 users that contain the real signaling content This chapter

describes the MTP2 fields and the functions they perform It begins by presenting the three SU formats

NOTE

The formats shown are for 64-kbps links The formats for high-speed (1.5/2.0 Mbps) signaling links might differ slightly in that the sequence number might be extended to 12 bits More details are available in Annex A of ITU-T Q.703 [51]

Fill-In Signal Units

FISUs are the most basic SU and carry only MTP2 information They are sent when there are no LSSUs or MSUs to be sent, when the signaling link would

otherwise be idle Sending FISUs ensures 100 percent link occupancy by SUs at all times A cyclic redundancy check (CRC) checksum is calculated for each FISU, allowing both signaling points at either end of the link to continuously check

signaling link quality This check allows faulty links to be identified quickly and taken out of service so that traffic can be shifted to alternative links, thereby

helping meet the SS7/C7 network's high availability requirement Because MTP2

is a point-to-point protocol, only the MTP2 level of adjacent signaling points

exchanges FISUs

The seven fields that comprise a FISU, shown in Figure 6-1, are also common to

LSSUs and MSUs MTP2 adds the fields at the originating signaling point and processes and removes them at the destination signaling point (an adjacent node)

Figure 6-1 FISU Format

Link Status Signal Units

LSSUs carry one or two octets of link status information between signaling points

at either end of a link The link status controls link alignment, indicates the link's status, and indicates a signaling point's status to the remote signaling point The presence of LSSUs at any time other than during link alignment indicates a fault— such as a remote processor outage or an unacceptably high bit error rate affecting the ability to carry traffic

The timers associated with a particular status indication govern the transmission interval After the fault is cleared, the transmission of LSSUs ceases, and normal traffic flow can continue As with FISUs, only MTP2 of adjacent signaling points exchanges LSSUs LSSUs are identical to FISUs, except that they contain an additional field called the Status field (SF) Figure 6-2 shows the eight fields that comprise an LSSU

Figure 6-2 LSSU Format

Currently only a single-octet SF is used, even though the specifications allow for a two-octet SF From the single octet, only the first 3 bits are defined These bits provide the status indications shown in Table 6-1

Table 6-1 Values in the Status Field

0 0 0 O: Out of Alignment SIO Link not aligned; attempting

alignment

0 0 1 N: Normal Alignment SIN Link is aligned

0 1 0 E: Emergency

Alignment

SIE Link is aligned

0 1 1 OS: Out of Service SIOS Link out of service; alignment

failure

1 0 0 PO: Processor Outage SIPO MTP2 cannot reach MTP3

Message Signal Units

As shown in Figure 6-3, MSUs contain the common fields of the FISU and two additional fields: the Signaling Information Field (SIF) and the Service

Information Octet (SIO) MSUs carry the signaling information (or messages) between both MTP3 and Level 4 users The messages include all call control, database query, and response messages In addition, MSUs carry MTP3 network management messages All messages are placed in the SIF of the MSU

Figure 6-3 MSU Format [View full size image]

MTP2 Overhead

Figure 6-4 shows an MSU The MTP2 overhead is exactly the same for both

LSSUs and FISUs, except that an LSSU has an SF

Figure 6-4 Fields Created and Processed by MTP2

Field Descriptions

Table 6-2 details the fields that are found inside the signal units MTP2 exclusively

processes all fields except the SIO and the SIF

Table 6-2 Field Descriptions

Field Length

in Bits

Description

Flag 8 A pattern of 011111110 to indicate the start and end of an SU BSN 7 Backward sequence number Identifies the last correctly

received SU

BIB 1 Backward indicator bit Toggled to indicate an error with the

received SU

FSN 7 Forward sequence number Identifies each transmitted SU FIB 1 Forward indicator bit Toggled to indicate the retransmission of

an SU that was received in error by the remote SP

LI 6 Length indicator Indicates how many octets reside between

itself and the CRC field The LI field also implies the type of signal unit LI = 0 for FISUs, LI = 1 or 2 for LSSUs, and LI >2 for MSUs

SF 8 to 16 Status field Provides status messages in the LSSU only

CK 16 Check bits Uses CRC-16 to detect transmission errors

SIO 8 Service Information Octet Specifies which MTP3 user has

placed a message in the SIF

SIF 16 to

2176

Signaling Information Field Contains the "real" signaling content The SIF is also related to call control, network management, or databases query/response