Section 4.2. Components of the PSTN


4.2. Components of the PSTN

The Public Switched Telephone Network and its signaling counterpart , SS7, connect, monitor, bill, and disconnect calls. At the edge of the PSTN are large, mainframe-like switches call exchanges, or central offices (COs). The role of each CO switch is to connect calls between channels on that switch and, when necessary, to connect calls to channels on other switches in the PSTN.

Calls travel along temporary pathways through the voice network. Though temporary, these pathways are end-to-end circuits, the root of the catchall moniker for legacy call management: circuit switching .

4.2.1. The Central Office

The CO is the building where the local exchange switch resides. A CO's switch may serve telephone service subscribers in a very narrow geographic areasuch as a single large building. Or the CO's switch may serve subscribers for miles around. The CO's scope of service depends on the density of subscribers in its neighborhood and on the capacity of the switch it houses . As with many electronic services, capacity and efficiency have increased over time, so late-model CO switches are generally able to provide more channels and greater utilization ceilings than older ones.

Conventionally, you can tell which central office subscribers belong to by looking at the first three digits of their seven-digit telephone number. This three-digit section is called the prefix. One prefix is usually set aside for one central office, or for groups of small, geographically close COs. This tradition is disappearing , however. Newer signaling protocols and networking standards have made the relationship between the CO and the prefix less of a requirement and more of a historical artifact.

4.2.2. PSTN Distribution Frames

Distribution frames surround the CO. They are high-density cross-connection points where multiple subscribers' loops are tapped into the feed cables (that is, 50 to 800 pairs in a single cable) supplied by the CO. Usually, all connections to a distribution frame are copper . Distribution frames allow the telephone company to use high-density copper cabling that is less susceptible to breakage to feed groups of subscribers on the edge with a connection to the CO.

Those subscribers usually have a low-density (2-25 pairs) copper cable running from their buildings to the distribution frame. The frame is the aggregation point for many customers in a small area. One customer can use a CO loop that was once used by another customer due to the cross-connect terminals located inside the distribution frame. New buildings can be built in the area, and they can tap into the distribution frame rather than having to be connected directly, and usually over a longer distance, to the CO.

4.2.3. Main Distribution Frames

MDFs are distribution frames that have some smart switching or encoding equipment in them. Usually, this equipment has been placed there in order to facilitate the use of DSL (digital subscriber line) for access to the Internet by telephone company customers. MDFs can also be used to aggregate large bunches of customer copper loops into super-high-capacity, often optical, links that connect to the CO instead of the high-density copper feeds to the CO used in regular distribution frames (see Figure 4-2).

4.2.4. Switch-to-Switch Trunks

A link between two switches is called a trunk. A switch-to-switch trunk is one that carries calls between two switches on the PSTN. When a call from a particular switch's subscriber is destined for a subscriber on a different switch on the PSTN, one or more switch-to-switch trunks will be used to create a circuit for that call.

The trunks that connect a PBX to the local phone company are also considered switch-to-switch trunks, since the PBX itself is a switch. Large PSTN trunks, like those between very busy COs, may consist of synchronous optical networking (SONET) connections, or DS3 connections, which are described in greater detail later in this chapter (see Figure 4-3).

4.2.5. In- Band Signaling (DTMF)

DTMF tones are the sounds you hear when you press the numbered keys on a telephone keypad. The tones are used as signals to the switch, telling it which phone number you're trying to call or to access a telephony feature such as automatic call return (discussed more in Chapter 5). DTMF stands for dual-tone multifrequency

Figure 4-2. On the PSTN, the "last mile" of the loop is connected using copper wire pairs and distribution frames

Figure 4-3. Connections that carry phone calls between phone switches on the PSTN or a private voice network are called trunks

signaling. Its tones consist of two pitches. DTMF is in-band signaling because the dual-tone signals occupy the same frequency band that is used to carry the voice sounds.

DTMF signaling replaced the electromechanical pulse-dial signaling that was prevalent as late as the late 1980s. The standard for DTMF is described by the International Telecommunication Union's Q.23 and Q.24 specs .

4.2.5.1 Standard tones

Sounds that you hear when the switch needs to notify the caller (or receiver) the progress of the call are called standard tones . These include the busy tone, the ringback tone (the ringing the caller hears while waiting for his call to be connected), and the dial-tone. Standard tones are also defined by the ITU, in specs E.180 and Q.35.

4.2.5.2 City codes, area codes, and country codes

COs are grouped into logical, compact geographic areas that are large enough to support somewhere around five or six million phone numbers apiece. These groups, known commonly as area codes, are used to help the PSTN and SS7 route calls between subscribers in remote areas of the network. In this way, a caller from Boston can reach a caller in Tampa and so forth. Area codes are prepended onto the local CO portion of the phone number in order to place interarea phone calls. Country codes are similar prefixes, but for calls between different national PSTN designations, like the United States and Japan. City codes are a variation on area codes and are mainly used outside of North America. More information about phone numbers and international designations can be found in ITU recommendation E.164.

4.2.6. Out-of-Band Signaling and SS7

Common Signaling System 7, also called SS7 or C7, was developed by the ITU Telecommunications Standardization Sector in order to increase the efficiency of the public voice system. SS7 is a separate network whose duties are setting up, tearing down, monitoring, and routing calls on the PSTN.

SS7 is akin to TCP/IP in that it operates at several layers of the OSI model. And, like TCP/IP, SS7 is packet-based. It is the conceptual basis for VoIP call signaling, because it is a software-based system that operates independently of the voice transport itself (the PSTN).

SS7 works behind the scenes, so interacting with SS7 is something that the CO switch, not your phone or PBX, must do. SS7 is called an out-of-band signaling standard because, unlike DTMF, it doesn't use the same frequency band, or even the same transport, as the voice transmission. This isn't meant to infer that DTMF and SS7 are comparablethey aren't. DTMF has a very narrow purpose: sending dialed digits.

Out-of-band signaling is also called CCS, or common channel signaling. It's the technique used by all telecommunication vendorsincluding cellular phone service providers, long-distance companies, and local exchange carriers (LECs). All of these networks share one thing in common: a common bond in SS7. The ITU-T recommendation for SS7 is, coincidentally, Q.7 (see Figure 4-4).

Figure 4-4. SS7 is accessed alongside the PSTN by CO switches and LD carrier switches in order to connect, disconnect, and bill calls

4.2.7. PRI/T1

The Primary Rate Interface is a digital access-signaling standard that allows 23 digitally encoded voice channels, called DS0s, to be carried across a T1 circuit between the CO and an enterprise PBX. These channels can be partitioned by an administrator using a DSU/CSU device, so that as few as a single one can be used for voice.

Channels not designated for voice tend to be used for data networking or not used at all. PRI service is sold by the telephone companies as a high-density dial-tone trunk solution. The 24th channel is called the carrier channel or the signaling channel , and it carries call-management signals over the T1. T1 is an integrated services digital network signaling standard and stems from ITU-T recommendation Q.931. For more about T1, see the later section titled "Time Division Multiplexing."

Digital signaling standards differ from analog ones because they don't rely on electrical signals; instead, they use signals passed through a bit stream.


4.2.8. SONET and DS3

Synchronous optical network STS-1 (SONET) and DS3 (sometimes called T3 because it is a high density of T1-style paths) are methods of providing high-bandwidth-signaling pathways between switches on the PSTN. These links can be used for actual transmission of digitized voice, as well as SS7 signals, but aren't normally used for both simultaneously .

SONET network links can be used to deliver telephone service to PSTN subscribers, too. A common use for optical high-bandwidth dial-tone services, which are delivered using OC (optical carrier) circuits, is in call centers, where thousands of phone calls must be connected to a central group of phone operators.

4.2.9. DID

Direct inward dial service is a solution that allows inbound PSTN calls to the PBX to be routed directly to certain private extensions based on the phone number that the caller dialed, without the need for a call transfer or a human greeting to determine which extension each call should be transferred to. The way this works is quite simple.

When the DID trunk rings, the PBX answers and produces a dial-tone and a "wink" signal. This is an indicator that the PBX is ready to receive digits from the CO. The CO sends the digits that the caller dialedusually just the last four digitsand the PBX decides how to connect the call based on them.

DID is most useful with PRI, where many DS0 channels can be assigned as DID channels. In this way, a single PRI trunk group can handle inbound calls to dozens or hundreds of phone numbers using 23 channels or less. An office with 80 desk phones could have 80 corresponding PSTN phone numbers and only a single PRI T1 for dial-tone trunking. Yet, the PBX knows which phone to ring when one of the 80 phone numbers is called because DID service tells the PBX which phone number the caller has dialed (see Figure 4-5).

Figure 4-5. A POTS trunk can be statically associated with a certain phone using a PBX

Whereas each POTS line has an assigned phone number that can't be changed, as in Figure 4-5, the DID channels on a PRI can, in effect, have a different phone number every time they are used to connect an incoming call to the PBX, as in Figure 4-6. More information on DID services can be found in the ITU-T's Q.951 recommendations.

Figure 4-6. DID is a signaling standard that allows a call on an incoming trunk to dynamically signal which phone number is being called so the PBX can ring the appropriate phone

4.2.10. Hunt Groups

There's nothing more frustrating than a busy signal when you're trying to order a pizza on Super Bowl Sunday. That's why most pizza establishments use hunt groups. Hunt groups are groups of trunks that allow incoming phone calls to circumvent the busy trunk and "roll" to the next available trunk in the group. That way, the pizza place never misses a call, even when their phones are ringing off the hook.

4.2.11. Centrex

Centrex is POTS enhanced with business-grade telephony features like call conferencing, four-digit dialing, and per-call billing rather than per-minute billing. It was designed to curb the need for small businesses to invest in PBX equipment in order to get modern telephony features. A single Centrex customer can use many Centrex lines, collectively called a Centrex group. Within the group, each line can be called using four-digit dialing instead of the usual 7-digit dialing (i.e., the caller can omit the prefix when placing calls within her Centrex group). Some other PBX-like features include the ability to easily transfer calls between lines in the same group, or enable and disable call forwarding for a given line by dialing a special sequence of DTMF digits. Normally, users of Centrex have to dial an 8 or a 9 at the beginning of each call that is destined for a receiver outside their Centrex group.



Switching to VoIP
Switching to VoIP
ISBN: 0596008686
EAN: 2147483647
Year: 2005
Pages: 172

Similar book on Amazon

flylib.com © 2008-2017.
If you may any questions please contact us: flylib@qtcs.net