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If you are using a T1 connection to the PSTN for telephone service you should determine the percentage of time your users are on telephone calls. Count the number of telephones in the office including conference rooms and fax machines. Try and find out the usage patterns for the phones. Is there ever time when everyone has to be on the phone? If not, fewer than the 23 channels may be enough for you office and you can rent a partial T1.

CODEC (Compessor Decompressor) change an analog voice signal into a digital data stream and back. Several different CODECs are supported. You can select the CODEC yo want to use. This process is described later.

For calling over the Internet or LAN, you must have network connectivity and sufficient bandwidth. Each telephone conversation will consume from 45 to 150 Kilo-bits per second of bandwidth depending on sound quality. At 50 Kbs call quality is comparable with a cell phone. At 75 Kbs call quality can rival a land line call.

The CODEC selection determines how many calls can be sent over your Internet connection. A tested various CODECs permitted the inclusion of his results here.

Buy Support Services

You may find that after you have purchased your hardware, purchasing installation and support system from Atek Canada is an advantage. This can dramatically reduce the risk of problems you will encounter and the time of setting up the system. An installation contract can include a support agreement.

Installation and Configuration Help

Introduction and preparation

You can add any interface boards. Add the drivers for the interface boards to your system. Note that the software included is always the most recent version.

You must configure your network. This may include making any FTP available. You will most likely need to configure DHCP (Dynamic Host Configuration Protocol.) For more information about DHCP, refer to RC 2131, 3396, and 3397

You must configure your server for your environment.

Configure any IP phones and IP adaptors. Install any analog telephone equipment.

Testing and Documentation

Please keep in mind:

Test your system thoroughly before letting your users try it. You must deliver a reliable, complete working system or you will alienate your users and your project may not success.

Test the full system including all the connections. Make sure any SIP, H.323 or PSTN connections operate correctly. Test all thePBX functions.

Test all the different ways to transfer calls. Do they all work with all the protocols and phones you are using? Does the transfer button on yourSIP phone transfer calls to other nonSIP phones or a different manufacturer's phones? Can non-SIP phones transfer calls to SIP phones? Create a grid of choices to assist your testing.

Test echo cancellation and change it as needed. If you don't test echo cancellation in advance, you are sure to get complaints from your users.

Document what you have done. Document your system hardware and software architecture.

Rollout

Test the system in the IT department before rolling it out to your company. Consider bringing a few users on line first. Don't try to bring the whole business up at once. Get some buy-in from early users A few happy test users will be very helpful in converting everyone else to happy users.

Train your stuff well. If your users aren't trained, they will fail and you will fail.

Provide at least some simple documentation for your users. User's rarely read documentation, but they may look at a short guide that gives them vital information quickly.

Keep clear records about hardware and software vendors, maintenance agreements and contact information.

We offer onsite 24h primary support in Montral area.

If you don't have primary support or if you are outside the great Montreal area, we recommand you to buy some critical parts or purchase duplicate system spares so that you can quickly respond if something goes wrong.

Telephony Hardware Selection

Inter- Exchange (IAX) connect to a remote SIP server. If the remote server has the required boards and an interface to the PSTN, the first server can access the PSTN through the remote server with IAX.

Even if you don't have any interface boards installed, you must install the drivers to use conferencing.

Telephone interface boards are available for POTS and local Analog devices.

Network Time Server

Your SIP server should send the Time with the name and number of the Caller ID.

It is important to configure your server to periodically set the system clock by accessing an Internet atomic time server.

Firewall

If you want to access the machine remotely, you will have to enable access to your machine for SSH remote access utilities at least during the initial steps of configuring and connecting server.

You may require a DHCP server, for example for configuring SIP phones dynamically. The Mepis distribution comes with an installed and operational DHCP server. This server has been configured to be the authoritative DNS server on its network. The DHCP configuration file is found in /etc/dhcp3.

After configuring DHCP, you can restart the DHCP daemon with the commands

FTP Server

Some phones, for example Cisco phones, require access to a FTP sever. They download their firmware and configuration settings from TFTP. TFTP is installed and enabled in your system.

If you would rather run TFTP from a Windows server, you will have to find and install a TFTP server. No TFTP server is included with Windows.

In other distributions, make sure the TFTP sever directory named in the configuration file exists. Make sure this directory has universal read and write permission. Make sure all files in theTFTPboot directory are readable

Be sure to test TFTP by requesting a file from a machine separate from you server. Many operating systems, including Windows, include a TFTP client. The Mepis TFTP installation writes log messages to /var/log/syslog. TFTP for Red Hat 8 leaves its message in the file /var/log/messages.

The cvs command will display many lines as the various sources are checked out of cvs and copied to your server.

Next, install any cards. Reboot the server.

Be sure to have all the hardware for your network, for example T1 cards, installed in your server before you compile. Any boards will need to be configured later.

The music on hold application and conferencing rely on access to a timing source. Three sources are available, the Zaptel drivers used with it's Wildcard boards, ztdummy, or zaprtc which uses the system clock.

If you install any other card, loading the driver for the card with the modprobe command automatically sets up the interface. Timing is then automatically available with no further configuration

The server provides timing information when no Wildcard board is installed.

Compile the Packages

Any telephony boards, for example a T1 card, should already be installed in your computer.

Compiling builds any drivers required for the installed telephony hardware. You do not need to restart your server after these compilation steps.

The last step, the make of the samples, creates a variety of sample configurations. Configuration is described in a later chapter.

If needed, edit the Makefile and try compiling again.

Before configuring, you must configure any hardware you are using. This includes SIP phones, soft phones, channel banks or communications boards.

Options

Options are set using the equals sign. Spaces are ignored. For example

An option can take multiple values. Multiple values are listed within square brackets and are separated by the pipe symbol "|".

Dial Plans

For any telephony system enterprise, a dial plan determines call routing and processing. For example, if a call comes in on a POTS line, where should that call be directed? If someone doesn't answer their phone, what should be done with the call? Should phones be answered after 5pm?

The configuration file contains the dial plan. The dial plan controls all call switching. Th dial plan controls the behavior of all connections through. The dial plan determines the route a call takes through the interfaces system and route calls based on either the called or caller number.

Personalized Configuration

Changing Default Configuration

Please refer to the included manual to change options.

Additional Onsite Services

Technical Support

Please contact-us for any Installation, Configuration and Support matter.

Available Services

T-Carrier and SONET

The most common business connection to the PSTN (Public Switched Telephone Network,) or Internet is a T1 line, or in areas outside the US an E1 line. A T1 line is often called a DS-1. The following sections describe T1 and other "T" type lines.

A T1 line, provides a point-to-point connection. For example, you can use a T1 line to connect your office to the telephone company central office switch for dial telephone service. You can use a T1 lin to connect your local computer network to an ISP to establish a connection to the Internet. You, the user, determine the end points. You, the user, determine what the T1 line is used for, voice or data o both.

T-Carrier is a series of digital communications systems used by telephone companies around the world. T-Carrier is a digital protocol developed by AT&T by 1957 and first implemented in the early 1960's. The T-Carrier was developed to support the transmission of digitized voice. T-Carrier provides telephone companies the technology to move voice or data digitally over what had been before an analog system

T-Carrier uses two pairs of wire. It is full-duplex, that is data can be sent and received at the same time. Signals are digitized and then sent over the T1 connection. Voice is sampled 8,000 times a second and converted into eight bit words. An frame is built that contains a word for each of the 24 channels. A frame is transmitted 8,000 times a second.

Digital T-Carrier circuits provide much greater bandwidth than analog circuits. A set of copper wires used to transmit an analog signal can instead transmit data digitally. Sending data digitally allow much more data, even much more digitized voice, to be sent over the same copper wires.

T-Carrier is used to build the ISDN, Integrated Services Data Network. ISDN is a set of integrated standards used to build a digital telephone network. With ISDN the same switches and digital transmission paths are used to establish connections for different services, for example data and voice.

The ISDN standard was first published as one of the 1984 ITU-T Red Book recommendations and expanded in the 1988 Blue Book. ISDN uses Public Switched Telephone Network ( PSTN) switches and wiring. This wiring is upgraded to support the basic"telephone call" on a digital network.

Different types of T-Carrier circuits are available. When you order T-Carrier line, for example a T1 line, you order a circuit with a specified amount of bandwidth. For example, a 24 channel T1 line wil provide 1.544 mbps of bandwidth or a T-3 line will provide 44.736 mbps of bandwidth.

T-Carrier costs are continually dropping. T-Carrier lines are extremely popular for business users who wish to connect to the Internet or the PSTN.

T-Carrier and DS0

The "T" designation specifies the physical interface for services obtained from a local carrier. That is, T-Carrier specifies a physical set of wires, repeaters, connectors, plugs, jacks, etc.

In terms of the OSI standard network model (briefly described in the appendix,) T-Carrier is the standard for layers one and two. T-Carrier specifies the physical connection and the carrier signal sent over that physical connection.

Data is carried on top of the T-Carrier. Data is carried on a T-Carrier channel at a digital data rate that is called Digital Signal Level Zero or DS0. DS0 is described below.

T-Carrier describes the physical layer interface to a provider network. A T-Carrier circuit is typically provided as two pairs of wire. These are bare wires that run directly from the central office to the customer premises without any conditioning1

The maximum T-Carrier signal distance is 3000 wire feet measured from the egress at the cnetral office. Repeaters are used to extend a T-Carrier signal further than 3000 wire feet. The first repeater is placed within 3000 wire feet of the CO. Successive repeaters are placed every 5000 wire feet. The las repeater is installed within 3000 wire feet of the customer's termination point

1. Conditioning devices like bridge taps and load coils are used on analog telephone lines to help maintain or improve signal quality. Splices, which are common, tend to degrade signal quality

Figure: 15-1 T1 Repeaters

Once the physical T-Carrier line is installed, you can use it to send and receive data. Customer data including voice (for telephone calls,) data or video can be sent over the T-Carrier line.

Note that this type of circuit is rapidly becoming obsolete. Many new DS-1 circuits are being delivered on one pair of copper wires using HDSL technology.

T-Carrier is a channelized system. In North America, the basic data channel is called a Digital Signal Zero (DS0) channel.

Digital Signal Zero was standardized by the ANSI T1.107 guidelines. The international ITU-T guidelines are slightly different.

DS0 is a dedicated, point-to-point line service. DS0 service can send voice and digital data including video.Each DS0 channel provides 64 kbs of bandwidth, enough bandwidth to transmit a digitize voice signal. Each DS0 provides a 64 kilobits per second PCM end-to-end channel transmitted over the T-Carrier. Voice signals are sampled 8,000 times a second. Each of the samples is digitized into an 8-bit word which supports a 64 Kbs signal. Each of the 8-bit words is sent over the DS0 channel.

The multiple T-Carrier channels in a single T-Carrier connection can transmit voice or to transmit data. The separate channels in a T-Carrier circuit can be assigned to different uses. Some channels can be dedicated to telephone usage while others are simultaneously dedicated to data

As described in the following section, DS0 channels can be combined to create high bandwidth connections.

T-Carrier systems combine channels to provide greater bandwidth. For example, in North America a T1 line provides 24 channels for a total bandwidth of 1.544 mbps and in Europe an E-1 line provides  

2.048 mbps of bandwidth and 30 channels. T-Carrier bandwidth is aggregated by combining DS0 channels.

There is a hierarchy of T-Carrier circuits. Each step provides more bandwidth. The hierarchy of combinations for T-Carrier circuits are shown in Table 1. It is possible to purchase a "fractional" T1 line where fewer than 24 channels are provided

Note 1: The DS designations other than DS0 are used in connection with the North American hierarchy only.

Note 2: Other data rates are in use. The military has systems that operate at six and eight times the DS1 rate. At least one commercial system operates at 90 Mb/s, twice the DS3 rate

Note 3: T1c, T-2 and T-4 are rarely used.

T1 lines are in common use today in for connections to the Internet. The T-3 line, providing 44.736 mbps, is commonly used between Internet service providers.

Integrated Services Data Network ( ISDN) was standardized in the 1980s. ISDN is an international standard interface protocol from The International Telecommunications Union (ITU-T formerly th CCITT,) providing single access to multiple services. ISDN signaling is SS7 compatible. ISDN subscribers can access SS7 network services and intelligence through ISDN.

ISDN provides a variety of communications services in circuit switched networks. These include bearer services for speech, 3.1 kHz audio for modems and 64 kbps digital data. Teleservice support fa and telex. Supplementary services include calling line identification (caller ID,) user-to-user signalling call waiting, and call hold among others.

ISDN provides D and B channels. Bearer (B) channels are bi-directional 64 kbs channels that carry user information. B channels do not carry signalling information. Bi-directional 9.6 kbs Data (D channels carry signalling information.

When someone talks about an ISDN connection, they are usually referring to a Basic Rate Interface. A (BRI) provides two 64K"bearer" DS0 channels and a single "delta" DS0 channel ("2B+D"). The bearer channels are used for data transmission. The delta channel is used for out of band signalling, fo example call setup. Because of tariffs, BRI ISDN is typically an expensive service to operate. BRI ISDN lines are typically charged by the minute which causes the cost to quickly rise. While ISDN has had some success in video conferencing, because of the cost it has never become very popular in Nort America. More modern DSL technology has replaced ISDN for anyone who has access to DSL. BRI is still popular and cost effective in many European locals.

A Primary Rate Interface ( PRI) in North America and Japan consists of 24 channels, usually 23 B + 1 D channel with the same physical interface as a T1 where all the channels operate at 64 kbps.Th combined PRI channels results in a digital signal 1 (T1) interface at the network boundary.

In some areas outside the US, the PRI usually has 30 B + 1 D channel and an E1 interface. As with the BRI, the D channel is used for out of band signalling. While a PRI is an ISDN connection, it is rarely referred to as such.

T-Carrier sends data over the line in bytes. Each byte is sent in order, one after the other in "frames." A single frame contains one Byte (8 bits) of data for each channel. An extra bit is then sent to synchronize the data stream. This extra bit is called a Frame Bit.

193 bits, 192 data bits and one framing bit, are sent for each frame. This increases the total bandwidth to 1.544 mbps. 24 64 Kb DS0 channels taken together provide 1.536 mbps. A T1 provides 1.544 mbps of bandwidth. The extra bandwidth comes from the Frame bits. T1C frames differ as they are made u of 1272 bits.

T-Carrier uses pulse code modulation and time-division multiplexing. The time division multiplexing is illustrated in the following figure. A frame is sent in 125 micro seconds. T-Carrier uses four wires and provides duplex capability. Two wires are used for receiving and two for simultaneous sendin

Figure: 15-2 Data Frame

In North America, Canada, Hong Kong, and Taiwan two framing formats are in use, Superframe and Extended Superframe.

A Superframe consists of twelve 193-bit frames. A framing bit can support different functions, depending upon which of the twelve frames it is in. There are two types of framing bits; Termina framing (Ft) and Signaling framing (Fs) bits.

With Superframe, the standard frame is 193 bits long and includes 1 Framing bit plus 24 8-bit time slots. Each Superframe time slots is scanned at a rate of 8000 times per second. Therefore, in one second, there are: (8000 * 8 bits)/TS * 24 TS = 1,536,000 Bits of payload data transmitted. There are 800 1 = 8,000 bits of synchronization bits transmitted within a one second interval. Therefore, the tota aggregate rate of the T1 signal is 1,544,000 bps (1.544 Mbps)

The standard frame is 193 bits long, 1 framing bit + 24 8-bit time slots. Each time slot is scanned at the rate of 8000 times per second, as in D4/SF. The line rate is 1.544 Mbps and supports a data paylo of 1.536 Mbp

Signalling states are transported within a Superframe. This is required to support Switched voice or data service. Signals are sent with a"Robbed Bit" bit:8 of each channel's time slot is "robbed" to indicate a signaling state in the 6th and 12th frames. Effective throughput for the A signaling bit (Frame  

6) is 666.66 BPS. Effective throughput for the B signaling bit (Frame 12) is the same (666.66 BPS).

An Extended Superframe consists of twenty-four 193-bit frames.There are three types of framing bits; Frame Pattern Sync (FPS), Datalink (DL), and Cyclic Redundancy Check (CRC) bits. Of the 8 kbs framing bit bandwidth 4 kbs is allocated to the Datalink, 2 kbs is allocated to the CRC-6 characte and 2 kbs is used for synchronization purpose

ESF (Extended Superframe Signaling) uses a "Robbed Bit" Each channel's timeslot is "robbed" to create a signaling in the 6th, 12th, 18th, and 24th frames. Effective throughput for the A signaling bit (Frame 6) is 333.33 BPS. Effective throughput for the B, C and D bits is the same (333.33 BPS)

After your T1 provider drops the T1 into your premises, they may then hand you a CSU/DSU or a router. This router will have a T1 connection on the back. The router contains circuitry that communicate with T-Carrier. A connection between the telephone company T1 drop and the router establishes the connection.

You can connect from the T1 drop to the router. It is advisable to use a real RJ45 cable instead of a CAT5 cable. This is described in the section on cables and connectors below. If you are using the T1 line only for data, your configuration may be complete when you configure your router and connect i to your LAN. This will provide a path for data from your company to the other end of the T1 line

A channel that is used to place telephone calls to the PSTN must be connected to the PSTN, for example a CO (central office.)

If you are using the T1 for telephone calls to the PSTN, you will need some piece of equipment that provides a connection between your analog telephones or fax machines and the T1 line. If you are using the T1 for making telephone calls, your router may have a connector on the back that accepts T1 cable. This means the router is smart enough to take telephone traffic off the T1 channels an route them to the telephone connectors on the back of the router. You may have a separate piece o equipment called a channel bank that accepts the T1 line

IP Phones will, of course, connect to your local area network, not the analog connectors. A VoIP call can be sent over a T1 DS0 channel as data. This data channel could be connected to your ISP. Th telephone call would then be routed over the Internet instead of the PSTN. Such a call might eventually be connected back to the PSTN through a gateway elsewhere.

When you hear someone say T1 you will probably have a hard time figuring out exactly what they mean. T-Carrier discussions are very confusing because of the interchangeability of words and the confusing requirements for connecting to the PSTN.

A T1 line can refer to a connection that has 1.544 mbps of bandwidth. It might be referring to a network that uses the T carrier electrical interface specification (DSX-1.) Or, it might mean that the network uses one of several framing formats, D4, ESF, etc.

A T1 cable is different from a CAT5 ethernet cable. Use a real T1 cable when a T1 cable is called for. When extending T1 lines from the phone company drop to your customer equipment, use a T1 cable not aCAT5 cable.

T1 cables use Individually Shielded Twisted Pair ( ISTP.) ISTP is used because of the susceptibility of T1 signals to Near End Cross Talk NEXT.)

Unshielded Twisted Pair ( UTP) cable characteristics are similar to ISTP. However, due to the unshielded characteristics of UTP, the proximity of the unshielded transmit and receive cable pairs can cause NEXT. This can result in link errors if you use a CAT5 cable.

Various vendors may have optional T1 services that you may want. Here is an example.

The AT&T Digital Carrier System is referred to as ACCUNET T1.5. It is described as a "two-point, dedicated, high capacity, digital service provided on terrestrial digital facilities capable of transmitting 1.544 Mbs" The interface to the customer can be either a T1 carrier or a higher order multiplexed facility such as those used to provide access from fiber optic and radio systems

AT&T offers services in addition to point-to-point data service. For example, four "transfer arrangements" can be purchased:

1. The customer can change the terminating location of a T1 link with AT&T assistance.
2. M24 Multiplexing allows the user to subscribe to any of the 24 T1 channels individually to switched and non-switched services offered by A&T.
3. M44 Multiplexing combines 2 T1 lines, each carrying up to 22 channels, onto one T1 line using Bit Compression Multiplexing (BCM)
4. Customer Controlled Reconfiguration (CCR) allows the customer to dynamically allocate circuits without A&T assistance.

AT&T states that their performance objective is 95% Error Free Seconds (EFS) on a daily basis and  

99.7% availability on a yearly basis.

Be sure to check what features your service provider that might be helpful in your application.

In 1917 AT&T deployed the first carrier system, called the "A" system. Seven A-systems, with four voice channels over pair of wires, were ever deployed. Over time, newer analog frequency division multiplex systems named B, C and D, were developed. Few of these saw commercial service. The L syste was very successful and provided 600 (L1) and later 1800 (L3) voice channels over a pair of coaxia cables.

The telephone companies refer to long distance service as "long haul" or "long lines." This system stayed in long haul service from 1944 to 1984 when the breakup of the Bell System forced A&T to move to optical fiber. The last analog carrier system was the N system. This system provided 12 voic channels and was used for intracity short haul. O, P, and U systems were never put into service, th emergence of T killed them

In 1957 digital systems were first proposed and developed. A manager at AT&T, then the only telephone company, decided to skip Q, R, S, and to use T, for Time Division. This was to be the world's first time division system. Except for"U", another system that was never deployed, this naming system ended

The variants of T1 called T1C, T2, and T4, all vanished. They are survived by signals that would have been carried on all these systems. These are called DS1, DS2, DS3, and DS4. The successor to the T-Carrier protocols are various protocols running on optical fiber, for example SONET, but they don't have a letter designation.

The next step up from T-Carrier is SONET, Synchronized Optical Network. SONET is a very high speed physical layer network protocol. It is designed to transmit large volumes of traffic over long distances on fiber optic cables. ANSI developedSONET for the public telephone network in the mid-1980s. You would be able to make a very large number of telephone calls over a SONET connection.

SONET specifies interoperability standards between products from different vendors. SONET can carry different data protocols including IP.SONET includes management and maintenance support. SONET is cost competitive with alternatives like ATM and Gigabit Ethernet

SONET specifies OC (optical carrier) signal levels. The OC signal levels place STS (synchronous transport signal) frames onto a multimode fiber-optic line at a variety of speeds. The base signal rate is  

51.84 Mbps (OC-1); each signal level thereafter operates at a speed divisible by that number (thus, OC-3 runs at 155.52 Mbps)

This system is built with multiplies of the OC-1 rate of 51.840 Mbps. This is called STS-1 (Synchronous Transport Signal, Level 1). T

This system uses a fundamental rate of 155.520 mbps, three times the speed of SONET. The fundamental signal is STM-1 (Synchronous Transport Module, Level 1). The transmission media fiber, butis the Broadband ISDN specifies a User-Network Interface STM-1 (155.520 mbps) operating over coaxial cables. Some typical rates within this hierarchy

The Public Switched Telephone Network started in 1876 when Alexander Grahm Bell made the first telephone call. The first call was from Mr. Bell to his assistanct Mr. Watson where he said,"Mr. Watson, come here. I need you." The second call was from a telemarketer.

This first call was made over a ring-down circuit. Two wires connected the two telephones. The first phone was always connected to the second phone and there was no ringing. This was a half-duplex circuit where only one person could talk at once. As shown in the following diagram, every phone wa connected to every other phone

Figure: 16-1 Fully Connected or Full-Mesh

It would be too expensive, and too difficult, to build a large telephone network with this topology. The solution to this topology problem is a switch. A switch only requires a wire pair from each phone to central office. At the central office, a switch is used to connect one call to another call. The origina switch was a person, the operator.

The PSTN quickly evolved to a full duplex system where both parties could talk at the same time. The person was soon replaced by a mechanical switch. Years later, the mechanical switch was replaced wit the electronic switch.

Figure: 16-2 Fully Connected

Sounds are analog. They are continuous wave forms that vary in frequency and amplitude. The PSTN originally sent analog signals from one phone to another. Over longer distances, the signals nee amplification. Unfortunately, amplification makes the noise louder as it makes the signal louder. Eac additional amplifier adds more noise and degrades the signal further as it traveled over longer distances.

More recent technology allows analog signals to be digitized. The original analog waveform can be represented as a stream of numbers. Digitization relies on the Nyquist theorem. A high quality digita representation of an analog wave form can be created by sampling the waveform twice as fast as th highest frequency found in the analog waveform.

The most common method of digitizing analog signals is Pulse Code Modulation. With PCM, the analog signal is first filtered, for example to remove any frequencies above 4kHz or below 100Hz. Thi signal is then sampled 8,000 times per second, twice the highest frequency.

The samples create a digital data stream. Each data element in the data stream represents the amplitude of the original analog waveform at the moment the sample was taken. PCM uses an eight bit coding scheme coupled with a logarithmic compression algorithm. Sampling eight bit values at 8,000 times a second produces a 64 kbs data stream.

A pair of wires running from a central office to a telephone is called a local loop. The local loop connects the telephone to a switch in the central office. The communications link between one central office and another is called a trunk. Central offices are connected hierarchically. Central office switche connect through trunks to tandem switches. Tandem switches are referred to as Class 4 switches.

Class 5 switches often connect directly to each other. These connections are put in place after analyzing calling patterns between switches. If there are enough calls between two class 5 switches a dedicated circuit is installed.

Figure: 16-3 PSTN

A local loop, that is a pair of copper wires, can transmit analog or digital data to a central office. There are two signalling paths in the PSTN. End users signal the PSTN with user-to-network signalling. Switches in the PSTN signal each other with network-to-network signalling.

Signals can be analog or digital. Dual Tone Multi-Frequency (DTMF) signalling sends two simultaneous tones over the voice path.

Signaling can be in-band or out-of-band. For example, DTMF is in-band signalling. Dialing a number sends analog DTMF signals to the central office switch over the voice circuit.

Out of band signalling sends signalling information on a separate channel from the transmitted voice. For example, a Basic Rate Interface provides two 64kbps bearer (B) channels used to send and receiv voice and a third 16 kbs D (data) channel used for out of band signalling.

Out of band signalling has several benefits including reduced dialing delay, higher signal bandwidth and the ability to multiplex multiple signals over single channel. Out-of-band signalling greatl improves call service including call completion.

Network-to-network signalling includes in-band signalling methods like Multi-Frequency (MF) and Robbed Bit Signalling (RBS.) MF is like DTMF but uses different frequencies.

SS7 (C7 in Europe) is the common out-of-band signalling protocol used between switches. SS7 is used to send messages between switches for basic call control. SS7 allows signalling to control th Intelligent Network. The Intelligent Network implements Custom Local Area Calling Services lik three way calling or call waitin

CLASS services include

Call Forwarding

Call Waiting

Three-way Calling

Speed Calling

Anonymous Call Rejection

Automatic Callback

Automatic Recall

Call Forwarding Busy

Call Forwarding No Answer

Call Name and Number Delivery

Call Name and Number Delivery w/Call Waiting

Call Number Delivery

Call Number Delivery w/Call Waiting

Call Number Delivery Blocking

Customer Originated Trace

Distinctive Ringing / Call Waiting

Selective Call Acceptance

Selective Call Forwarding

Selective Call Rejection

Voice Mail

SS7 to database connections support network-based services including 800-number service and Local Number Portability.

The following sequence diagram shows a typical SS7 call flow. In this example, picking up the phone sends an off-hook signal to the SS7 switch at the local office. The switch sends dial tone to the phone The caller presses buttons on the phone. This sends a message to the switch containing a telephon number. The Switch responds to the dialed number with a setup or Initial Address Message (IAM.) The local switch sends a new IAM across the SS7 network to the second switch. The second switc sends an Address Complete Message (ACM) back over the SS7 network. The called phone rings. Th calling party hears a ringing sound. The called user picks up the phone. This action sends an off hoo message back to the switch. The switch send an alerting message back over the SS7 network. Hangin up a phone disconnects the call.

Figure: 16-4 SS7 Call Flow

A local call can usually be dialed with seven digits. Dialing a long distance call requires dialing 1, and then an area code, and then the three digit exchange number, and then the last four digits of the telephone number. This scheme is the dialing plan for the PSTN.

The number of telephone numbers that are needed has dramatically grown over the years. Because of this, the current dialing plan may have to be changed to demand eleven digit dialing for all numbers.

Dial around is now available for a user to specify a long distance carrier. Dialing some number like 10+XX+XXX can switch a call to the desired long distance carrier.

The ITU-T Recommendation E.164 International Numbering Plan uses a Country Code (CC), national Destination Code (NDC) and Subscriber Number (SN) to switch a call to a user. The CC can be one, two or three digits. The NDC and SN can vary in length from country to country. Neither can have more than 15 digits.

The PSTN has held up well over the years for switching telephone calls from one user to another. On many networks built for voice, there is more data being sent than voice. This data is being sent over network that was optimized for voice. The PSTN was never designed for data traffic and suffers for it.

In the near future, most voice will be carried as data over networks that were designed to carry data. In the future, more and more voice traffic will be sent over IP or ATM telephone company networks.

This chapter briefly addresses VoIP standards, especially H.323 and SIP. SIP is obsoleting H.323 so the emphasis is onSIP. For a more comprehensive discussion of SIP, consult the SIP standard or the book Internet Communications Using SIP by Henry Sinnreich.

You do not need an in-depth understanding of VoIP standards to build systems. hides most of the complexity of VoIP protocols for you. A more detailed understanding of these protocols could be necessary if you decide to become an developer.

Open VoIP separates calling into bearer (IP, RTP) streams, services and call control. Standards define each of these three protocol stacks.

This book assumes you are already familiar with networks and TCP/IP. There is no attempt here to describe basic networking. There are many excellent references for this.

Data networks, both IP and ATM, are packet based. Packet networks are obsoleting circuit switched networks.

IP is particularly attractive for data transport. IP is a transparent transport layer. It is a widely adopted standard and provides the most common application interface. IP transparently transports data endto-end regardless of the application.

Packet loss is common in IP networks. IP networks are self-healing. Dynamic routing protocols allows a network to re-converge to overcome packet loss or to find the best possible route. Dynamic routin means the packets in a data stream can travel separate paths. This means that packet transit and arriva times can vary from packet to packet.

Packet loss is a normal occurrence in an IP network. TCP/IP uses packet loss to control packet flow. If a packet is lost, TCP re-sends the packet. TCP uses packet loss to tune packet transmission.

ITU-T recommends a one way packet delay of no more than 150 ms. This is why TCP suffers over a satellite link. TCP does not deal well with the extremely long propagation delays of a satellite link.

IP does not directly support real-time traffic sessions. Real-Time Transport Protocol ( RTP) is the emergent protocol for real-time traffic sessions over IP networks. The packets for a particular RTP session are referred to as an RTP stream or a media stream. RTP is commonly used to transport voice traffic. Many applications, for example Microsoft Net Meeting, use RTP.

In a real-time environment like voice, re-sending a lost packet is too time consuming. Small numbers of lost packets in a voice stream are not noticeable to a listener. It's better to ignore the lost packet than re-transmit them. Unlike TCP, UDP is an unreliable protocol. That is, there is no guaranteed delivery of a packet with UDP. This is one of the reasons why RTP runs over UDP instead of TCP.

Packets that are part of a real-time session can arrive out of order. RTP packets each contain a timestamp. The timestamp allows the receiving application to reassemble incoming packets in the correct order. RTP uses the packet timestamps to tune its settings. RTP can use the timing information to adjust for network problems like delay and jitter as well as packet loss

Call control is the process of managing and routing a call. For the PSTN, management and routing are both managed by SS7. VoIP IP bearer streams are separate from call control.

An enterprise class switch is circuit switched. Like the PSTN, channels are usually 64 kbps. The PSTN and enterprise switches can both offer services like call waiting, call hold and call transfer. While a Class 5 switch can handle hundreds of thousands of simultaneous calls, enterprise switches ar typically much smaller.

Class 5 is an telephone industry call control standard. Central office switches use Class 5. Most enterprise switches use proprietary manufacturer protocols. Most proprietary enterprise switches provide advanced features that are not available on Class 5 switches. Class 5 switches were developed to support residential telephony, not complex business services. Enterprise switches typically provide much, much richer feature set. The high-use feature-rich services available on proprietary enterprise switche are available on Aterisk.

There are a variety of IP routing protocols including Router Information Protocol (RIP,) Interior Gateway Routing Protocol IGRP,) Enhanced Interior Gateway Routing Protocol ( EIGRP,) Intermediary System to intermediary System (IS-IS,) Open Shortest Path First ( OSPF,) and Border gateway protocol BGP.) Each of these protocols provides a different solution to the problem of routing updates that solves a different problem. Each of these accomplishes the same thing, routing a packet from th source to the destination.

Similarly, there are several Internet open call control protocols. They all resolve traffic to IP addresses. They currently include H.323, SGCP, MGCP, andSIP. There are proprietary protocols like the Cisco Skinny protocol. More protocols will appear in the future to address new needs.

There is no need to standardize on a single call control protocol. These protocols enable standards for applications at the call-control layer. With the open protocols, applications from different vendors ar interoperatble. operates with many of these protocols including Skinny.

H.323 is currently the most widely deployed VoIP call-control protocol. H.323 is not robust enough to use in a system that can compete with the SS7 PSTN. SIP is the most likely packet based competitor to SS7.

H.323 is an International Telecommunications Union Telecommunications Standardization Sector (ITU-T) specification for transmitting multimedia traffic including video and voice over an IP network. H.323 works with other existing standards like Q.931. Compliant vendor products an applications can communicate with each other via this protocol.

H323 is complex. It's not easy to create H.323 applications. H.323 applications do not scale well.

H.323 comprises the following components and protocols

H.323 elements include terminals, gateways, gatekeepers and multipoint control units (MCU.)

Terminals, often called endpoints, provide point-to-point and multipoint conferencing for audio, video and data. Gateways can interconnect to the PSTN or ISDN networks.

Gateways are used to connect between a Switched Circuit Network (SCN) endpoints and H.323 endpoints. Gateways are only needed when an H.323 endpoint needs to interconnect to a different network.

Gateways provide address translation services and admission control. Gateways translate between audio, video and data transmission formats. Gateways interconnect communication systems and protocols

A gatekeeper provides pre-call and call-level control services to H.323 endpoints. H.323 gatekeepers are separated logically from the other network elements. Inter-gateway communications isn't currentl specified by H.323. A gatekeeper can provide call control signalling, call authorization, bandwidt management and call management functions.

A multipoint controller (MC) supports conferencing between three or more endpoints. A multipoint processor (MP) receives audio, video and data streams and then redistributes those streams to the endpoints in a multipoint conference.

An MCU is an endpoint that supports multipoint conferences. An MCU must include at least an MC and one or more MPs. A typical MCU for centralized multipoint conferences includes an MC, a audio MP, a video MP and a data MP.

An H.323 proxy server operates at the application layer. It examines packets sent between to communicating applications. The proxy supports reservations, H.323 traffic routing and Network Address Translation NAT.)

The following figure shows a sequence diagram for the call flows between two IP addresses. This example assumes that the two endpoints have already resolved each other's address.

Figure: 16-5 H.323

In the example, endpoint one sends a setup message to endpoint B. This message is sent to TCP port 1720. Endpoint B replies with an alerting message that includes a port number. This message initiates  

H.245 negotiations.

The H.245 negotiations setup the codec types and port numbers for the RTP streams. The Codec types are specified by G.729 and G.723.1. Any other capabilities the endpoints share are negotiated.

Logical channels for the UDP streams are negotiated, opened and acknowledged. The two endpoints can now send and receive the media stream containing the voice traffic.

Real Time Control Protocol can transmit information about the RTP stream to the two endpoints during the session

This call-flow shows an example of H.323 version one. H.323 version two allow H.245 to be negotiated through a tunnel in the H.225 setup message. This is called fast-start. A fast-start reduces the number of messages needed to initiate a call.

SIP is described in RFC.2543. SIP is an application-layer control protocol used to create, modify and terminate a communications session. ASIP invitation can establish sessions and describe sessions. SIP features of user location, user capability, user availability, call setup and call handling can initiate or en communications sessions.

Henning Schulzrinne, one of the original architects of SIP, said that the objective of SIP is the "re engineering of the telephone system from the ground up" He said this is an "opportunity that appears only once after 100 years"

A SIP session can have one or more participants. Sessions can include audio, video and data streams. SIP is flexible enough to support ad-hoc conferencing. Multi-media SIP sessions can be multicast, unicast, point-to-point, or combine broadcast methods.

While SIP is not yet as widespread as H.323, it is catching up fast. Most modern application implementations are relying on SIP rather than H.323. SIP is extensible and will easily support additional functionality as it is needed.SIP will outmode any proprietary protocols.

A sip user agent is a client end application continuing a user-agent client (UAC) and user-agent server (UAS.) These are know as aSIP client and SIP server. The client initiates SIP requests as a user's agent. A server gets requests. ASIP server acts as a user's agent.

There are two types of SIP network servers: proxy servers and redirect servers. Proxy servers contain client and server functions. A proxy server acts on the behalf of other clients. It can rewrite headers t identify the proxy as the request initiator. The proxy server makes sure that traffic is sent back to th correct client.

A redirect server accepts SIP requests and responds to the client with the address of the next server. A redirect server doesn't manage calls. A redirect server doesn't process or forwardSIP requests.

A SIP client must be able to locate a SIP server. A SIP client must determine the IP address and port number of a target server. The defaultSIP port is 5060. The SIP client can query a Domain Name Server DNS) for a sever IP address.

After SIP address resolution, the SIP client sends one or more SIP requests and gets back one or more SIP responses. All the requests and responses are part of a SIP transaction.

Signalling sets up, mantains and terminates calls. SIP provides a rich set of signaling facilities for VoIP. SIP can

• * Register IP phones.
• * Register other SIP devices.
• * Register end-user preferences.
• * Authentication, authorization and accounting.
• * Address resolution, name mapping, and call redirection.
• * Find the media capabilities of a target endpoint using Session Description Protocol.
• * Determine the availability of a target endpoint.
• * Establish a session between an originating and target endpoint.
• * Allow mid-call changes like the addition of another endpoint to a conference.
• * Report call progress including call success and failure.
• * Transfer and terminate.

SIP supports a variety of intelligent network services. These include:

• * Call Hold
• * Consultation Hold
• * Unattended Transfer
• * Unconditional Call Forward
• * Call Forward on Busy
• * Call Forward on No Answer
• * Three-Way Conferencing
• * Single Line Extension
• * Find-Me
• * Incoming Call Screening
• * Outgoing Call Screening
• * Secondary Number In
• * Secondary Number Out
• * Do Not Disturb
• * Call Waiting

SIP was designed to support multimedia conferencing. SIP also supports multimedia conferencing, multipoint conferencing and call control for conferencing.SIP enables instant messaging and instant communications.

SIP is a powerful, general protocol for establishing interactive communications sessions. SIP provides facilities for initiating, modifying and terminating interactive communications sessions.SIP is not a resource reservation or prioritization protocol. There is no Quality of Service (QOS) support inSIP. SIP is not a data transport protocol. SIP is not designed for managing interactive sessions after the sessions have been established. SIP is not designed to replace all the features and services provided by the PSTN. Many of the Class 5 features are not needed in the context of the Internet. Some features are provided by other protocols besidesSIP.

SIP elements are User Agents, Servers and Location servers. User Agents are the endpoints of a SIP network. User Agents originateSIP requests to start and stop sessions and to send and receive data. A User Agent can be a hardware phone, a software phone running on a PC, or a gateway to another network like the PSTN.

Every SIP User Agent includes a User Agent Client and a User Agent Server. A User Agent Client (UAC) is the component of the User Agent that initiates requests. The User Agent server (UAS) is th component of the User Agent that responds to requests. Both are typically used during aSIP session.

Servers are intermediaries. They help User Agents establish and manage a SIP session. There are three types ofSIP server. SIP proxies forward SIP requests. Redirect servers get a request from a user agent, they return an indication of where the request should be resent to. Registrar servers update location o other database information.

Location servers maintain databases of information like URLs, IP addresses, scripts, features and preferences. User agents usually interact with Location Servers through a SIP proxy.

SIP Uniform Resource Locators (URLs) provide addressing similar to e-mail addressing. A SIP URL can have various forms and can include a telephone number, for example,

Figure: 16-6 SIP Address Resolution

SIP support of telephone number addressing and Web addressing supports bridging between the two networks. If aSIP endpoint knows the URL of another SIP endpoint, direct communications is possible.

SIP address resolution starts with a URI that resolves to a username at an IP address. The figure above shows a sequence diagram for a typical address resolution sequence where a URI is resolved to a user a an IP address.

Session Setup is the primary function of SIP. SIP sends an invite request. The invite request can contain a message describing the desired session type. The following sequence diagram shows a typical session setup.

Figure: 16-7 SIP Session Setup

This has been a fast introduction to a very complex topic. For more information please consult one of the excellent references.

In older usage, a private telephone switchboard that provided on-premises dial services.

Voicemail A system that receives, stores, plays and manages voice messages.