Dear teacher
My insert mite in school of information and communication technology
How insert own mite in school of information and communication technology?
In present, information and communication technology are developing rapidly from day to day. Therefore, I must be to late from it. In my opinion, country’s development base on information and communication technology’s development. To not late this development, skillful specialists are important in our country.
There are many universities which prepare specialists of information and communication technology but I chose School of information and communication technology, Mongolian university of science and technology and now I am studying here. Because my school has been preparing skillful specialists as this many years.
Developing technology rapidly is impression of how preparing skillful specialists well as this route.
To insert own mite in our school, we must expand our knowledge, study lesson well and be creative if we can these, it is our mite in our school. If we can prepare specialists who accepted in world level, we will make our bright future by our hand.
We want study in foreign country’s university especially in America. Why?
Perhaps lesson plan of our country’s university is wrong? I don’t think so. It only depend on us. Actually, our lesson plan of university is similar to lesson plan of foreign university and skill of teachers is similar to them. We trend too careless, so we late lesson’s standard. Then we can’t understand narrow things of science.
That is why we have to strive yourselves. We acquire education in our country and be skillful specialist, we can do bigger change in development of our country and our school.
This is our beside good engineer of our country also we will be face of fase of or school
Engineers of Telecommunication will describe development of country.
sincerely,
Baasanjav.D
Thursday, December 9, 2010
Wednesday, December 8, 2010
INTERVIEW
1. What are your long-range goals and objectives?
- My long range goal is to be high level of professional specialist by my profession. Also I will prefer responsibility and good position which can give effective result to workers
2. What are your short-range goals and objectives?
- My short range goal is to improve my education. It will be preparation my future success. I will collect experience working in company. I hope there will be chance to me actualize my knowledge acquired from university to my job.
3. In what ways do you think you can make a contribution to our organization?
- I had been working on the optical network projects and these experiences would help your organization.
I think I can make my contribution improving my high level of education and my profession. Therefore, I will study by more degree of my profession besides of my job. I prefer my private and team contribution to organization. Contribution will be new idea to the electrical field.
4. What have you learned from participation in extra-curricular activities?
- I like participation in activities which outside from class. This has been become more and more abilities for me. And I skilled in teamwork and able to rise easily.
5. How do you determine or evaluate success?
- I like fair everything. So I want to determine for justices. Head person must say openly to all success. I usually define my success for estimation of other people.
6. How do you plan to achieve your career goals?
- I will continue my professional development by participating in conference, attending seminars and continuing my education.
7. What makes a job enjoyable for you?
-My work is a very interesting and when I was student I don’t know it. After I study this school I understand it
8. Why did you decide to seek a position with this communication organization?
-Even if I am 22, and have smaller experience than others, I consider I can work in this company successfully caused I possible to input or treat to my job via new method, initiative which minded from old idea.
9. What two or three things would be most important to you in your job?
- For, that is interesting things to my job. Because I have my own searching behavior. Therefore, I think searching work will be very enjoyable for me. Innovator and industrious team and promoter good leader will be most important to me in my job.
10. What do you know about the job?
- My career is telecommunication engineer. Almost we fix the broken cable. But women’s often do operator. In my opinion I want be header engineer. So I need I can any thing.
11. How important is communication and interaction with others on your job?
- it is most important issue. I demand to interactive with all user. Because I should know for request of subscribers.
12. Do you prefer to work by yourself or with others?
- I prefer to work with others. Because I think that good employee must can work by oneself in indispensable time. Also I have good skills and experience to work with team.
- My long range goal is to be high level of professional specialist by my profession. Also I will prefer responsibility and good position which can give effective result to workers
2. What are your short-range goals and objectives?
- My short range goal is to improve my education. It will be preparation my future success. I will collect experience working in company. I hope there will be chance to me actualize my knowledge acquired from university to my job.
3. In what ways do you think you can make a contribution to our organization?
- I had been working on the optical network projects and these experiences would help your organization.
I think I can make my contribution improving my high level of education and my profession. Therefore, I will study by more degree of my profession besides of my job. I prefer my private and team contribution to organization. Contribution will be new idea to the electrical field.
4. What have you learned from participation in extra-curricular activities?
- I like participation in activities which outside from class. This has been become more and more abilities for me. And I skilled in teamwork and able to rise easily.
5. How do you determine or evaluate success?
- I like fair everything. So I want to determine for justices. Head person must say openly to all success. I usually define my success for estimation of other people.
6. How do you plan to achieve your career goals?
- I will continue my professional development by participating in conference, attending seminars and continuing my education.
7. What makes a job enjoyable for you?
-My work is a very interesting and when I was student I don’t know it. After I study this school I understand it
8. Why did you decide to seek a position with this communication organization?
-Even if I am 22, and have smaller experience than others, I consider I can work in this company successfully caused I possible to input or treat to my job via new method, initiative which minded from old idea.
9. What two or three things would be most important to you in your job?
- For, that is interesting things to my job. Because I have my own searching behavior. Therefore, I think searching work will be very enjoyable for me. Innovator and industrious team and promoter good leader will be most important to me in my job.
10. What do you know about the job?
- My career is telecommunication engineer. Almost we fix the broken cable. But women’s often do operator. In my opinion I want be header engineer. So I need I can any thing.
11. How important is communication and interaction with others on your job?
- it is most important issue. I demand to interactive with all user. Because I should know for request of subscribers.
12. Do you prefer to work by yourself or with others?
- I prefer to work with others. Because I think that good employee must can work by oneself in indispensable time. Also I have good skills and experience to work with team.
reading method
1 read the topic
2 think about this topic
3 read the first paragraph
4 If I don’t understand I need translate
5 And I find a new word from dictionary
6 so I will understand it
7 if it is interesting I read a full
8 if it boring I close the book
9 I try to find a knowledge
10think about learned thing from a this book
11 write a abstract
12 I must read it to remember when I forget this book
13 finally I put my book in my library
2 think about this topic
3 read the first paragraph
4 If I don’t understand I need translate
5 And I find a new word from dictionary
6 so I will understand it
7 if it is interesting I read a full
8 if it boring I close the book
9 I try to find a knowledge
10think about learned thing from a this book
11 write a abstract
12 I must read it to remember when I forget this book
13 finally I put my book in my library
TRANSLATION METHODS
1. I read to translate of text title.
2.Thus make word to word translation.
3. Next I take out by translated word of title content.
4. I read first paragraph.
5. Next I make word to word translation first sentence
6. I read and understand out by translated word of sentence content.
7. Need look conjunction, gram subject and predicate while take out of sentence content.
8. Read next sentences translate by likeness with above mentioned.
9. Composly writing and reading the first paragraph.
10. Read next paragraph.
11. Starting translate by likeness with above mentioned the first sentence.
12. Composly writing the paragraphs.
13. Understand main idea of source.
14. Memorize the translated new words.
15. Finally I will glad in my own wrought of translation by result.
МИНИЙ ОРЧУУЛДАГ АРГА
1. Эхлээд би орчуулах тэкстийнхээ гарчгийг уншдаг.
2.Тэгээд гарчгаа үгчлэн орчуулна.
3. Дараа нь би орчуулсан үгсээрээ гарчгийнхаа утгыг гаргаж авна.
4. Тэгээд би эхний параграфыг уншина.
5.Дараа нь нэгдүгээр өгүүлбэрээс эхлэн үгчлэн орчуулна.
6.Орчуулсан үгсийнхээ тусламжтай эхний өгүүлбэрээ найруулан бичиж уншиж ойлгоно. 7.Өгүүлбэрийн утгыг гаргаж авах үедээ холбоос үг, өгүүлэхүүн, өгүүлэгдэхүүн зэргийг харах хэрэгтэй.
8.Дараагийн өгүүлбэрүүдээ уншиж дээрх аргаар орчуулна.
9.Эхний параграфаа найруулан бичиж уншина.
10.Дараагийн параграфыг уншина.
11.Тэгээд эхний өгүүлбэрээс эхлэн дээрх аргаар орчуулна.
12.Параграфуудаа найруулан бичнэ.
13.Эхийн гол утга санааг уншиж ойлгоно.
14.Орчуулсан шинэ үгсээ цээжлэнэ.
15.Эцэст нь би өөрийнхөө хийсэн орчуулагын үр дүнд сэтгэл хангалуун байх болно.
2.Thus make word to word translation.
3. Next I take out by translated word of title content.
4. I read first paragraph.
5. Next I make word to word translation first sentence
6. I read and understand out by translated word of sentence content.
7. Need look conjunction, gram subject and predicate while take out of sentence content.
8. Read next sentences translate by likeness with above mentioned.
9. Composly writing and reading the first paragraph.
10. Read next paragraph.
11. Starting translate by likeness with above mentioned the first sentence.
12. Composly writing the paragraphs.
13. Understand main idea of source.
14. Memorize the translated new words.
15. Finally I will glad in my own wrought of translation by result.
МИНИЙ ОРЧУУЛДАГ АРГА
1. Эхлээд би орчуулах тэкстийнхээ гарчгийг уншдаг.
2.Тэгээд гарчгаа үгчлэн орчуулна.
3. Дараа нь би орчуулсан үгсээрээ гарчгийнхаа утгыг гаргаж авна.
4. Тэгээд би эхний параграфыг уншина.
5.Дараа нь нэгдүгээр өгүүлбэрээс эхлэн үгчлэн орчуулна.
6.Орчуулсан үгсийнхээ тусламжтай эхний өгүүлбэрээ найруулан бичиж уншиж ойлгоно. 7.Өгүүлбэрийн утгыг гаргаж авах үедээ холбоос үг, өгүүлэхүүн, өгүүлэгдэхүүн зэргийг харах хэрэгтэй.
8.Дараагийн өгүүлбэрүүдээ уншиж дээрх аргаар орчуулна.
9.Эхний параграфаа найруулан бичиж уншина.
10.Дараагийн параграфыг уншина.
11.Тэгээд эхний өгүүлбэрээс эхлэн дээрх аргаар орчуулна.
12.Параграфуудаа найруулан бичнэ.
13.Эхийн гол утга санааг уншиж ойлгоно.
14.Орчуулсан шинэ үгсээ цээжлэнэ.
15.Эцэст нь би өөрийнхөө хийсэн орчуулагын үр дүнд сэтгэл хангалуун байх болно.
Monday, December 6, 2010
abstract
The main purpose of both TDM and IP customer-premise switching systems is identical. They connect voice users in the office to one another and to external users over a pool of shared trunks. The differences lie in the way the two types of switch accomplish their objectives. This chapter discusses the principal features of CPE switching systems.
Most manufacturers offer separate key system and PBX product lines, although the functions may be nearly identical, at least in larger line sizes. Most TDM manufacturers also offer a cross between a PBX and a key system known as a hybrid.
sThese buttons and lamps define the following features, which are common to all key systems:
• Call pickup- Any station can access a line by pressing a line button.
• Call hold- A hold button (usually red) can be pressed to hold the line
in the central unit. By contrast, the hold button on a POTS phone holds
the line in the telephone so the line cannot be used for another call.
• Intercom- Acommon path shared by all telephones is used to announce calls.
• Supervisory signals- Lamps show when a line is ringing, in use, or on hold.
• Common bell- A bell common to all lines signals an incoming call. A slow
lamp flash shows which line is ringing.
• Automatic line selection: When the user picks up the phone, an outgoing
line is selected automatically.
• Bridged call appearance: The same extension number can be terminated
on multiple phone sets.
• Call drop: A call can be terminated without hanging up the receiver.
• Call forwarding: Users can forward their calls to another station
in the system.
• Call park: This feature places a call in a parking orbit so it can be
retrieved from any telephone in the system.
• Call transfer: An incoming or outgoing call can be transferred
to another user.
• Callback: If someone transfers a call to an extension that does not answer
after a set number of rings, the call returns to the original station.
• Camp-on: Users or the attendant can send an external call to another
telephone even if it is busy. The callee hears a faint camp-on tone.
When the user hangs up, the camped-on call rings at the station.
• Conferencing: Stations can bridge two or more lines together for
a multiparty conversation.
• Distinctive ringing: Different ringing tones enable users to distinguish
between internal and PSTN calls.
• Do not disturb: Users can press a button that silences the bell and
prevents intercom calls from reaching the station.
• Forward all calls: Users can redirect all calls to another station or
destination.
• Forward on busy or no answer: Users can redirect calls to another station
or destination if the line is busy or does not answer.
• Held-line reminder: After a call has been left on hold for a specified
period, the telephone emits a warning tone.
• Missed-call indicator: A list of unanswered calls is displayed
on the telephone.
• Music on hold: While a call is on hold, music or a promotional
announcement is played.
• Mute: A mute button on the telephone disables the microphone.
• Paging: Stations can page over the telephone speaker.
• Privacy: Prevents other stations from picking up a line that is in use.
In some systems privacy is automatic unless the user presses a privacy
release key.
• Station restriction: Stations can be assigned to different classes of service
for restricting long distance calls.
• Voice call: A user can place a call directly to the speaker of another user’s
telephone.
• Volume control: The volume of the handset, speaker, and ringer
can be adjusted.
The main features that most PBXs and many hybrids, both
TDM and IP, support. These features are in addition to the key system features
discussed in the previous section. Two features on the key system list, flash and
paging, are generally available on hybrids but unavailable on PBXs.
Direct Inward Dialing (DID)- DID offers station users the ability to receive calls from outside the system without going through the attendant.
Automatic Route Selection (ARS)- Most PBXs terminate a combination of public switched and private trunks on the system. For example, in addition to local trunks, the PBX may terminate T1/E1
lines to the IXC, FEX lines, and tie trunks to another PBX.
Networking Options- Most PBXs offer networking options, which allow multiple PBXs to operate as a
single system.
Direct Inward System Access (DISA)- The DISA feature enables external callers to dial a telephone number and password to gain access to PBX features.
N × 64 Capability- With the growth of video conferencing, it is often desirable to dial more bandwidth
than an ordinary BRI connection provides. Conference-quality video usually
requires at least 384 Kbps, which is six 64 Kbps channels. A PBX with N × 64
capability enables the user to dial as many channels of contiguous bandwidth as
required.
PBX Voice Features
As all PBXs are designed for voice switching service, they have features intended
for the convenience and productivity of the users. Not all the features listed below
are universally available, and many systems provide features not listed.
• Automatic call trace: Harassing or nuisance call can be traced to the origin
by dialing an access code.
• Call blocking: Users can selectively block calls such as specific extensions,
numbers, or calls from particular trunk groups.
• Call coverage: Users can have one or more coverage paths to direct
how calls route when the called station is busy, does not answer, or is
in do-not-disturb status. External calls can take a different path than
internal calls.
• Executive override: This feature allows a station to interrupt a busy line
or preempt a long distance trunk.
• Forced account code: On long distance calls, this feature prompts callers to
enter an identification code, which is registered on the CDR.
• Hoteling: A station user can temporarily move to another location, log in,
and have station features including the extension number follow to the
new location. Intervention from the administrator is not required.
• Paging access: The PBX can be equipped with paging trunks that
connect to an external paging system
• Personal call routing: Users can define routing of incoming calls based
on variables such as time of day, calling number, etc.
• Portable directory number: Allows a user on a networked PBX to move
from one switch to another without changing the telephone number.
• Priority ringing: A distinctive ring is used for calls from specified
numbers.
• Recorded announcements: This feature provides announcements for vacant
and disconnected numbers.
• Trunk answer any station: This feature allows stations to answer incoming
trunks when the attendant station is busy.
• Whisper page: A user can bridge into a call and speak to the local user
without the other end hearing.
Attendant Features
Most PBXs have attendant consoles for incoming call answer and supervision. The latter is increasingly popular because it can be easily integrated with a directory. The following features are important for most consoles and represent only a fraction of the features available.
• Attendant controlled conferencing: Attendant can set up multiport
conference calls.
• Automatic timed reminders: Alerts the attendant when a called line has not
answered within a prescribed time.
• Busy lamp field: When the station is busy or in do-not-disturb mode,
an LED associated with the station is lighted.
• Direct station selection (DSS): Allows the attendant to call stations by
pressing an illuminated button associated with the line.
• Directory features: Attendants with PC-based consoles may be able
to search by first and last name, department, and extension.
• Night service: Calls are automatically transferred to an alternate destination
when the console is closed.
System Administration Features
System administration is a costly element of every PBX, so features that ease
the administrator’s job are valuable. The following are some of the more popular
features.
• Automatic set relocation: Allows users to move their telephones from one
location to another without the need to retranslate. The administrator
gives users a code and instructions to carry the set to the new location,
plug it in, and dial the code.
• LDAP synchronization: Enables the system to update its PBX and voice
mail database from customer’s LDAP directory. Eliminates or reduces
redundant database entries
• Network move: Similar to automatic set relocation, this feature works
across a network, where automatic set relocation works only in the
same PBX.
TELECOMMUNICATIONS CONVERGENCE
Convergence: Merging real-time applications such as voice, video, and instant messaging together with data onto a single broadband infrastructure that is based on IP.
THE BUSINESS CASE CONVERGENCE
• Competitive Advantage- Businesses can tie customers to their internal processes through e-commerce.
• Lower Cost of Ownership- The cost equation is difficult to factor into the decision to migrate to a converged network.
• Flexibilty – The TDM PBX is a proprietary and inflexible device, closed in every respect except
for its CTI interface, which provides for limited call control.
• Create Value-Added Service- Convergence opens an enormous variety of opportunities to provide new services that are infeasible with circuit switching.
BARRIERS TO CONVERGENCE
New telecommunications developments have always been over-hyped, and with notable exceptions such as fiber optics, many of these have flared briefly and fizzled out.
• Network infrastructure- Carriers are converting portions of their network to VoIP, but islands of VoIP cannot support the performance and security that commercial-grade voice
• Flow control- Another key issue is congestion control, which is a vital feature of any voice or
Most manufacturers offer separate key system and PBX product lines, although the functions may be nearly identical, at least in larger line sizes. Most TDM manufacturers also offer a cross between a PBX and a key system known as a hybrid.
sThese buttons and lamps define the following features, which are common to all key systems:
• Call pickup- Any station can access a line by pressing a line button.
• Call hold- A hold button (usually red) can be pressed to hold the line
in the central unit. By contrast, the hold button on a POTS phone holds
the line in the telephone so the line cannot be used for another call.
• Intercom- Acommon path shared by all telephones is used to announce calls.
• Supervisory signals- Lamps show when a line is ringing, in use, or on hold.
• Common bell- A bell common to all lines signals an incoming call. A slow
lamp flash shows which line is ringing.
• Automatic line selection: When the user picks up the phone, an outgoing
line is selected automatically.
• Bridged call appearance: The same extension number can be terminated
on multiple phone sets.
• Call drop: A call can be terminated without hanging up the receiver.
• Call forwarding: Users can forward their calls to another station
in the system.
• Call park: This feature places a call in a parking orbit so it can be
retrieved from any telephone in the system.
• Call transfer: An incoming or outgoing call can be transferred
to another user.
• Callback: If someone transfers a call to an extension that does not answer
after a set number of rings, the call returns to the original station.
• Camp-on: Users or the attendant can send an external call to another
telephone even if it is busy. The callee hears a faint camp-on tone.
When the user hangs up, the camped-on call rings at the station.
• Conferencing: Stations can bridge two or more lines together for
a multiparty conversation.
• Distinctive ringing: Different ringing tones enable users to distinguish
between internal and PSTN calls.
• Do not disturb: Users can press a button that silences the bell and
prevents intercom calls from reaching the station.
• Forward all calls: Users can redirect all calls to another station or
destination.
• Forward on busy or no answer: Users can redirect calls to another station
or destination if the line is busy or does not answer.
• Held-line reminder: After a call has been left on hold for a specified
period, the telephone emits a warning tone.
• Missed-call indicator: A list of unanswered calls is displayed
on the telephone.
• Music on hold: While a call is on hold, music or a promotional
announcement is played.
• Mute: A mute button on the telephone disables the microphone.
• Paging: Stations can page over the telephone speaker.
• Privacy: Prevents other stations from picking up a line that is in use.
In some systems privacy is automatic unless the user presses a privacy
release key.
• Station restriction: Stations can be assigned to different classes of service
for restricting long distance calls.
• Voice call: A user can place a call directly to the speaker of another user’s
telephone.
• Volume control: The volume of the handset, speaker, and ringer
can be adjusted.
The main features that most PBXs and many hybrids, both
TDM and IP, support. These features are in addition to the key system features
discussed in the previous section. Two features on the key system list, flash and
paging, are generally available on hybrids but unavailable on PBXs.
Direct Inward Dialing (DID)- DID offers station users the ability to receive calls from outside the system without going through the attendant.
Automatic Route Selection (ARS)- Most PBXs terminate a combination of public switched and private trunks on the system. For example, in addition to local trunks, the PBX may terminate T1/E1
lines to the IXC, FEX lines, and tie trunks to another PBX.
Networking Options- Most PBXs offer networking options, which allow multiple PBXs to operate as a
single system.
Direct Inward System Access (DISA)- The DISA feature enables external callers to dial a telephone number and password to gain access to PBX features.
N × 64 Capability- With the growth of video conferencing, it is often desirable to dial more bandwidth
than an ordinary BRI connection provides. Conference-quality video usually
requires at least 384 Kbps, which is six 64 Kbps channels. A PBX with N × 64
capability enables the user to dial as many channels of contiguous bandwidth as
required.
PBX Voice Features
As all PBXs are designed for voice switching service, they have features intended
for the convenience and productivity of the users. Not all the features listed below
are universally available, and many systems provide features not listed.
• Automatic call trace: Harassing or nuisance call can be traced to the origin
by dialing an access code.
• Call blocking: Users can selectively block calls such as specific extensions,
numbers, or calls from particular trunk groups.
• Call coverage: Users can have one or more coverage paths to direct
how calls route when the called station is busy, does not answer, or is
in do-not-disturb status. External calls can take a different path than
internal calls.
• Executive override: This feature allows a station to interrupt a busy line
or preempt a long distance trunk.
• Forced account code: On long distance calls, this feature prompts callers to
enter an identification code, which is registered on the CDR.
• Hoteling: A station user can temporarily move to another location, log in,
and have station features including the extension number follow to the
new location. Intervention from the administrator is not required.
• Paging access: The PBX can be equipped with paging trunks that
connect to an external paging system
• Personal call routing: Users can define routing of incoming calls based
on variables such as time of day, calling number, etc.
• Portable directory number: Allows a user on a networked PBX to move
from one switch to another without changing the telephone number.
• Priority ringing: A distinctive ring is used for calls from specified
numbers.
• Recorded announcements: This feature provides announcements for vacant
and disconnected numbers.
• Trunk answer any station: This feature allows stations to answer incoming
trunks when the attendant station is busy.
• Whisper page: A user can bridge into a call and speak to the local user
without the other end hearing.
Attendant Features
Most PBXs have attendant consoles for incoming call answer and supervision. The latter is increasingly popular because it can be easily integrated with a directory. The following features are important for most consoles and represent only a fraction of the features available.
• Attendant controlled conferencing: Attendant can set up multiport
conference calls.
• Automatic timed reminders: Alerts the attendant when a called line has not
answered within a prescribed time.
• Busy lamp field: When the station is busy or in do-not-disturb mode,
an LED associated with the station is lighted.
• Direct station selection (DSS): Allows the attendant to call stations by
pressing an illuminated button associated with the line.
• Directory features: Attendants with PC-based consoles may be able
to search by first and last name, department, and extension.
• Night service: Calls are automatically transferred to an alternate destination
when the console is closed.
System Administration Features
System administration is a costly element of every PBX, so features that ease
the administrator’s job are valuable. The following are some of the more popular
features.
• Automatic set relocation: Allows users to move their telephones from one
location to another without the need to retranslate. The administrator
gives users a code and instructions to carry the set to the new location,
plug it in, and dial the code.
• LDAP synchronization: Enables the system to update its PBX and voice
mail database from customer’s LDAP directory. Eliminates or reduces
redundant database entries
• Network move: Similar to automatic set relocation, this feature works
across a network, where automatic set relocation works only in the
same PBX.
TELECOMMUNICATIONS CONVERGENCE
Convergence: Merging real-time applications such as voice, video, and instant messaging together with data onto a single broadband infrastructure that is based on IP.
THE BUSINESS CASE CONVERGENCE
• Competitive Advantage- Businesses can tie customers to their internal processes through e-commerce.
• Lower Cost of Ownership- The cost equation is difficult to factor into the decision to migrate to a converged network.
• Flexibilty – The TDM PBX is a proprietary and inflexible device, closed in every respect except
for its CTI interface, which provides for limited call control.
• Create Value-Added Service- Convergence opens an enormous variety of opportunities to provide new services that are infeasible with circuit switching.
BARRIERS TO CONVERGENCE
New telecommunications developments have always been over-hyped, and with notable exceptions such as fiber optics, many of these have flared briefly and fizzled out.
• Network infrastructure- Carriers are converting portions of their network to VoIP, but islands of VoIP cannot support the performance and security that commercial-grade voice
• Flow control- Another key issue is congestion control, which is a vital feature of any voice or
new words
CHAPTER 24
1. Customer-Premise Switching
System Features- хэрэглэгчийн холболтын системын онцлог
2. Accustomed- дассан, сурсан
3. Predecessor- өмнөх хүн
4. Regardless- үл тоомсорлосон, эс анхаарсан
5. Pooling- нэгдэл
6. Requirements- шаардлага, хэрэгцээ
7. Platform- тавцан, мөрийн хөтөлбөр
8. Impossible- боломжгүй, бүтэшгүй
9. Investmen- хөрөнгө оруулалт
10. Principal- гол, үндсэн,
11. Hybrid- эрлийз
12. Distinctions- ялгаа, зөрөө, онцлог шинж
13. Absolute- бүрэн, бүрэн төгс
14. Varied- өөр,төрөл бүрийн
15. Available- хүчинтэй
16. Common- ерөнхий, нийтийн
17. Terminated- дуусгах, төгсгөх
18. Button – товчлуур
19. Invariably- хувиралгүйгээр
20. Retrieve- буцаах, сэргээх
21. Pressing- яаралтай, чухал
22. Announced- зарлах, мэдэгдэх
23. Intercom- хоёр талын шуурхай холбоо
24. Dozen- олон тооны. их хэмжээтэй, олон
25. Electromechanical- цахилгаан механикийн
26. Comparison- харьцуулалт
27. Terminate- төгсгөх, төгсөх
28. Attendant- үйлчилж байгаа
29. Directory- удирдлага, лавлах
30. Transfer- шилжүүлэг, зөөвөр
31. Compatibility- нийцэх, таарах
32. Illuminated- тайлбарлах, гэрэлтүүлэх
33. Denotes- тэмдэглэх, төлөөлөх
34. Solid- биет, бат бөх
35. Call pickup- дуудлага цуглуулагч
36. Call hold- дуудлага барих
37. Intercom- дотоод холбоо
38. Supervisory signals- хянагч дохио
39. Common bell- ерөнхий хонх
40. Temporarily- түр зуур
41. Extension- өргөтгөл, нэмэлт
42. Intervention- хөндлөнгөөс оролцох
1. Customer-Premise Switching
System Features- хэрэглэгчийн холболтын системын онцлог
2. Accustomed- дассан, сурсан
3. Predecessor- өмнөх хүн
4. Regardless- үл тоомсорлосон, эс анхаарсан
5. Pooling- нэгдэл
6. Requirements- шаардлага, хэрэгцээ
7. Platform- тавцан, мөрийн хөтөлбөр
8. Impossible- боломжгүй, бүтэшгүй
9. Investmen- хөрөнгө оруулалт
10. Principal- гол, үндсэн,
11. Hybrid- эрлийз
12. Distinctions- ялгаа, зөрөө, онцлог шинж
13. Absolute- бүрэн, бүрэн төгс
14. Varied- өөр,төрөл бүрийн
15. Available- хүчинтэй
16. Common- ерөнхий, нийтийн
17. Terminated- дуусгах, төгсгөх
18. Button – товчлуур
19. Invariably- хувиралгүйгээр
20. Retrieve- буцаах, сэргээх
21. Pressing- яаралтай, чухал
22. Announced- зарлах, мэдэгдэх
23. Intercom- хоёр талын шуурхай холбоо
24. Dozen- олон тооны. их хэмжээтэй, олон
25. Electromechanical- цахилгаан механикийн
26. Comparison- харьцуулалт
27. Terminate- төгсгөх, төгсөх
28. Attendant- үйлчилж байгаа
29. Directory- удирдлага, лавлах
30. Transfer- шилжүүлэг, зөөвөр
31. Compatibility- нийцэх, таарах
32. Illuminated- тайлбарлах, гэрэлтүүлэх
33. Denotes- тэмдэглэх, төлөөлөх
34. Solid- биет, бат бөх
35. Call pickup- дуудлага цуглуулагч
36. Call hold- дуудлага барих
37. Intercom- дотоод холбоо
38. Supervisory signals- хянагч дохио
39. Common bell- ерөнхий хонх
40. Temporarily- түр зуур
41. Extension- өргөтгөл, нэмэлт
42. Intervention- хөндлөнгөөс оролцох
application
Company name: Telecommunication
Register number: ia90022304
Health insurance :48 .
Passport number: 2472003
Sex: female
1. Last name: Davaanyam
2. First name: Baasanjav
3. Birthday: 1990.02.23
4. Parentage: xalx
6. Family :6
Your Last name, first name , Birth year , Job
Father Chimedravdan, Davaanyam , ,1950.03.05 , Herder
Mother Tseveenjav, Uranchimeg , ,1950.09.05 , Herder
7. Education
Where? What school? Entried year Graduated year Profession, diplom number
Uliastai, Zavkhan “Chandmani-Erdene tsogtsolbor” 1998- 2008 and now
Infornmation Communication Technology University 2008- 2012 Telecommunication and network engineer
8. Home address: Uliastai, Zavkhan
9.phone number:98466696
10. About prize:Year What prize
11. Degree: bachelor. ¹ Year Month Day Office Writ date
Register number: ia90022304
Health insurance :48 .
Passport number: 2472003
Sex: female
1. Last name: Davaanyam
2. First name: Baasanjav
3. Birthday: 1990.02.23
4. Parentage: xalx
6. Family :6
Your Last name, first name , Birth year , Job
Father Chimedravdan, Davaanyam , ,1950.03.05 , Herder
Mother Tseveenjav, Uranchimeg , ,1950.09.05 , Herder
7. Education
Where? What school? Entried year Graduated year Profession, diplom number
Uliastai, Zavkhan “Chandmani-Erdene tsogtsolbor” 1998- 2008 and now
Infornmation Communication Technology University 2008- 2012 Telecommunication and network engineer
8. Home address: Uliastai, Zavkhan
9.phone number:98466696
10. About prize:Year What prize
11. Degree: bachelor. ¹ Year Month Day Office Writ date
reference letter
Davaanyam Baasanjav
Recommendation letter
School of Telecommunication and Information technology,MUST
Apt 30-47, 5th khoroo, Bayanzurkh district, Post Box-7116, Post Office Ulaanbaatar- 56
Ulaanbaatar Mongolia,
To: "MIAT" company
I have known Baasanjav for 3 years. During the years of our acquaintance, I have seen many abilities of his. He is able to study anything new and always tries to know anything new. I strongly recommend his for your offering position of work.
As Baasanjav's teacher, have given his many to asks to do and she always done it on time. He skilled in communication software and hardware. He also has experience in Switching System as engineer and solved many problems occurred on Switching System.
If you're looking for experienced candidate with ability to rise, Baasanjav is an excellent choice. She is self-educated, responsible, skilled in responsible and able to rise easily. Please do not hesitate to contact us. We will be glad to answer further question about Baasanjav.
Sincerely ,
lecturer: Bayarmaa. D
Recommendation letter
School of Telecommunication and Information technology,MUST
Apt 30-47, 5th khoroo, Bayanzurkh district, Post Box-7116, Post Office Ulaanbaatar- 56
Ulaanbaatar Mongolia,
To: "MIAT" company
I have known Baasanjav for 3 years. During the years of our acquaintance, I have seen many abilities of his. He is able to study anything new and always tries to know anything new. I strongly recommend his for your offering position of work.
As Baasanjav's teacher, have given his many to asks to do and she always done it on time. He skilled in communication software and hardware. He also has experience in Switching System as engineer and solved many problems occurred on Switching System.
If you're looking for experienced candidate with ability to rise, Baasanjav is an excellent choice. She is self-educated, responsible, skilled in responsible and able to rise easily. Please do not hesitate to contact us. We will be glad to answer further question about Baasanjav.
Sincerely ,
lecturer: Bayarmaa. D
cover letter
2010.10.25
baaskad022gmail.com
Baasanjav davaanyam
Ulaanbaatar,
Telecommunication engineer
To” MIAT” company
Dear Zorigtbaatar:
To work as telecommunication’s engineer in your company. The reason that it write this letter is meet my requirements as your action company’s placement and public’s position. I finished school of Information and technology ago 3 years. My previous work in Mobicom was by engineer.
I graduated in Mongolian university of information telecommunication’s engineer with in bachelor 2012. I interesting for the profession and want to do so much. If I will work in the company . I will become to work hard because I ‘m widening more than our public’s position and introducing new technology.
You will find me to be a strong analytical problem solver possessing the communication skills to actively manage and work in cooperation with teams and individuals to achieve desired goals efficiently. I have demonstrated succees in doing so over the past several years and intend to continue this trend long into the future .
I am confident that you will be pleased with the skills and experience portrayed in the accompanying resume. I will call your office in a few days inquire about the possibility of a meeting
Thank you in advance for your time and consideration .
Sincerely Yours,
Baasanjav, Davaanyam
baaskad022gmail.com
Baasanjav davaanyam
Ulaanbaatar,
Telecommunication engineer
To” MIAT” company
Dear Zorigtbaatar:
To work as telecommunication’s engineer in your company. The reason that it write this letter is meet my requirements as your action company’s placement and public’s position. I finished school of Information and technology ago 3 years. My previous work in Mobicom was by engineer.
I graduated in Mongolian university of information telecommunication’s engineer with in bachelor 2012. I interesting for the profession and want to do so much. If I will work in the company . I will become to work hard because I ‘m widening more than our public’s position and introducing new technology.
You will find me to be a strong analytical problem solver possessing the communication skills to actively manage and work in cooperation with teams and individuals to achieve desired goals efficiently. I have demonstrated succees in doing so over the past several years and intend to continue this trend long into the future .
I am confident that you will be pleased with the skills and experience portrayed in the accompanying resume. I will call your office in a few days inquire about the possibility of a meeting
Thank you in advance for your time and consideration .
Sincerely Yours,
Baasanjav, Davaanyam
Wednesday, November 24, 2010
chapter39
Lack of Carrier Agreements
IP networks are today a loose confederation of agreements among carriers.
A carrier-class VoIP network cannot depend on a service provider’s decision
whether to carry transit traffic. This, in turn, means that international division of
revenue processes must be developed. Competing carriers must deal promptly
with impairments and failures and must cooperate to ensure that service is not
affected by the structure of the interface points of diverse networks.
Carrier and Regulator Inertia
The existing PSTN infrastructure works well, represents a considerable investment,
and is maintained and managed by workers that lack the skills to manage an
all IP infrastructure. ILECs, in particular, will retain what they have until it is
functionally obsolete or competitive pressures force them to change. Numerous
other forces resist change. Congress is subject to numerous pressures that affect
the shape telecommunications will take, and money is at the root of it. Cash-rich
companies such as the ILECs have the ability to inject huge amounts of money to
purchase influence and they have been successful in this endeavor.
The regulatory framework is based on conventional telephone technology as
are the structure of taxes and fees and these tend to change slowly. Much of the
economies of VoIP in the U.S. today result from Congress’ reluctance to burden IP
with the fees, restrictions, and taxes that it loads on the conventional telephone
industry. Access charges, USF fees, excise taxes, and myriad state and local taxes,
not to mention carriers’ miscellaneous fees complicate the picture. Moreover,
regulations are inconsistent. LEC cable pairs are subject to unbundled access
regulations, but new fiber investments are not. Cable companies can prohibit
other carriers from using their access facilities and have both the ability and the
motivation to exclude other providers from using them for VoIP. All of this creates
an atmosphere in which progress is impeded because there is little assurance that
the rules will not change.
Reliability and Availability
IP networks are inherently robust with their ability to route around failures, but
router convergence time is too lengthy for real-time applications. Operational
changes such as hardware and software upgrades and configuration changes cannot
be done in real time with many router platforms. The solution requires a new
generation of routers and protocols, which will be slow to propagate themselves
through the network.
Compliance Issues
Making VoIP comply with E-911 and CALEA is a major unresolved problem.
Many countries censor Web content, block access to certain sites, or monitor access
to parts of the Internet. These requirements are incompatible with using the
Internet as a reliable communications channel.
Interworking between the PSTN and IP
Until the transition to an all-IP infrastructure is complete, communication
between the two networks is required. This requires gateways, signaling, addressing,
and numerous other complexities that must be transparent to the users. Much
work remains to be done to make the interface with the PSTN seamless.
THE IPsphere INITIATIVE
Many of the changes necessary to make VoIP equivalent to the PSTN require
voluntary or compulsory adherence to centralized authority, a concept that is anathema
to the Internet community. If we conclude that the Internet in its present state is
not an appropriate vehicle for isochronous traffic, then the solution may be an overlay
network that is designed to provide the elements missing from the Internet: QoS,
predictability, end-to-end management, security, and carrier interconnections with
division of revenues or settlement process. Such a network would not be accessible
from the public Internet and would provide the benefits of IP without subjecting its
subscribers to the chaotic conditions that prevail on the Internet today.
An initiative known as the IPsphere, previously known as the Infranet is
underway to create such a network. Equipment manufacturers and carriers are
not unanimous in their support of the IPsphere and there is no assurance that
such a network will happen. The goal of IPsphere is to deliver performance over
a virtual network that is predictable, flexible, and secure so subscribers can entrust
mission-critical information to it. At the edge of the IPsphere is a barrier that
requires customers to authenticate themselves before admission.
The IPsphere requires communication between the subscriber’s application
and the network to enable the application to request the level of security, quality,
and bandwidth it needs. Costs would be based on what the application needs, in
contrast to the Internet where the cost is independent of the application. Since no
single provider can guarantee worldwide connectivity, connections are needed
between networks so providers can communicate levels of service and security
when handing off traffic. In addition, accounting mechanisms are needed to
enable carriers to bill each other for carrying traffic.
CONVERGENCE APPLICATION ISSUES
The motivation for the converged network in the enterprise network can be
summed up in one word: productivity, both of personnel and capital. In the public
network the motivation is also clear. For consumers it is saving money and getting
enhanced services. For service providers it is the opportunity to make money. It
will be many years before the technology has advanced to the point of replacing the
PSTN for more than a narrow spectrum of users, but the obstacles will gradually
be surmounted.
VoIP in the Public Network
At this stage of development, the ways of implementing VoIP are many and varied.
From a residential or small business user’s standpoint the ideal would be to connect
a VoIP telephone into a wired or wireless LAN, assign it a telephone number, and
use it as if it were a wired phone. The reality, however, is considerably different.
Several companies have jumped onto the VoIP bandwagon and by the time this is
published, many more would have joined the fray. IDC estimates that by 2008 some
14 million customers worldwide will subscribe to VoIP services. In this section we
briefly discuss the alternative configurations and some of the considerations in
selecting them. The services they offer are changing, however, so it is best to refer
to the vendor’s description on its Web page before relying on this discussion.
Computer-to-Computer Services
Representative services include Pulver, Skype, and Dialpad. These services
generally cannot be reached from the PSTN because the user does not have an
E.164 number. In the process of registering with the service provider, the user
obtains an address that is valid within that network. Calls within the network can
be made to others who have registered and are online.
The telephone instrument is usually a softphone and client, which can be
downloaded without cost. These services usually do not carry a monthly fee,
although some offer off-net prepaid packages. PC-to-PC calling is free. Calls that
hop off to a wireline phone carry a charge that is usually much lower than small
users can obtain, but may not be much of a saving for large users. The greatest
savings are on international calls. Features include buddy lists, redial, and running
account balance.
Firewalls and NAT generally do not bother these types of service because
nothing identifies the call as a voice call. To place or receive calls the user logs onto
the provider’s Web service, so the session looks to the network like any other Web
connection.
Computer to PSTN Services
Representative services include AT&T CallVantage, Vonage, Net2Phone, and
Go2Call. A major difference between these and computer-to-computer services is
the provision of an E.164 number, which enables the IP phone to be called as if it
were a wireline phone. Since the service provider controls the design, the portion
of the call that uses the Internet is under its control, so the service provider can
control the quality. The architecture is invisible to the customer, so there is no way
to evaluate the service in advance except to try it.
Some products permit or require the use of a VoIP terminal adapter, which
connects an analog phone to the network. Some will work with downloaded softphone
products. In most cases existing numbers can be ported to the IP service, and
the IP phone is not tied to a physical location. This means it can be transported to
another place and operate as it does from the primary location. Vonage offers a virtual
number in certain local calling areas. This permits someone with a landline telephone
to make a local call, which is transported to the destination across the Internet.
Most of these services carry a monthly rate, which may include unlimited
domestic calling. The rates are generally lower than LEC phones, and may include
a package of special features that the LEC charges for. Voice mail is typically available,
with message retrieval from either telephone or browser. Voice mail messages
can also be forwarded as e-mail attachments. Other typical features include
caller ID, call forwarding, call logging, do-not-disturb, conferencing, and locate
service.
The service has much in common with cell phone service. Most users will
not give up their PSTN phone, but it is a good solution for a second line. The
service has several downsides that must be considered, not the least of which is
lack of compatibility with E-911. In addition, it does not work through power
failures. If the line is to be used as an additional line throughout the house or
business, the adapter must be wired in place, which defeats the easy portability.
Finally, the service is not as simple to set up as buying a telephone and plugging
it in. The author’s experience with AT&T CallVantage is a case in point. After
several hours of attempting to make it work through Comcast cable, the AT&T
technician gave up concluding that Comcast’s routers were blocking the service.
TDM over IP (TDMoIP) Multiplexers
Several manufacturers provide TDMoIP multiplexers for applications such
as the one shown in Figure 39-1. Here, the company has an Ethernet connection
between sites and T1/E1 compatible PBX and key systems. The IP multiplexers connect
to the CPE devices with TDM and to the Ethernet switches with 100Base-T,
sharing the bandwidth between sites with data. The multiplexers shown use T1
on both ends, but they could just as easily use analog trunks, in which case the
connection would typically be FXS/FXO. The TDM frames are encapsulated into
IP packets that are transported over the fast Ethernet ports. The ToS bits of the
packets are set to classify the packets as high priority.
TDMoIP provides a circuit-emulation service, also called pseudowire, that
is transparent to protocols and signaling. The multiplexer repeats the contents
of each channel to the other end. The IETF PWE3 Working Group is working
on protocol standards that are in draft form as this book is published, so products
are likely to be proprietary. Typically, the payload of each channel connects to a
48-octet ATM cell, which does not have the 5-octet header. These are encapsulated
into IP frames in some multiple. As the number of TDM octets per frame increases
throughput increases because of lower packet overhead, but the effects of frame
loss are more severe.
Compared to VoIP gateways, TDMoIP multiplexers have lower latency, so
the circuit quality is likely to be better, provided packet loss is not excessive. As
with VoIP gateways, the multiplexers compensate for packet loss by repeating the
contents of the previous packet. The multiplexer transmits channel timeslots
whether they are empty or not, so from this standpoint it is less bandwidthefficient
than VoIP. The impact of this is generally irrelevant, however, because a
TDMoIP multiplexer is not used where bandwidth is restricted. For example, it
would not be usable over the Internet. Some products support fractional T1/E1 so
vacant channels are not transmitted.
VoIP Business Applications
VoIP has four principal categories of applications in the enterprise network:
_ Branch office. Since the branch office is already equipped with a router
or FRAD, adding voice may be a natural outgrowth of the existing
network. For distant branch offices, the existing frame relay network
may be used. For local branches, it may be feasible to replace off-premise
extensions with VoIP.
_ Telecommuting. Similar in architecture to the branch office, telecommuters
may use VoIP to provide a voice and data connection to the main office.
_ Toll bypass. Carrying long distance traffic for next to nothing is attractive,
particularly when the distances are great. The ability to send large
quantities of fax messages, which tend not to be time sensitive, over
the Internet, can be particularly attractive.
_ Web-enabled call center. Callers can browse the company’s Web page
by computer, and then click an icon to talk to a live agent. Costs are
reduced and customer convenience is enhanced.
IP networks are today a loose confederation of agreements among carriers.
A carrier-class VoIP network cannot depend on a service provider’s decision
whether to carry transit traffic. This, in turn, means that international division of
revenue processes must be developed. Competing carriers must deal promptly
with impairments and failures and must cooperate to ensure that service is not
affected by the structure of the interface points of diverse networks.
Carrier and Regulator Inertia
The existing PSTN infrastructure works well, represents a considerable investment,
and is maintained and managed by workers that lack the skills to manage an
all IP infrastructure. ILECs, in particular, will retain what they have until it is
functionally obsolete or competitive pressures force them to change. Numerous
other forces resist change. Congress is subject to numerous pressures that affect
the shape telecommunications will take, and money is at the root of it. Cash-rich
companies such as the ILECs have the ability to inject huge amounts of money to
purchase influence and they have been successful in this endeavor.
The regulatory framework is based on conventional telephone technology as
are the structure of taxes and fees and these tend to change slowly. Much of the
economies of VoIP in the U.S. today result from Congress’ reluctance to burden IP
with the fees, restrictions, and taxes that it loads on the conventional telephone
industry. Access charges, USF fees, excise taxes, and myriad state and local taxes,
not to mention carriers’ miscellaneous fees complicate the picture. Moreover,
regulations are inconsistent. LEC cable pairs are subject to unbundled access
regulations, but new fiber investments are not. Cable companies can prohibit
other carriers from using their access facilities and have both the ability and the
motivation to exclude other providers from using them for VoIP. All of this creates
an atmosphere in which progress is impeded because there is little assurance that
the rules will not change.
Reliability and Availability
IP networks are inherently robust with their ability to route around failures, but
router convergence time is too lengthy for real-time applications. Operational
changes such as hardware and software upgrades and configuration changes cannot
be done in real time with many router platforms. The solution requires a new
generation of routers and protocols, which will be slow to propagate themselves
through the network.
Compliance Issues
Making VoIP comply with E-911 and CALEA is a major unresolved problem.
Many countries censor Web content, block access to certain sites, or monitor access
to parts of the Internet. These requirements are incompatible with using the
Internet as a reliable communications channel.
Interworking between the PSTN and IP
Until the transition to an all-IP infrastructure is complete, communication
between the two networks is required. This requires gateways, signaling, addressing,
and numerous other complexities that must be transparent to the users. Much
work remains to be done to make the interface with the PSTN seamless.
THE IPsphere INITIATIVE
Many of the changes necessary to make VoIP equivalent to the PSTN require
voluntary or compulsory adherence to centralized authority, a concept that is anathema
to the Internet community. If we conclude that the Internet in its present state is
not an appropriate vehicle for isochronous traffic, then the solution may be an overlay
network that is designed to provide the elements missing from the Internet: QoS,
predictability, end-to-end management, security, and carrier interconnections with
division of revenues or settlement process. Such a network would not be accessible
from the public Internet and would provide the benefits of IP without subjecting its
subscribers to the chaotic conditions that prevail on the Internet today.
An initiative known as the IPsphere, previously known as the Infranet is
underway to create such a network. Equipment manufacturers and carriers are
not unanimous in their support of the IPsphere and there is no assurance that
such a network will happen. The goal of IPsphere is to deliver performance over
a virtual network that is predictable, flexible, and secure so subscribers can entrust
mission-critical information to it. At the edge of the IPsphere is a barrier that
requires customers to authenticate themselves before admission.
The IPsphere requires communication between the subscriber’s application
and the network to enable the application to request the level of security, quality,
and bandwidth it needs. Costs would be based on what the application needs, in
contrast to the Internet where the cost is independent of the application. Since no
single provider can guarantee worldwide connectivity, connections are needed
between networks so providers can communicate levels of service and security
when handing off traffic. In addition, accounting mechanisms are needed to
enable carriers to bill each other for carrying traffic.
CONVERGENCE APPLICATION ISSUES
The motivation for the converged network in the enterprise network can be
summed up in one word: productivity, both of personnel and capital. In the public
network the motivation is also clear. For consumers it is saving money and getting
enhanced services. For service providers it is the opportunity to make money. It
will be many years before the technology has advanced to the point of replacing the
PSTN for more than a narrow spectrum of users, but the obstacles will gradually
be surmounted.
VoIP in the Public Network
At this stage of development, the ways of implementing VoIP are many and varied.
From a residential or small business user’s standpoint the ideal would be to connect
a VoIP telephone into a wired or wireless LAN, assign it a telephone number, and
use it as if it were a wired phone. The reality, however, is considerably different.
Several companies have jumped onto the VoIP bandwagon and by the time this is
published, many more would have joined the fray. IDC estimates that by 2008 some
14 million customers worldwide will subscribe to VoIP services. In this section we
briefly discuss the alternative configurations and some of the considerations in
selecting them. The services they offer are changing, however, so it is best to refer
to the vendor’s description on its Web page before relying on this discussion.
Computer-to-Computer Services
Representative services include Pulver, Skype, and Dialpad. These services
generally cannot be reached from the PSTN because the user does not have an
E.164 number. In the process of registering with the service provider, the user
obtains an address that is valid within that network. Calls within the network can
be made to others who have registered and are online.
The telephone instrument is usually a softphone and client, which can be
downloaded without cost. These services usually do not carry a monthly fee,
although some offer off-net prepaid packages. PC-to-PC calling is free. Calls that
hop off to a wireline phone carry a charge that is usually much lower than small
users can obtain, but may not be much of a saving for large users. The greatest
savings are on international calls. Features include buddy lists, redial, and running
account balance.
Firewalls and NAT generally do not bother these types of service because
nothing identifies the call as a voice call. To place or receive calls the user logs onto
the provider’s Web service, so the session looks to the network like any other Web
connection.
Computer to PSTN Services
Representative services include AT&T CallVantage, Vonage, Net2Phone, and
Go2Call. A major difference between these and computer-to-computer services is
the provision of an E.164 number, which enables the IP phone to be called as if it
were a wireline phone. Since the service provider controls the design, the portion
of the call that uses the Internet is under its control, so the service provider can
control the quality. The architecture is invisible to the customer, so there is no way
to evaluate the service in advance except to try it.
Some products permit or require the use of a VoIP terminal adapter, which
connects an analog phone to the network. Some will work with downloaded softphone
products. In most cases existing numbers can be ported to the IP service, and
the IP phone is not tied to a physical location. This means it can be transported to
another place and operate as it does from the primary location. Vonage offers a virtual
number in certain local calling areas. This permits someone with a landline telephone
to make a local call, which is transported to the destination across the Internet.
Most of these services carry a monthly rate, which may include unlimited
domestic calling. The rates are generally lower than LEC phones, and may include
a package of special features that the LEC charges for. Voice mail is typically available,
with message retrieval from either telephone or browser. Voice mail messages
can also be forwarded as e-mail attachments. Other typical features include
caller ID, call forwarding, call logging, do-not-disturb, conferencing, and locate
service.
The service has much in common with cell phone service. Most users will
not give up their PSTN phone, but it is a good solution for a second line. The
service has several downsides that must be considered, not the least of which is
lack of compatibility with E-911. In addition, it does not work through power
failures. If the line is to be used as an additional line throughout the house or
business, the adapter must be wired in place, which defeats the easy portability.
Finally, the service is not as simple to set up as buying a telephone and plugging
it in. The author’s experience with AT&T CallVantage is a case in point. After
several hours of attempting to make it work through Comcast cable, the AT&T
technician gave up concluding that Comcast’s routers were blocking the service.
TDM over IP (TDMoIP) Multiplexers
Several manufacturers provide TDMoIP multiplexers for applications such
as the one shown in Figure 39-1. Here, the company has an Ethernet connection
between sites and T1/E1 compatible PBX and key systems. The IP multiplexers connect
to the CPE devices with TDM and to the Ethernet switches with 100Base-T,
sharing the bandwidth between sites with data. The multiplexers shown use T1
on both ends, but they could just as easily use analog trunks, in which case the
connection would typically be FXS/FXO. The TDM frames are encapsulated into
IP packets that are transported over the fast Ethernet ports. The ToS bits of the
packets are set to classify the packets as high priority.
TDMoIP provides a circuit-emulation service, also called pseudowire, that
is transparent to protocols and signaling. The multiplexer repeats the contents
of each channel to the other end. The IETF PWE3 Working Group is working
on protocol standards that are in draft form as this book is published, so products
are likely to be proprietary. Typically, the payload of each channel connects to a
48-octet ATM cell, which does not have the 5-octet header. These are encapsulated
into IP frames in some multiple. As the number of TDM octets per frame increases
throughput increases because of lower packet overhead, but the effects of frame
loss are more severe.
Compared to VoIP gateways, TDMoIP multiplexers have lower latency, so
the circuit quality is likely to be better, provided packet loss is not excessive. As
with VoIP gateways, the multiplexers compensate for packet loss by repeating the
contents of the previous packet. The multiplexer transmits channel timeslots
whether they are empty or not, so from this standpoint it is less bandwidthefficient
than VoIP. The impact of this is generally irrelevant, however, because a
TDMoIP multiplexer is not used where bandwidth is restricted. For example, it
would not be usable over the Internet. Some products support fractional T1/E1 so
vacant channels are not transmitted.
VoIP Business Applications
VoIP has four principal categories of applications in the enterprise network:
_ Branch office. Since the branch office is already equipped with a router
or FRAD, adding voice may be a natural outgrowth of the existing
network. For distant branch offices, the existing frame relay network
may be used. For local branches, it may be feasible to replace off-premise
extensions with VoIP.
_ Telecommuting. Similar in architecture to the branch office, telecommuters
may use VoIP to provide a voice and data connection to the main office.
_ Toll bypass. Carrying long distance traffic for next to nothing is attractive,
particularly when the distances are great. The ability to send large
quantities of fax messages, which tend not to be time sensitive, over
the Internet, can be particularly attractive.
_ Web-enabled call center. Callers can browse the company’s Web page
by computer, and then click an icon to talk to a live agent. Costs are
reduced and customer convenience is enhanced.
chapter39
Flexibility
The TDM PBX is a proprietary and inflexible device, closed in every respect except
for its CTI interface, which provides for limited call control. Although IP PBXs
(with the exception of open-source systems) do not open their call-control programs,
they provide more flexible interfaces such as SIP to permit development
of server-based features. They also open options for branch and home offices that
are expensive or difficult with traditional architectures.
Create Value-Added Services
Convergence opens an enormous variety of opportunities to provide new services
that are infeasible with circuit switching. The user interface for telephone service
has improved little over the years. This is not because of a lack of imagination on
how to improve it or a lack of APIs for hooking new applications to proprietary
systems. It is more because the applications must be customized for each type of
CPE system, and no manufacturer has enough market share to generate a mass
market. For example, unified messaging has been available for years, and there is
little variance in the features that various products support, but it has not achieved
enough market penetration to bring the cost down to the point of becoming popular.
Open protocols such as SIP can separate the services from the call control,
which by its nature must be closed. Carriers and third-party developers can create
countless new services and make them operable across a variety of platforms.
Enriched User Experience
Ultimately, new services will change the way people work and communicate.
Personal communication assistants can enable critical employees to be contacted
while still screening unwanted calls without the need for human assistance.
Functions that are difficult with a standard telephone interface, such as setting
up conference calls and dialing by directory name, will become easier with an
improved user interface. Productivity should improve through remote collaboration
and shared access to documents or whiteboard. Just as the PC is a standard
office tool today, these new applications will become such a way of office life that
users will expect them to be available.
Rapid Deployment of New Applications
New and innovative applications can be deployed more rapidly with IP than
with traditional fixed telephone systems, and the pace of improvement does not
depend on the actions of a single vendor. Furthermore, in a geographically dispersed
organization, new applications can be downloaded onto desktop clients
without involving a generic program upgrade.
Barriers to Convergence
New telecommunications developments have always been over-hyped, and with
notable exceptions such as fiber optics, many of these have flared briefly and fizzled
out. After an initial period of exuberance, convergence is proceeding slowly for a
variety of reasons, not the least of which is the difficulty in demonstrating a suitable
return on the investment. Most of the advantages listed above require the
organization to change and adapt to a new environment, and this often happens
slowly.
The initial impetus was long distance cost saving, but that argument has
largely disappeared now that long distance costs are so low. Savings from managing
the converged network are difficult to prove unless the workforce shrinks
with the new technology. When ROI is calculated, many of the savings are in soft
dollars and demonstrating real cash saving is more difficult. Convergence will
develop in time, but many issues listed in this section remain to be resolved.
Network Infrastructure
Carriers are converting portions of their network to VoIP, but islands of VoIP
cannot support the performance and security that commercial-grade voice
communication demands. For VoIP to be a viable alternative to the PSTN, it must
support carrier interoperability. For services such as worldwide VPNs and telephone
connections to be effective, they must transcend carrier boundaries because
no carrier can fulfill all of the needs without relying on other providers. Carriers
must be able to provide services to any corner of any rural area of any country,
and no carrier has sufficient reach without relying on other service providers.
Carriers therefore must be able to interconnect with appropriate levels of security
and service definition and the sessions must be metered to compensate carriers for
handling transit traffic.
Today, many outsiders expect the Internet to become the backbone for this
multiservice, multinational, multiowner network, but those expectations are unrealistic.
Anetwork converged over IP does not mean the public Internet as it is now
structured. Obviously, it can carry voice. It does it every day, but it cannot carry
voice with the consistent quality that the world has enjoyed since the conversion
to an all-digital network. The alternatives are developing an overlay network that
has the stability isochronous applications need, or hardening the Internet. The
latter means changing the basic design concepts that keep the Internet cheap
and fast-paced.
Peering points on the Internet today do not meet any specified performance
criteria and there is no incentive to support a guaranteed level of service. The
converged network must provide an appropriate level of assured delivery in
response to requests from the application. This is inconsistent with the intent of
the Internet, which is to deliver inexpensive connections that are not sensitive to
usage or distance. Either the Internet must be split to provide a separate network
with reliability and security, or the cost of service must increase.
Flow Control
Another key issue is congestion control, which is a vital feature of any voice or
data network. The difficulty is that voice and data behave differently when it
comes to congestion. Both can throttle traffic back at the source, but the nature of
the traffic flow is much different. Many data applications have peaks of high
bandwidth demand for short intervals, but then demand drops to zero as the user
operates on a downloaded file. If the network is congested, it is apt to be for only
a brief interval, after which traffic begins to flow normally. During the heavy flow
periods, TCP closes its window or routers discard traffic, but the process is transparent
to the users, who see a slow response, but the session continues without
interruption.
Unlike data with its heavy peaks, voice is a relatively even flow of half-duplex
traffic that is predictable. Traffic engineers have mounds of data that enable them
to predict voice loads by hour, day, and season until something unusual happens.
Storms, disasters, significant news events, and other external events usually
inspire an extraordinary number of customers to place telephone calls. These
cause traffic to fall outside the normal range and the network has to protect itself
while prioritizing service to essential customers. Voice networks shed load by a
variety of techniques, the first of which is to delay dial tone. During heavy load
periods the LEC can operate line-load control, but this is done only in extreme
circumstances. Common-control equipment such as DTMF registers are engineered
for normal peaks. In abnormal peaks, the registers may be tied up, so the
caller does not receive dial tone. The caller can remain off the hook and dial tone
will eventually be provided. If the congestion is in the trunking network, calls will
not go outside the serving class 5 switch. The user hears reorder and must redial.
Flow control is a standard feature of TCP, but real-time packets work under
UDP, which does not provide flow control. An IETF working group is working
on Datagram Congestion Control Protocol, which is intended as an alternate
transport protocol. DCCP offers functions that bridge the gap between TCP and
UDP. These include packet acknowledgement, congestion notification and
control, packet sequencing, and protection against denial-of-service attacks. This
protocol may resolve flow control issues.
Security
This is the issue that is the most difficult to resolve on the Internet, while still
keeping the service manageable. While users must get involved with encryption,
tunneling, firewalls, and similar security provisions, VoIP will be confined to a
narrow spectrum of users that are willing to put up with the complexity in
exchange for the benefits. A major strength of the PSTN is the fact that uninvited
guests cannot ride the coattails of a file or message, latch onto the telephone, and
infect it with a virus. While trust is not an issue with the PSTN, eternal vigilance
is required on the Internet to thwart a coterie of miscreants who, for whatever
malevolent motivation are attempting to inflict damage.
Broadband Penetration
For VoIP to be successful, broadband must become nearly as ubiquitous as the
PSTN. In 2004, the latest figures available as this book goes to press, broadband
penetration in the U.S. is reported to be 42.5 percent of households with Internet
access. About three-fourths of households have Internet access, which means
broadband penetration is roughly one-third of the households. Other countries,
notably Sweden, Japan, and South Korea, have penetration in the order of threefourths
of all households, significantly greater bandwidth, and at a cost that is
more affordable than in North America. This issue must be resolved before the
converged network can claim success.
Service Complexity
For VoIP to become universally accepted, it must be simplified. Today, customers
need to know too much about VoIP to make it work. The strength of the PSTN is
its simplicity. Customers do not need to know anything about addresses, E-911
access, NAT, and other such technical issues to make the telephone work. For VoIP
to be ready for real time, users must be able to plug the service into a wall jack and
have it work without the need to configure firewalls, install VoIP terminal
adapters, or worry about IP addresses. Users must be able to connect to other VoIP
users without using the PSTN as an intermediary and without the need to know
the identity of the callee’s service provider.
The TDM PBX is a proprietary and inflexible device, closed in every respect except
for its CTI interface, which provides for limited call control. Although IP PBXs
(with the exception of open-source systems) do not open their call-control programs,
they provide more flexible interfaces such as SIP to permit development
of server-based features. They also open options for branch and home offices that
are expensive or difficult with traditional architectures.
Create Value-Added Services
Convergence opens an enormous variety of opportunities to provide new services
that are infeasible with circuit switching. The user interface for telephone service
has improved little over the years. This is not because of a lack of imagination on
how to improve it or a lack of APIs for hooking new applications to proprietary
systems. It is more because the applications must be customized for each type of
CPE system, and no manufacturer has enough market share to generate a mass
market. For example, unified messaging has been available for years, and there is
little variance in the features that various products support, but it has not achieved
enough market penetration to bring the cost down to the point of becoming popular.
Open protocols such as SIP can separate the services from the call control,
which by its nature must be closed. Carriers and third-party developers can create
countless new services and make them operable across a variety of platforms.
Enriched User Experience
Ultimately, new services will change the way people work and communicate.
Personal communication assistants can enable critical employees to be contacted
while still screening unwanted calls without the need for human assistance.
Functions that are difficult with a standard telephone interface, such as setting
up conference calls and dialing by directory name, will become easier with an
improved user interface. Productivity should improve through remote collaboration
and shared access to documents or whiteboard. Just as the PC is a standard
office tool today, these new applications will become such a way of office life that
users will expect them to be available.
Rapid Deployment of New Applications
New and innovative applications can be deployed more rapidly with IP than
with traditional fixed telephone systems, and the pace of improvement does not
depend on the actions of a single vendor. Furthermore, in a geographically dispersed
organization, new applications can be downloaded onto desktop clients
without involving a generic program upgrade.
Barriers to Convergence
New telecommunications developments have always been over-hyped, and with
notable exceptions such as fiber optics, many of these have flared briefly and fizzled
out. After an initial period of exuberance, convergence is proceeding slowly for a
variety of reasons, not the least of which is the difficulty in demonstrating a suitable
return on the investment. Most of the advantages listed above require the
organization to change and adapt to a new environment, and this often happens
slowly.
The initial impetus was long distance cost saving, but that argument has
largely disappeared now that long distance costs are so low. Savings from managing
the converged network are difficult to prove unless the workforce shrinks
with the new technology. When ROI is calculated, many of the savings are in soft
dollars and demonstrating real cash saving is more difficult. Convergence will
develop in time, but many issues listed in this section remain to be resolved.
Network Infrastructure
Carriers are converting portions of their network to VoIP, but islands of VoIP
cannot support the performance and security that commercial-grade voice
communication demands. For VoIP to be a viable alternative to the PSTN, it must
support carrier interoperability. For services such as worldwide VPNs and telephone
connections to be effective, they must transcend carrier boundaries because
no carrier can fulfill all of the needs without relying on other providers. Carriers
must be able to provide services to any corner of any rural area of any country,
and no carrier has sufficient reach without relying on other service providers.
Carriers therefore must be able to interconnect with appropriate levels of security
and service definition and the sessions must be metered to compensate carriers for
handling transit traffic.
Today, many outsiders expect the Internet to become the backbone for this
multiservice, multinational, multiowner network, but those expectations are unrealistic.
Anetwork converged over IP does not mean the public Internet as it is now
structured. Obviously, it can carry voice. It does it every day, but it cannot carry
voice with the consistent quality that the world has enjoyed since the conversion
to an all-digital network. The alternatives are developing an overlay network that
has the stability isochronous applications need, or hardening the Internet. The
latter means changing the basic design concepts that keep the Internet cheap
and fast-paced.
Peering points on the Internet today do not meet any specified performance
criteria and there is no incentive to support a guaranteed level of service. The
converged network must provide an appropriate level of assured delivery in
response to requests from the application. This is inconsistent with the intent of
the Internet, which is to deliver inexpensive connections that are not sensitive to
usage or distance. Either the Internet must be split to provide a separate network
with reliability and security, or the cost of service must increase.
Flow Control
Another key issue is congestion control, which is a vital feature of any voice or
data network. The difficulty is that voice and data behave differently when it
comes to congestion. Both can throttle traffic back at the source, but the nature of
the traffic flow is much different. Many data applications have peaks of high
bandwidth demand for short intervals, but then demand drops to zero as the user
operates on a downloaded file. If the network is congested, it is apt to be for only
a brief interval, after which traffic begins to flow normally. During the heavy flow
periods, TCP closes its window or routers discard traffic, but the process is transparent
to the users, who see a slow response, but the session continues without
interruption.
Unlike data with its heavy peaks, voice is a relatively even flow of half-duplex
traffic that is predictable. Traffic engineers have mounds of data that enable them
to predict voice loads by hour, day, and season until something unusual happens.
Storms, disasters, significant news events, and other external events usually
inspire an extraordinary number of customers to place telephone calls. These
cause traffic to fall outside the normal range and the network has to protect itself
while prioritizing service to essential customers. Voice networks shed load by a
variety of techniques, the first of which is to delay dial tone. During heavy load
periods the LEC can operate line-load control, but this is done only in extreme
circumstances. Common-control equipment such as DTMF registers are engineered
for normal peaks. In abnormal peaks, the registers may be tied up, so the
caller does not receive dial tone. The caller can remain off the hook and dial tone
will eventually be provided. If the congestion is in the trunking network, calls will
not go outside the serving class 5 switch. The user hears reorder and must redial.
Flow control is a standard feature of TCP, but real-time packets work under
UDP, which does not provide flow control. An IETF working group is working
on Datagram Congestion Control Protocol, which is intended as an alternate
transport protocol. DCCP offers functions that bridge the gap between TCP and
UDP. These include packet acknowledgement, congestion notification and
control, packet sequencing, and protection against denial-of-service attacks. This
protocol may resolve flow control issues.
Security
This is the issue that is the most difficult to resolve on the Internet, while still
keeping the service manageable. While users must get involved with encryption,
tunneling, firewalls, and similar security provisions, VoIP will be confined to a
narrow spectrum of users that are willing to put up with the complexity in
exchange for the benefits. A major strength of the PSTN is the fact that uninvited
guests cannot ride the coattails of a file or message, latch onto the telephone, and
infect it with a virus. While trust is not an issue with the PSTN, eternal vigilance
is required on the Internet to thwart a coterie of miscreants who, for whatever
malevolent motivation are attempting to inflict damage.
Broadband Penetration
For VoIP to be successful, broadband must become nearly as ubiquitous as the
PSTN. In 2004, the latest figures available as this book goes to press, broadband
penetration in the U.S. is reported to be 42.5 percent of households with Internet
access. About three-fourths of households have Internet access, which means
broadband penetration is roughly one-third of the households. Other countries,
notably Sweden, Japan, and South Korea, have penetration in the order of threefourths
of all households, significantly greater bandwidth, and at a cost that is
more affordable than in North America. This issue must be resolved before the
converged network can claim success.
Service Complexity
For VoIP to become universally accepted, it must be simplified. Today, customers
need to know too much about VoIP to make it work. The strength of the PSTN is
its simplicity. Customers do not need to know anything about addresses, E-911
access, NAT, and other such technical issues to make the telephone work. For VoIP
to be ready for real time, users must be able to plug the service into a wall jack and
have it work without the need to configure firewalls, install VoIP terminal
adapters, or worry about IP addresses. Users must be able to connect to other VoIP
users without using the PSTN as an intermediary and without the need to know
the identity of the callee’s service provider.
WHY CONVERGENCE?
Above the physical layer, the Internet is diametrically opposite the PSTN in most
ways, so why is the push on for convergence? The answer lies in a complex brew
of economics, politics, culture, and technology. The term convergence holds
different meanings for many people, so let us begin this section with a definition.
Convergence Merging real-time applications such as voice, video, and instant
messaging together with data onto a single broadband infrastructure that is
based on IP.
The reasons for the trend toward convergence are many, not the least of
which is an urgent need of manufacturers to find innovative new products to help
them emerge from the technology slump that began in 2002. This is coupled with
the zeal of the international Internet community that is persistently pursuing
a vision of a unified network. Companies that see expanded opportunities for
innovation in such online applications as gaming, music, television, education,
and countless services that have scarcely been glimpsed support this vision.
In 2004 NTT DoCoMo introduced services in Tokyo that give us a preview
of the kinds of applications that converged services make possible. Dubbed
“Felicia,” the service marries a 3G handset, a smartcard, and GPS to enable users
to perform a variety of functions that are otherwise impractical or require separate
devices. The smartcard, coupled to the target by wireless, is loaded with identification
and financial information that enables users to make purchases from vending
machines, payments at restaurants, and to board trains and buses without
tickets. Auser standing at a bus stop wondering when the next bus will arrive can
link to the transit company to obtain the answer. The mobile handset provides its
location from its GPS and the application compares it to the bus’ location, also
obtained from GPS. Users can purchase tickets to events, locate restaurants and
shops in their locale, and link to RFID devices in the store to download more
information about the products. Devices such as these will change the way we
work and live, and the functions are impractical without a converged network.
The Business Case for Convergence
Convergence has a lot of appeal for the enterprise. If its promise is realized,
operational costs will drop and better utilization will be made of an expensive
resource. An all-IP network will allow service providers to create new valueadded
services. Much of the time-consuming provisioning process will become
unnecessary as the network evolves to respond dynamically to changes in demand.
Business models will change and new ones will emerge as inexpensive and ubiquitous
bandwidth becomes available. In some countries with extensive broadband
access, for example, the market for downloaded music exceeds that for CDs.
Video rentals will fade in favor of video-on-demand, and presence engines that
advertise personal preferences for reachability will drive personal communications.
Barriers to entry of local phone markets will disappear as consumers move
their telephone service to IP networks, and the phone will be a multimedia device
more or less permanently associated with an individual, possibly with the address
issued at birth. Let us look in more detail at these benefits.
Competitive Advantage
For some businesses, convergence is necessary to support business goals including
reduced cost of doing business, increased productivity, and improved customer
service. Aconverged network supports collaboration with customers, which results
in improved customer retention. Businesses can tie customers to their internal
processes through e-commerce. Multimedia contact centers can enable organizations
to create an on-line experience that approximates face-to-face service delivery,
enabling the customer to choose the preferred method of communication and shift
to another mode in the middle of a session.
At first the converged network will catch on slowly with the early adopters
gaining the most advantage, but gradually it will change the business model to
the point that customers will expect the service. Today in Japan’s Felicia service,
for example, a customer can surf the Web from her 3G phone, inspect a photo of a
product, order it, and pay for it, all from the same hand-held instrument.
Lower Cost of Ownership
The cost equation is difficult to factor into the decision to migrate to a converged
network. To realize cost reductions, cultural changes and dedicated management
effort are required. The following are some of the ways ownership costs can be
reduced:
_ With only one unified network infrastructure to design, manage, and
support, labor costs should be reduced. The same staff that support
servers, switches, routers, and LAN equipment can maintain the voice
switching system, and reduce or eliminate the need for outsourced
expertise.
_ Users can do many of their own moves, adds, and changes without
involving the IT staff.
_ A converged network enables enterprises to make more effective use of
access bandwidth, while simplifying administration and maintenance.
_ Host-based services can eliminate the need to invest in a dedicated PBX
or upgrade an obsolescent system, while still providing the productivity
enhancements of feature telephones.
_ Call processing can be centralized for multiple independent sites,
eliminating the need for separate key systems, resulting in lower branch
office costs.
_ The quantity of station wiring drops can be reduced by combining both
PCs and VoIP on the same Ethernet port.
_ IP PBXs are more scalable than TDM PBXs, which have a maximum size
and grow to that size by adding port cards and cabinets.
Support a Mobile and Geographically Independent Workforce
This advantage is important for telecommuters, sales people, and others who
spend a great deal of time away from the office and to people who work from
home at least part of the time. It is particularly difficult to provide conventional
voice communications that are integrated with the office telephone system to such
people. With VoIP the benefits of the PBX can be extended outside the bounds
of the office. The need for office space is reduced and the organization is able
to comply with government regulations that require employers to reduce traffic
congestion. Part-time people can work from home to supplement a fixed call
center workforce
ways, so why is the push on for convergence? The answer lies in a complex brew
of economics, politics, culture, and technology. The term convergence holds
different meanings for many people, so let us begin this section with a definition.
Convergence Merging real-time applications such as voice, video, and instant
messaging together with data onto a single broadband infrastructure that is
based on IP.
The reasons for the trend toward convergence are many, not the least of
which is an urgent need of manufacturers to find innovative new products to help
them emerge from the technology slump that began in 2002. This is coupled with
the zeal of the international Internet community that is persistently pursuing
a vision of a unified network. Companies that see expanded opportunities for
innovation in such online applications as gaming, music, television, education,
and countless services that have scarcely been glimpsed support this vision.
In 2004 NTT DoCoMo introduced services in Tokyo that give us a preview
of the kinds of applications that converged services make possible. Dubbed
“Felicia,” the service marries a 3G handset, a smartcard, and GPS to enable users
to perform a variety of functions that are otherwise impractical or require separate
devices. The smartcard, coupled to the target by wireless, is loaded with identification
and financial information that enables users to make purchases from vending
machines, payments at restaurants, and to board trains and buses without
tickets. Auser standing at a bus stop wondering when the next bus will arrive can
link to the transit company to obtain the answer. The mobile handset provides its
location from its GPS and the application compares it to the bus’ location, also
obtained from GPS. Users can purchase tickets to events, locate restaurants and
shops in their locale, and link to RFID devices in the store to download more
information about the products. Devices such as these will change the way we
work and live, and the functions are impractical without a converged network.
The Business Case for Convergence
Convergence has a lot of appeal for the enterprise. If its promise is realized,
operational costs will drop and better utilization will be made of an expensive
resource. An all-IP network will allow service providers to create new valueadded
services. Much of the time-consuming provisioning process will become
unnecessary as the network evolves to respond dynamically to changes in demand.
Business models will change and new ones will emerge as inexpensive and ubiquitous
bandwidth becomes available. In some countries with extensive broadband
access, for example, the market for downloaded music exceeds that for CDs.
Video rentals will fade in favor of video-on-demand, and presence engines that
advertise personal preferences for reachability will drive personal communications.
Barriers to entry of local phone markets will disappear as consumers move
their telephone service to IP networks, and the phone will be a multimedia device
more or less permanently associated with an individual, possibly with the address
issued at birth. Let us look in more detail at these benefits.
Competitive Advantage
For some businesses, convergence is necessary to support business goals including
reduced cost of doing business, increased productivity, and improved customer
service. Aconverged network supports collaboration with customers, which results
in improved customer retention. Businesses can tie customers to their internal
processes through e-commerce. Multimedia contact centers can enable organizations
to create an on-line experience that approximates face-to-face service delivery,
enabling the customer to choose the preferred method of communication and shift
to another mode in the middle of a session.
At first the converged network will catch on slowly with the early adopters
gaining the most advantage, but gradually it will change the business model to
the point that customers will expect the service. Today in Japan’s Felicia service,
for example, a customer can surf the Web from her 3G phone, inspect a photo of a
product, order it, and pay for it, all from the same hand-held instrument.
Lower Cost of Ownership
The cost equation is difficult to factor into the decision to migrate to a converged
network. To realize cost reductions, cultural changes and dedicated management
effort are required. The following are some of the ways ownership costs can be
reduced:
_ With only one unified network infrastructure to design, manage, and
support, labor costs should be reduced. The same staff that support
servers, switches, routers, and LAN equipment can maintain the voice
switching system, and reduce or eliminate the need for outsourced
expertise.
_ Users can do many of their own moves, adds, and changes without
involving the IT staff.
_ A converged network enables enterprises to make more effective use of
access bandwidth, while simplifying administration and maintenance.
_ Host-based services can eliminate the need to invest in a dedicated PBX
or upgrade an obsolescent system, while still providing the productivity
enhancements of feature telephones.
_ Call processing can be centralized for multiple independent sites,
eliminating the need for separate key systems, resulting in lower branch
office costs.
_ The quantity of station wiring drops can be reduced by combining both
PCs and VoIP on the same Ethernet port.
_ IP PBXs are more scalable than TDM PBXs, which have a maximum size
and grow to that size by adding port cards and cabinets.
Support a Mobile and Geographically Independent Workforce
This advantage is important for telecommuters, sales people, and others who
spend a great deal of time away from the office and to people who work from
home at least part of the time. It is particularly difficult to provide conventional
voice communications that are integrated with the office telephone system to such
people. With VoIP the benefits of the PBX can be extended outside the bounds
of the office. The need for office space is reduced and the organization is able
to comply with government regulations that require employers to reduce traffic
congestion. Part-time people can work from home to supplement a fixed call
center workforce
Chapter 39 Telecommunications Convergence
If the telecommunication experts are unanimous on anything, it is the conviction
that all forms of communication are gravitating toward a single unified IP network.
There is less unanimity about when this transformation will be complete,
but few expect it to happen overnight. Both customers and carriers have too much
invested in the PSTN to abandon it, and while the industry has solutions for most
of the technical obstacles, the technologies are immature and changing steadily.
Furthermore, the economics of convergence are insufficient to justify the capital
expenditures of anything but an evolutionary transition.
Historically, enhanced features have started in private networks and gravitated
to the public. While the world lingered over ISDN standards, for example, PBX
manufacturers brought its features to the office. The telephones and protocols are
proprietary, but the features are proven and available in every product line. The
same is happening with the transition to IP. International Data Corporation (IDC)
projects that about 1.4 million IP PBXs will be in service by 2008, but circuit-switched
PBX lines will still outnumber IP lines by a factor of 3 to 1. Convergence faces fewer
obstacles in the private arena because enterprise systems have shorter service lives
than public and the customer has better control of the infrastructure.
Convergence in the public network, where switches typically have service
lives of 20 years or more, is a different matter. The PSTN core is moving toward
IP, but class 5 switches remain circuit-switched. ILECs are beginning to offer VoIP
service over softswitches, but the motivation is more to meet competition than
because of any technical advantage of IP. For what it does, the PSTN is good
enough. It meets the needs of most residence and small business subscribers, and
for voice alone it is superior to IP. Several factors are standing in the way of moving
public voice and video to IP. The most important of these is the lack of a suitable
infrastructure. The IP equivalent of the PSTN is the Internet, but it is not suitable
for commercial-grade telephone service. The Internet is designed to be cheap
and ubiquitous, but it is a chaotic model, not because of the behavior of the
network, but because of the nature of the service. You can have low cost or QoS,
but you cannot have both.
Some observers suggest that the QoS issue is overblown, pointing to cell
phones as evidence that people are willing to forego service quality in exchange
for convenience. This observation is accurate as far as it goes, but it ignores two
issues. One is that business-class telephone service demands consistent quality
that the Internet is incapable of providing. VoIP over the Internet can supplement
the PSTN, but it is not a suitable business model to replace it. The second issue is
the fact that every telephone session has two parties. Some users may decide to
forego quality in favor of low-cost Internet connections, which may be suitable for
family and friends, but enterprises cannot afford to impose poor quality on their
customers. The telecommunications industry has elevated transmission quality to
its present state through a long series of technological advances. To diminish it for
the sake of expediency would be a serious mistake.
That said, the industry is evolving toward the converged network. It will be
a gradual transition; one that will occur only after some significant barriers have
been overcome. The obstacles will be resolved in time with more new protocols
and a lot of industry work, particularly on the infrastructure. The Internet could
be adapted for commercial grade telephone service, but several things would
have to change. ISPs, which are competitive and independent, would have to
agree to adhere to service and quality standards. Open interconnection with their
competitors would need to occur and today’s pricing model would have to change.
The Internet is designed and constructed to offer best-effort service at low cost
and that is inconsistent with the needs of time-sensitive applications.
Service quality notwithstanding, a growing amount of voice traffic is moving
to the Internet as service providers such as Skype, Vonage, and AT&T’s CallVantage
offer subscribers service at prices so low that millions are signing up. So what if the
service is not toll quality. If people can call halfway around the world for nothing
over a broadband connection that they have already paid for, some sacrifice in
reliability is a small price to pay. Cable companies with their broadband access
into most residences and many small businesses are in a particularly advantageous
position to offer VoIP service because they have control of the access channel.
VoIP in the public network is a disruptive technology. Although its impact
has been slight so far, it forces the LECs to review their business models carefully.
When a company can come from nowhere and furnish telephone service over a
broadband connection with minimal investment, it causes concern for telephone
companies and regulators alike. Regulators’ traditional modes of taxation and
control are impossible with Internet telephony, at least at this point. In 2004 the
State of Minnesota attempted to impose telephone regulation on Vonage despite
the fact that the PUC’s control does not extend to interstate calls and it is impractical
to segregate VoIP calls. This prompted the FCC to issue an order preempting the
Minnesota order, but the issue has not been put to rest.
In this chapter we look first at the forces that are driving toward a converged
network. We look at services that countries such as Japan, which has a much
higher degree of broadband development than North America, are beginning to
provide and how these motivate the development of the converged network. We
examine the advantages of convergence and the barriers that are impeding its
development. We look briefly at the Infranet, which is an initiative of several
companies to develop an IP-based alternative to the Internet. The Infranet, if it is
successful, may prove to be the enabling factor that removes most of the barriers.
We conclude the chapter with an Applications section that discusses the ways in
which the converged network is available today and considerations in applying it.
that all forms of communication are gravitating toward a single unified IP network.
There is less unanimity about when this transformation will be complete,
but few expect it to happen overnight. Both customers and carriers have too much
invested in the PSTN to abandon it, and while the industry has solutions for most
of the technical obstacles, the technologies are immature and changing steadily.
Furthermore, the economics of convergence are insufficient to justify the capital
expenditures of anything but an evolutionary transition.
Historically, enhanced features have started in private networks and gravitated
to the public. While the world lingered over ISDN standards, for example, PBX
manufacturers brought its features to the office. The telephones and protocols are
proprietary, but the features are proven and available in every product line. The
same is happening with the transition to IP. International Data Corporation (IDC)
projects that about 1.4 million IP PBXs will be in service by 2008, but circuit-switched
PBX lines will still outnumber IP lines by a factor of 3 to 1. Convergence faces fewer
obstacles in the private arena because enterprise systems have shorter service lives
than public and the customer has better control of the infrastructure.
Convergence in the public network, where switches typically have service
lives of 20 years or more, is a different matter. The PSTN core is moving toward
IP, but class 5 switches remain circuit-switched. ILECs are beginning to offer VoIP
service over softswitches, but the motivation is more to meet competition than
because of any technical advantage of IP. For what it does, the PSTN is good
enough. It meets the needs of most residence and small business subscribers, and
for voice alone it is superior to IP. Several factors are standing in the way of moving
public voice and video to IP. The most important of these is the lack of a suitable
infrastructure. The IP equivalent of the PSTN is the Internet, but it is not suitable
for commercial-grade telephone service. The Internet is designed to be cheap
and ubiquitous, but it is a chaotic model, not because of the behavior of the
network, but because of the nature of the service. You can have low cost or QoS,
but you cannot have both.
Some observers suggest that the QoS issue is overblown, pointing to cell
phones as evidence that people are willing to forego service quality in exchange
for convenience. This observation is accurate as far as it goes, but it ignores two
issues. One is that business-class telephone service demands consistent quality
that the Internet is incapable of providing. VoIP over the Internet can supplement
the PSTN, but it is not a suitable business model to replace it. The second issue is
the fact that every telephone session has two parties. Some users may decide to
forego quality in favor of low-cost Internet connections, which may be suitable for
family and friends, but enterprises cannot afford to impose poor quality on their
customers. The telecommunications industry has elevated transmission quality to
its present state through a long series of technological advances. To diminish it for
the sake of expediency would be a serious mistake.
That said, the industry is evolving toward the converged network. It will be
a gradual transition; one that will occur only after some significant barriers have
been overcome. The obstacles will be resolved in time with more new protocols
and a lot of industry work, particularly on the infrastructure. The Internet could
be adapted for commercial grade telephone service, but several things would
have to change. ISPs, which are competitive and independent, would have to
agree to adhere to service and quality standards. Open interconnection with their
competitors would need to occur and today’s pricing model would have to change.
The Internet is designed and constructed to offer best-effort service at low cost
and that is inconsistent with the needs of time-sensitive applications.
Service quality notwithstanding, a growing amount of voice traffic is moving
to the Internet as service providers such as Skype, Vonage, and AT&T’s CallVantage
offer subscribers service at prices so low that millions are signing up. So what if the
service is not toll quality. If people can call halfway around the world for nothing
over a broadband connection that they have already paid for, some sacrifice in
reliability is a small price to pay. Cable companies with their broadband access
into most residences and many small businesses are in a particularly advantageous
position to offer VoIP service because they have control of the access channel.
VoIP in the public network is a disruptive technology. Although its impact
has been slight so far, it forces the LECs to review their business models carefully.
When a company can come from nowhere and furnish telephone service over a
broadband connection with minimal investment, it causes concern for telephone
companies and regulators alike. Regulators’ traditional modes of taxation and
control are impossible with Internet telephony, at least at this point. In 2004 the
State of Minnesota attempted to impose telephone regulation on Vonage despite
the fact that the PUC’s control does not extend to interstate calls and it is impractical
to segregate VoIP calls. This prompted the FCC to issue an order preempting the
Minnesota order, but the issue has not been put to rest.
In this chapter we look first at the forces that are driving toward a converged
network. We look at services that countries such as Japan, which has a much
higher degree of broadband development than North America, are beginning to
provide and how these motivate the development of the converged network. We
examine the advantages of convergence and the barriers that are impeding its
development. We look briefly at the Infranet, which is an initiative of several
companies to develop an IP-based alternative to the Internet. The Infranet, if it is
successful, may prove to be the enabling factor that removes most of the barriers.
We conclude the chapter with an Applications section that discusses the ways in
which the converged network is available today and considerations in applying it.
chapter24
CALL ACCOUNTING SYSTEMS
All PBXs, most hybrids, and many key telephone systems include a CDR port that
receives call details at the conclusion of each call. The call details can be printed
or passed to a call accounting system for further processing. The CDR output of
most systems is of little value by itself because calls are presented in order of completion
and lack rates, identification of the called number, and other such details
needed for control of long distance costs. Call accounting systems add details to
create management reports, a complete long distance statement for each user, and
departmental summaries. The primary purposes of a call accounting system are to
discourage unauthorized use and to distribute costs to users. They also have other
uses in some companies. For example, a supervisor may use the CDR record to
check the effectiveness of an employee’s outgoing sales calls.
Most call accounting systems on the market are software programs for PCs.
CDR data either feeds directly into an on-line PC or it feeds into a buffer that
stores call details until it is polled. A buffer makes it unnecessary to tie up a PC in
collecting call details. If the power fails, the battery backup in the buffer retains
the stored information.
In multi-PBX environments, a networked call accounting system may be
required. These systems use buffers or computers to collect information at remote
sites and upload it to a central processor at the end of the collection interval.
If long distance calls can be placed from one PBX over trunks attached to another,
a tie line reconciliation program may be needed. The tie line reconciliation
program uses the completion time of calls to match calls that originate on one
PBX and terminate on trunks connected to another. Networked PBXs send originating
station identification over the signaling channel to a remote PBX. If the
remote PBX is equipped to extract the calling station identification from the
network and associate it with the CDR output, the need for tie line reconciliation
is eliminated.
Most PBXs can output to the CDR port any combination of long distance,
local, outgoing, and incoming calls. The amount of detail to collect is a matter of
individual judgment, but sufficient buffer and disk storage space must be provided
to hold all the information collected.
CALL ACCOUNTING APPLICATION ISSUES
Application information for PBXs and key systems is included in the next two
chapters. This section covers application information for call accounting systems.
Call Accounting Evaluation
Most PBXs today are purchased with a call accounting system that is normally
programmed and supported by a third-party manufacturer. The following are
some criteria for selecting a call accounting system.
Reports
The main reason for buying a call accounting system is for its reports. Evaluate
factors such as these:
_ What kinds of special reports are provided? Do they meet the
organization’s requirements? Examples are unused extensions,
long or short duration calls, unused trunks, and calls to emergency
numbers.
_ Can reports be distributed over the Internet or a company intranet?
_ Can users access their reports with a browser?
_ Are custom-designed reports possible?
_ Is it possible to export report information to an external program, such
as a spreadsheet or database management system, to produce custom
reports?
_ Are traffic reports produced? If so, are they accurate?
_ Are management reports, such as inventories, provided?
_ What kind of manual effort is needed to produce reports? Does it
require a trained operator, or can clerical people perform the month-end
operations with little or no formal training?
_ Is tie line reconciliation required? If so, does the manufacturer support it?
Operational Issues
Most call accounting systems are not completely automatic. The functions
required are downloading the call data from buffers (if they are used), rating calls,
and producing end-of-period reports. The most effective systems provide dragand-
drop capabilities for setting up and scheduling reports and distributing them
to users.
Features
Many call accounting systems provide features that are of extra value. Common
features are toll fraud alerts, telephone directory, and equipment inventory. Some
high-end systems offer telemanagement packages, which typically include service
orders, repair, and inventory in addition to directory and call accounting.
Vendor Support
As with most software packages, vendor support is important for installing and
maintaining the system. Evaluate the vendor’s experience in supporting the package.
Determine whether the vendor has people who have been specifically
trained. Evaluate the amount of support the package developer has available and
what it costs. Some vendors sell ongoing support packages, and where these are
available, the cost-effectiveness should be evaluated.
Call Rating
Most call accounting packages have call-rating tables based on V&H (vertical and
horizontal) tables. These divide the United States and Canada into a grid from
which point-to-point mileage is calculated. Tables must be updated regularly as
rates change. Also, consider that many companies do not need absolute rate accuracy.
To distribute costs among organizational units, precision is usually not
required. Many long distance rate plans use rates that are not distance sensitive,
so V&H rating accuracy is not required. The rating tables identify the called location,
so if rating tables are not used, the called city and state will not be printed on
toll statements unless the vendor offers an abbreviated table. Determine facts such
as these:
_ What kind of rating tables does the manufacturer support?
_ How frequently are tables updated?
_ What do updates cost?
_ What IXCs’ rates does the package support?
_ How are intrastate rates calculated?
_ Do you need to bill back to user departments with high accuracy?
Capacity
Call storage equipment is intended to maintain information on a certain number
of calls. When buffer storage is full, it must be unloaded and calls processed.
Usually, the system must store at least 1 month’s worth of calls. Evaluate questions
such as these:
_ How much storage space is required?
_ What is the capacity in number of calls, both incoming and outgoing?
_ How much growth capacity is provided?
_ Is storage nonvolatile, so if power fails calls are not
All PBXs, most hybrids, and many key telephone systems include a CDR port that
receives call details at the conclusion of each call. The call details can be printed
or passed to a call accounting system for further processing. The CDR output of
most systems is of little value by itself because calls are presented in order of completion
and lack rates, identification of the called number, and other such details
needed for control of long distance costs. Call accounting systems add details to
create management reports, a complete long distance statement for each user, and
departmental summaries. The primary purposes of a call accounting system are to
discourage unauthorized use and to distribute costs to users. They also have other
uses in some companies. For example, a supervisor may use the CDR record to
check the effectiveness of an employee’s outgoing sales calls.
Most call accounting systems on the market are software programs for PCs.
CDR data either feeds directly into an on-line PC or it feeds into a buffer that
stores call details until it is polled. A buffer makes it unnecessary to tie up a PC in
collecting call details. If the power fails, the battery backup in the buffer retains
the stored information.
In multi-PBX environments, a networked call accounting system may be
required. These systems use buffers or computers to collect information at remote
sites and upload it to a central processor at the end of the collection interval.
If long distance calls can be placed from one PBX over trunks attached to another,
a tie line reconciliation program may be needed. The tie line reconciliation
program uses the completion time of calls to match calls that originate on one
PBX and terminate on trunks connected to another. Networked PBXs send originating
station identification over the signaling channel to a remote PBX. If the
remote PBX is equipped to extract the calling station identification from the
network and associate it with the CDR output, the need for tie line reconciliation
is eliminated.
Most PBXs can output to the CDR port any combination of long distance,
local, outgoing, and incoming calls. The amount of detail to collect is a matter of
individual judgment, but sufficient buffer and disk storage space must be provided
to hold all the information collected.
CALL ACCOUNTING APPLICATION ISSUES
Application information for PBXs and key systems is included in the next two
chapters. This section covers application information for call accounting systems.
Call Accounting Evaluation
Most PBXs today are purchased with a call accounting system that is normally
programmed and supported by a third-party manufacturer. The following are
some criteria for selecting a call accounting system.
Reports
The main reason for buying a call accounting system is for its reports. Evaluate
factors such as these:
_ What kinds of special reports are provided? Do they meet the
organization’s requirements? Examples are unused extensions,
long or short duration calls, unused trunks, and calls to emergency
numbers.
_ Can reports be distributed over the Internet or a company intranet?
_ Can users access their reports with a browser?
_ Are custom-designed reports possible?
_ Is it possible to export report information to an external program, such
as a spreadsheet or database management system, to produce custom
reports?
_ Are traffic reports produced? If so, are they accurate?
_ Are management reports, such as inventories, provided?
_ What kind of manual effort is needed to produce reports? Does it
require a trained operator, or can clerical people perform the month-end
operations with little or no formal training?
_ Is tie line reconciliation required? If so, does the manufacturer support it?
Operational Issues
Most call accounting systems are not completely automatic. The functions
required are downloading the call data from buffers (if they are used), rating calls,
and producing end-of-period reports. The most effective systems provide dragand-
drop capabilities for setting up and scheduling reports and distributing them
to users.
Features
Many call accounting systems provide features that are of extra value. Common
features are toll fraud alerts, telephone directory, and equipment inventory. Some
high-end systems offer telemanagement packages, which typically include service
orders, repair, and inventory in addition to directory and call accounting.
Vendor Support
As with most software packages, vendor support is important for installing and
maintaining the system. Evaluate the vendor’s experience in supporting the package.
Determine whether the vendor has people who have been specifically
trained. Evaluate the amount of support the package developer has available and
what it costs. Some vendors sell ongoing support packages, and where these are
available, the cost-effectiveness should be evaluated.
Call Rating
Most call accounting packages have call-rating tables based on V&H (vertical and
horizontal) tables. These divide the United States and Canada into a grid from
which point-to-point mileage is calculated. Tables must be updated regularly as
rates change. Also, consider that many companies do not need absolute rate accuracy.
To distribute costs among organizational units, precision is usually not
required. Many long distance rate plans use rates that are not distance sensitive,
so V&H rating accuracy is not required. The rating tables identify the called location,
so if rating tables are not used, the called city and state will not be printed on
toll statements unless the vendor offers an abbreviated table. Determine facts such
as these:
_ What kind of rating tables does the manufacturer support?
_ How frequently are tables updated?
_ What do updates cost?
_ What IXCs’ rates does the package support?
_ How are intrastate rates calculated?
_ Do you need to bill back to user departments with high accuracy?
Capacity
Call storage equipment is intended to maintain information on a certain number
of calls. When buffer storage is full, it must be unloaded and calls processed.
Usually, the system must store at least 1 month’s worth of calls. Evaluate questions
such as these:
_ How much storage space is required?
_ What is the capacity in number of calls, both incoming and outgoing?
_ How much growth capacity is provided?
_ Is storage nonvolatile, so if power fails calls are not
chapter24
Unified Messaging
This feature, which is discussed in more detail in Chapter 29, integrates the PBX
with voice mail, fax, and e-mail so that messages can be viewed and handled on
a PC screen. The feature may also enable users to translate messages from one format
to another. For example, e-mail messages may be read in synthesized voice if
the user is calling from a telephone and wants them read out. Eventually, with
improvement in speech-to-text software, voice-mail messages will be converted to
e-mail or fax. Currently, speech to text enables users to speak limited commands
to read, forward, and delete voice mail messages from a telephone.
Emergency Service Interface
Most of the developed world has adopted a special dialing code such as 999 or 911
for universal access to emergency services. The basic service enables the PSAP to
hold up the line so it can be traced in case the caller is unable to report the address
of the emergency. Enhanced emergency services contain a database that associates
telephone numbers with street addresses. The street address is often not a fine
enough distinction, however. Users who dial the emergency code from hotels,
apartments with a shared-tenant PBX, campuses, and multibuilding developments
and the like may be difficult or impossible to locate. Therefore, a trend is
toward reporting the station identification to the PSAP so it can be associated with
the room number or building name. The service is known as private system automatic
line identification (PS/ALI). The PRI feature in most PBXs can relay the station
identification to the LEC, which passes it to the PSAP. PBXs lacking PRI can
use a CAMA trunk as an alternative. The need for this feature is particularly acute
in PBXs with remote switch units that can be located in a different PSAP’s jurisdiction
from the host PBX. In this case, a separate trunk group to the local central
office is usually provided. The ARS is programmed so it always seizes a local
trunk when the emergency code is dialed from a station served by the remote.
A related feature that is important in any large PBX is the ability to route
emergency calls simultaneously to the console attendant or a security position.
The law requires that the call route directly to the PSAP, but management wants
to be informed of any such call. It is also a useful feature in hotels, campuses,
schools, hospitals, and other organizations where people may place emergency
calls inadvertently or as a prank.
Multitenant Service
PBXs that provide service to users from different organizations can use multitenant
software to give each organization the appearance of a private switch.
Multitenant service is a software partition in the PBX. Separate attendant consoles
can be provided, and each organization can have its own group of trunks and
block of numbers.
Property Management Interface
Hotels, hospitals, dormitories, and other organizations that resell service often
connect the PBX to a computer to provide features such as checking room status
information, disabling the telephone set from the attendant console, and determining
check-in or check-out status. The PBX provides information to the computer,
and accepts orders from the front desk via computer terminal. The PMI is a
specialized type of computer-telephony integration (see Chapter 27).
Uniform Dialing Plan (UDP)
UDP software in a multi-PBX network enables the caller to dial an extension number
and have the call completed over a tie line network without the caller’s being
concerned about where the extension is located. The PBX selects the route and
takes care of station number translations. UDP software is effective only among
PBXs of the same manufacture although it can work with QSIG-compatible PBXs
if they are so equipped.
Simplified Message Desk Interface (SMDI)
SMDI is a standard way of interfacing a switch to peripheral equipment such as
voice mail. The voice mail connects to the PBX over analog ports or line-side T1
and to the SMDI with a serial connection. For a call going to a messaging unit such
as voice mail, the SMDI link indicates the port the call is using, the type of call,
information about the call such as the source and destination, and the reason the
call is forwarded such as busy or no answer. The SMDI is an open protocol for
interfacing voice mail to the switch as an alternative to the manufacturer’s
proprietary interface.
PBX Voice Features
As all PBXs are designed for voice switching service, they have features intended
for the convenience and productivity of the users. Not all the features listed below
are universally available, and many systems provide features not listed. This list,
in addition to the key system features discussed earlier, briefly describes the most
popular voice features found in PBXs.
_ Automatic call trace: Harassing or nuisance call can be traced to the origin
by dialing an access code.
_ Call blocking: Users can selectively block calls such as specific extensions,
numbers, or calls from particular trunk groups.
_ Call coverage: Users can have one or more coverage paths to direct
how calls route when the called station is busy, does not answer, or is
in do-not-disturb status. External calls can take a different path than
internal calls.
_ Executive override: This feature allows a station to interrupt a busy line
or preempt a long distance trunk.
_ Forced account code: On long distance calls, this feature prompts callers to
enter an identification code, which is registered on the CDR. It is often
used in colleges and universities where roommates share the same
extension number. Many professional organizations use account codes
to allocate calls to clients.
_ Hoteling: A station user can temporarily move to another location, log in,
and have station features including the extension number follow to the
new location. Intervention from the administrator is not required.
_ Paging access: The PBX can be equipped with paging trunks that
connect to an external paging system. The trunk is reached by dialing
an extension number or trunk access code. Zone paging, which allows
paging in specific locations rather than the entire building is available
on most systems.
_ Personal call routing: Users can define routing of incoming calls based
on variables such as time of day, calling number, etc.
_ Portable directory number: Allows a user on a networked PBX to move
from one switch to another without changing the telephone number.
_ Priority ringing: A distinctive ring is used for calls from specified
numbers.
_ Recorded announcements: This feature provides announcements for vacant
and disconnected numbers.
_ Trunk answer any station: This feature allows stations to answer incoming
trunks when the attendant station is busy.
_ Whisper page: A user can bridge into a call and speak to the local user
without the other end hearing.
Attendant Features
Most PBXs have attendant consoles for incoming call answer and supervision. The
attendant can also act as a central information source for directory and call assistance.
The console is either a specialized telephone instrument or a PC running a
console program. The latter is increasingly popular because it can be easily integrated
with a directory. The following features are important for most consoles
and represent only a fraction of the features available.
_ Attendant controlled conferencing: Attendant can set up multiport
conference calls.
_ Automatic timed reminders: Alerts the attendant when a called line has not
answered within a prescribed time.
_ Busy lamp field: When the station is busy or in do-not-disturb mode,
an LED associated with the station is lighted.
_ Direct station selection (DSS): Allows the attendant to call stations by
pressing an illuminated button associated with the line. The line button
shows busy or idle status.
_ Directory features: Attendants with PC-based consoles may be able
to search by first and last name, department, and extension.
_ Night service: Calls are automatically transferred to an alternate destination
when the console is closed. In many systems this feature is sensitive
to time of day and day of week.
System Administration Features
System administration is a costly element of every PBX, so features that ease
the administrator’s job are valuable. The following are some of the more popular
features.
_ Automatic set relocation: Allows users to move their telephones from one
location to another without the need to retranslate. The administrator
gives users a code and instructions to carry the set to the new location,
plug it in, and dial the code. When this is complete the system moves
the station translations to the new port. This feature is inherent with
IP systems, which may enable a user to log in from any available
Ethernet port.
_ LDAP synchronization: Enables the system to update its PBX and voice
mail database from customer’s LDAP directory. Eliminates or reduces
redundant database entries. The application may also permit the administrator
to work translations in software in advance, and then upload
them to the PBX.
_ Network move: Similar to automatic set relocation, this feature works
across a network, where automatic set relocation works only in the
same PBX.
This feature, which is discussed in more detail in Chapter 29, integrates the PBX
with voice mail, fax, and e-mail so that messages can be viewed and handled on
a PC screen. The feature may also enable users to translate messages from one format
to another. For example, e-mail messages may be read in synthesized voice if
the user is calling from a telephone and wants them read out. Eventually, with
improvement in speech-to-text software, voice-mail messages will be converted to
e-mail or fax. Currently, speech to text enables users to speak limited commands
to read, forward, and delete voice mail messages from a telephone.
Emergency Service Interface
Most of the developed world has adopted a special dialing code such as 999 or 911
for universal access to emergency services. The basic service enables the PSAP to
hold up the line so it can be traced in case the caller is unable to report the address
of the emergency. Enhanced emergency services contain a database that associates
telephone numbers with street addresses. The street address is often not a fine
enough distinction, however. Users who dial the emergency code from hotels,
apartments with a shared-tenant PBX, campuses, and multibuilding developments
and the like may be difficult or impossible to locate. Therefore, a trend is
toward reporting the station identification to the PSAP so it can be associated with
the room number or building name. The service is known as private system automatic
line identification (PS/ALI). The PRI feature in most PBXs can relay the station
identification to the LEC, which passes it to the PSAP. PBXs lacking PRI can
use a CAMA trunk as an alternative. The need for this feature is particularly acute
in PBXs with remote switch units that can be located in a different PSAP’s jurisdiction
from the host PBX. In this case, a separate trunk group to the local central
office is usually provided. The ARS is programmed so it always seizes a local
trunk when the emergency code is dialed from a station served by the remote.
A related feature that is important in any large PBX is the ability to route
emergency calls simultaneously to the console attendant or a security position.
The law requires that the call route directly to the PSAP, but management wants
to be informed of any such call. It is also a useful feature in hotels, campuses,
schools, hospitals, and other organizations where people may place emergency
calls inadvertently or as a prank.
Multitenant Service
PBXs that provide service to users from different organizations can use multitenant
software to give each organization the appearance of a private switch.
Multitenant service is a software partition in the PBX. Separate attendant consoles
can be provided, and each organization can have its own group of trunks and
block of numbers.
Property Management Interface
Hotels, hospitals, dormitories, and other organizations that resell service often
connect the PBX to a computer to provide features such as checking room status
information, disabling the telephone set from the attendant console, and determining
check-in or check-out status. The PBX provides information to the computer,
and accepts orders from the front desk via computer terminal. The PMI is a
specialized type of computer-telephony integration (see Chapter 27).
Uniform Dialing Plan (UDP)
UDP software in a multi-PBX network enables the caller to dial an extension number
and have the call completed over a tie line network without the caller’s being
concerned about where the extension is located. The PBX selects the route and
takes care of station number translations. UDP software is effective only among
PBXs of the same manufacture although it can work with QSIG-compatible PBXs
if they are so equipped.
Simplified Message Desk Interface (SMDI)
SMDI is a standard way of interfacing a switch to peripheral equipment such as
voice mail. The voice mail connects to the PBX over analog ports or line-side T1
and to the SMDI with a serial connection. For a call going to a messaging unit such
as voice mail, the SMDI link indicates the port the call is using, the type of call,
information about the call such as the source and destination, and the reason the
call is forwarded such as busy or no answer. The SMDI is an open protocol for
interfacing voice mail to the switch as an alternative to the manufacturer’s
proprietary interface.
PBX Voice Features
As all PBXs are designed for voice switching service, they have features intended
for the convenience and productivity of the users. Not all the features listed below
are universally available, and many systems provide features not listed. This list,
in addition to the key system features discussed earlier, briefly describes the most
popular voice features found in PBXs.
_ Automatic call trace: Harassing or nuisance call can be traced to the origin
by dialing an access code.
_ Call blocking: Users can selectively block calls such as specific extensions,
numbers, or calls from particular trunk groups.
_ Call coverage: Users can have one or more coverage paths to direct
how calls route when the called station is busy, does not answer, or is
in do-not-disturb status. External calls can take a different path than
internal calls.
_ Executive override: This feature allows a station to interrupt a busy line
or preempt a long distance trunk.
_ Forced account code: On long distance calls, this feature prompts callers to
enter an identification code, which is registered on the CDR. It is often
used in colleges and universities where roommates share the same
extension number. Many professional organizations use account codes
to allocate calls to clients.
_ Hoteling: A station user can temporarily move to another location, log in,
and have station features including the extension number follow to the
new location. Intervention from the administrator is not required.
_ Paging access: The PBX can be equipped with paging trunks that
connect to an external paging system. The trunk is reached by dialing
an extension number or trunk access code. Zone paging, which allows
paging in specific locations rather than the entire building is available
on most systems.
_ Personal call routing: Users can define routing of incoming calls based
on variables such as time of day, calling number, etc.
_ Portable directory number: Allows a user on a networked PBX to move
from one switch to another without changing the telephone number.
_ Priority ringing: A distinctive ring is used for calls from specified
numbers.
_ Recorded announcements: This feature provides announcements for vacant
and disconnected numbers.
_ Trunk answer any station: This feature allows stations to answer incoming
trunks when the attendant station is busy.
_ Whisper page: A user can bridge into a call and speak to the local user
without the other end hearing.
Attendant Features
Most PBXs have attendant consoles for incoming call answer and supervision. The
attendant can also act as a central information source for directory and call assistance.
The console is either a specialized telephone instrument or a PC running a
console program. The latter is increasingly popular because it can be easily integrated
with a directory. The following features are important for most consoles
and represent only a fraction of the features available.
_ Attendant controlled conferencing: Attendant can set up multiport
conference calls.
_ Automatic timed reminders: Alerts the attendant when a called line has not
answered within a prescribed time.
_ Busy lamp field: When the station is busy or in do-not-disturb mode,
an LED associated with the station is lighted.
_ Direct station selection (DSS): Allows the attendant to call stations by
pressing an illuminated button associated with the line. The line button
shows busy or idle status.
_ Directory features: Attendants with PC-based consoles may be able
to search by first and last name, department, and extension.
_ Night service: Calls are automatically transferred to an alternate destination
when the console is closed. In many systems this feature is sensitive
to time of day and day of week.
System Administration Features
System administration is a costly element of every PBX, so features that ease
the administrator’s job are valuable. The following are some of the more popular
features.
_ Automatic set relocation: Allows users to move their telephones from one
location to another without the need to retranslate. The administrator
gives users a code and instructions to carry the set to the new location,
plug it in, and dial the code. When this is complete the system moves
the station translations to the new port. This feature is inherent with
IP systems, which may enable a user to log in from any available
Ethernet port.
_ LDAP synchronization: Enables the system to update its PBX and voice
mail database from customer’s LDAP directory. Eliminates or reduces
redundant database entries. The application may also permit the administrator
to work translations in software in advance, and then upload
them to the PBX.
_ Network move: Similar to automatic set relocation, this feature works
across a network, where automatic set relocation works only in the
same PBX.
chapter24
Direct Inward System Access (DISA)
The DISA feature enables external callers to dial a telephone number and password
to gain access to PBX features. The DISA port can be restricted to limit calls
to internal extensions, tie lines, local calls, or any other restriction level used in the
PBX. If the DISA port is unrestricted, callers can gain access to long distance services.
DISA helps reduce credit card calls by enabling users outside the PBX to
access low-cost long distance services.
Security is an obvious problem with DISA. It is one of the most prevalent targets
for toll thieves, who use it to place calls at the company’s expense. The best
practice is to disable DISA. If it must be used, managers should change the password
frequently and check the call accounting system for evidence of misuse.
N × 64 Capability
With the growth of video conferencing, it is often desirable to dial more bandwidth
than an ordinary BRI connection provides. Conference-quality video usually
requires at least 384 Kbps, which is six 64 Kbps channels. A PBX with N × 64
capability enables the user to dial as many channels of contiguous bandwidth as
required.
Centralized Attendant Service (CAS)
CAS enables attendants at one location to handle attendant functions for remote
PBXs over a network. Although each PBX has its own group of trunks, all calls
routed to the attendant flow to the centralized location over the network. The
attendant can terminate the call to any station or hunt group. A related feature is
release-link trunk, which enables the PBX to release the attendant trunk after setting
up the call. Without this feature the trunk is tied up for the duration of the call.
Power-Fail Transfer
Unless a PBX is configured to run from batteries or from an uninterruptible power
supply, a commercial power failure will cause the system to fail. The power-failtransfer
feature connects central office trunks to standard DTMF telephones. Since
most PBXs require ground-start trunks, provisions must be made to operate from
loop-start telephones. This can be accomplished by two methods: use a separate
loop-start-to-ground-start converter or equip the telephones with a ground start
button. The former method is prevalent.
Power-fail transfer is an inexpensive and effective way to obtain minimum
service during power failure conditions. Even users of systems with battery
backup or UPS should consider power-fail transfer to retain some service if the
PBX itself fails. Some manufacturers offer power-fail transfer for digital or ISDN
trunks, which can enable the owner to avoid analog trunks.
Automatic Call Distribution (ACD)
ACD enables PBXs to route incoming calls to a group of service positions. Typical
applications are sales and customer service positions. Incoming calls route to an
agent position based on logic programmed into the switch. Calls can be routed
based on the toll-free number that was dialed using DNIS. The caller’s telephone
number may be delivered by the network and used to route calls, or an automated
attendant or call-prompting software in the switch can prompt the caller to select
from a menu of routing options.
When agent positions are idle, the call routes to an agent immediately. If all
positions are occupied, the ACD holds calls in queue and notifies the caller by
recorded announcement that the call is being delayed. Calls can be overflowed to
other agent groups, routed to voice mail so the caller can request a callback, or
handled in a variety of different ways, which Chapter 27 discusses in more detail.
ACD is one of the most important features in a PBX, and is included in more than
three-fourths of the systems shipped.
Uniform Call Distribution (UCD)
UCD distributes calls evenly among a group of stations. When one or more active
stations are idle, incoming calls are directed to the station that is next in line to
receive a call. When all stations in the UCD group are busy, incoming calls are
answered with a recording and held in queue. When a UCD station becomes idle,
the call that has been in queue the longest is directed to the station. In many UCD
systems, a station user can toggle between active and inactive status by dialing a
code or pressing a feature button. Compared to ACD, UCD is unsophisticated,
lacking the supervisory, management, and reporting features that an ACD offers.
Chapter 27 discusses UCD further
The DISA feature enables external callers to dial a telephone number and password
to gain access to PBX features. The DISA port can be restricted to limit calls
to internal extensions, tie lines, local calls, or any other restriction level used in the
PBX. If the DISA port is unrestricted, callers can gain access to long distance services.
DISA helps reduce credit card calls by enabling users outside the PBX to
access low-cost long distance services.
Security is an obvious problem with DISA. It is one of the most prevalent targets
for toll thieves, who use it to place calls at the company’s expense. The best
practice is to disable DISA. If it must be used, managers should change the password
frequently and check the call accounting system for evidence of misuse.
N × 64 Capability
With the growth of video conferencing, it is often desirable to dial more bandwidth
than an ordinary BRI connection provides. Conference-quality video usually
requires at least 384 Kbps, which is six 64 Kbps channels. A PBX with N × 64
capability enables the user to dial as many channels of contiguous bandwidth as
required.
Centralized Attendant Service (CAS)
CAS enables attendants at one location to handle attendant functions for remote
PBXs over a network. Although each PBX has its own group of trunks, all calls
routed to the attendant flow to the centralized location over the network. The
attendant can terminate the call to any station or hunt group. A related feature is
release-link trunk, which enables the PBX to release the attendant trunk after setting
up the call. Without this feature the trunk is tied up for the duration of the call.
Power-Fail Transfer
Unless a PBX is configured to run from batteries or from an uninterruptible power
supply, a commercial power failure will cause the system to fail. The power-failtransfer
feature connects central office trunks to standard DTMF telephones. Since
most PBXs require ground-start trunks, provisions must be made to operate from
loop-start telephones. This can be accomplished by two methods: use a separate
loop-start-to-ground-start converter or equip the telephones with a ground start
button. The former method is prevalent.
Power-fail transfer is an inexpensive and effective way to obtain minimum
service during power failure conditions. Even users of systems with battery
backup or UPS should consider power-fail transfer to retain some service if the
PBX itself fails. Some manufacturers offer power-fail transfer for digital or ISDN
trunks, which can enable the owner to avoid analog trunks.
Automatic Call Distribution (ACD)
ACD enables PBXs to route incoming calls to a group of service positions. Typical
applications are sales and customer service positions. Incoming calls route to an
agent position based on logic programmed into the switch. Calls can be routed
based on the toll-free number that was dialed using DNIS. The caller’s telephone
number may be delivered by the network and used to route calls, or an automated
attendant or call-prompting software in the switch can prompt the caller to select
from a menu of routing options.
When agent positions are idle, the call routes to an agent immediately. If all
positions are occupied, the ACD holds calls in queue and notifies the caller by
recorded announcement that the call is being delayed. Calls can be overflowed to
other agent groups, routed to voice mail so the caller can request a callback, or
handled in a variety of different ways, which Chapter 27 discusses in more detail.
ACD is one of the most important features in a PBX, and is included in more than
three-fourths of the systems shipped.
Uniform Call Distribution (UCD)
UCD distributes calls evenly among a group of stations. When one or more active
stations are idle, incoming calls are directed to the station that is next in line to
receive a call. When all stations in the UCD group are busy, incoming calls are
answered with a recording and held in queue. When a UCD station becomes idle,
the call that has been in queue the longest is directed to the station. In many UCD
systems, a station user can toggle between active and inactive status by dialing a
code or pressing a feature button. Compared to ACD, UCD is unsophisticated,
lacking the supervisory, management, and reporting features that an ACD offers.
Chapter 27 discusses UCD further
C H A P T E R 24 Customer-Premise Switching System Features
QSIG
TDM PBXs use a proprietary protocol similar to ISDN for networking between
their own products, but networking with switches of another manufacturer is
impractical. The QSIG protocol, named after Q.931 ISDN signaling, is designed to
support feature transparency and sharing of common resources such as voice mail
between disparate products in a private integrated services network (PISN). The
official ISO name for the protocol is private signaling system No. 1 (PSS1). QSIG
separates the bearer channels from the signaling, which uses a separate packetbased
signaling channel. QSIG can also be used with VoIP, where it offers the
advantage of potentially reducing the number of hops needed for a call.
Networking Comparison between TDM and VolP
The first layer of the protocol is called QSIG basic call, which supports transparency
between multivendor nodes. All products that claim QSIG compliance
must support basic services. The BC feature set is intended for call control, but a
higher layer known as QSIG generic function (GF) or QSIG supplementary services
supports additional services such as calling line identification. QSIG capability
is important for users with PBXs from different manufacturers to network
them together. QSIG support is rarely found in hybrids.
Station Restrictions
An important feature of every PBX is its ability to limit the calling privileges of
selected stations. Even companies that leave employees’ extensions unrestricted
normally require toll-restricted telephones in public locations such as waiting
areas and lunchrooms. The type of restriction varies with manufacturer, but it is
possible with most systems to restrict incoming, outgoing, and any type of long
distance. One class of employee could be given international access, for example,
while others are restricted to domestic calls. Some systems can restrict down to a
specific telephone number. All restriction systems should be able to restrict
selected area codes and prefixes. Area code restriction is necessary to prevent
users from calling certain chargeable numbers, such as 900 numbers and to certain
area codes that are known destinations for toll thieves.
Many systems provide an override feature that enables a user to dial an
access code and identification number. This removes the restriction from a phone
for the current session, and restores it when the call terminates. Another common
feature is time-of-day restriction, which leaves phones open during working
hours, but restricts them after hours.
Follow-Me Forwarding
With the increasing importance of telecommuting, several manufacturers are
offering this feature, which allows the user to receive telephone calls at home, on
a cell phone, or in a remote location such as a conference center. The user keeps
the PBX informed of his or her location, and the PBX forwards calls accordingly.
With caller ID and the appropriate programming, the system can screen calls as
well, and forward calls from only selected users. Forwarding can be selective
depending on time of day and day of week. At the user’s option, the system can
be programmed to ring to different destinations simultaneously or sequentially in
patterns that can be changed for different days or times. This feature is available
from both TDM and IP systems, usually as an extra-cost option. It requires a central
server that typically is accessed with a browser over the Web or a private IP
network. The protocol is usually proprietary on TDM systems and either proprietary
or SIP in IP systems. The most effective products link the application to an
electronic calendar.
When a user is available via e-mail, some applications can download voice
mail as e-mail file attachments that the user can play back on a laptop computer.
This is usually a feature of unified messaging (UM), which is discussed briefly
later in this chapter and in more detail in Chapter 29.
Call Detail Recording (CDR)
This feature, sometimes known as station message detail recording (SMDR), in
combination with a call accounting system provides the equivalent of a detailed
toll statement for PBX users. Many businesses require call detail to control
long distance usage and to spread costs among the user departments. The CDR
port is a serial connection that outputs the raw call details in ASCII using a
proprietary field format. A call accounting system, which is discussed later in
this chapter, connected to the serial port parses the detail, rates the calls, and
formats various management reports such as budgetary detail and individual
toll statements.
Voice Mail
Voice mail (see Chapter 28) is available as an optional feature of all PBXs and
hybrids and is one that is almost invariably applied. When a station is busy or
unattended, the caller can leave a message, which is stored digitally on a hard
disk. The station user can dial an access and identification code to retrieve the
message. Most voice-mail systems include automated attendant, an option that
enables callers with a DTMF dial to route their own calls within the system.
Incoming calls are greeted with an announcement that invites them to dial the
extension number if they know it or to stay on the line for an attendant. Most
voice-mail systems also support dial by name for callers who reach the automated
attendant and do not know the extension number.
Dialed Number Identification System (DNIS)
Offered by IXCs along with T1/E1-based toll-free services, DNIS provides the
equivalent of DID for toll-free calls. If multiple toll-free numbers are terminated
on the same switch, DNIS digits are sent with the call to identify the number
dialed to the PBX so it can route the call to the appropriate station or group. DNIS
enables an organization to have several toll-free numbers and to route each call to
a different station, UCD or ACD hunt group, voice mail, or any other destination
within the PBX. For example, if a company has different ACD groups for sales,
service, and order inquiry, it can assign each of these groups a different toll-free
number and use DNIS to route the calls appropriately. This alternative is often
more effective than using an auto attendant to answer the call and offer a menu of
choices.
TDM PBXs use a proprietary protocol similar to ISDN for networking between
their own products, but networking with switches of another manufacturer is
impractical. The QSIG protocol, named after Q.931 ISDN signaling, is designed to
support feature transparency and sharing of common resources such as voice mail
between disparate products in a private integrated services network (PISN). The
official ISO name for the protocol is private signaling system No. 1 (PSS1). QSIG
separates the bearer channels from the signaling, which uses a separate packetbased
signaling channel. QSIG can also be used with VoIP, where it offers the
advantage of potentially reducing the number of hops needed for a call.
Networking Comparison between TDM and VolP
The first layer of the protocol is called QSIG basic call, which supports transparency
between multivendor nodes. All products that claim QSIG compliance
must support basic services. The BC feature set is intended for call control, but a
higher layer known as QSIG generic function (GF) or QSIG supplementary services
supports additional services such as calling line identification. QSIG capability
is important for users with PBXs from different manufacturers to network
them together. QSIG support is rarely found in hybrids.
Station Restrictions
An important feature of every PBX is its ability to limit the calling privileges of
selected stations. Even companies that leave employees’ extensions unrestricted
normally require toll-restricted telephones in public locations such as waiting
areas and lunchrooms. The type of restriction varies with manufacturer, but it is
possible with most systems to restrict incoming, outgoing, and any type of long
distance. One class of employee could be given international access, for example,
while others are restricted to domestic calls. Some systems can restrict down to a
specific telephone number. All restriction systems should be able to restrict
selected area codes and prefixes. Area code restriction is necessary to prevent
users from calling certain chargeable numbers, such as 900 numbers and to certain
area codes that are known destinations for toll thieves.
Many systems provide an override feature that enables a user to dial an
access code and identification number. This removes the restriction from a phone
for the current session, and restores it when the call terminates. Another common
feature is time-of-day restriction, which leaves phones open during working
hours, but restricts them after hours.
Follow-Me Forwarding
With the increasing importance of telecommuting, several manufacturers are
offering this feature, which allows the user to receive telephone calls at home, on
a cell phone, or in a remote location such as a conference center. The user keeps
the PBX informed of his or her location, and the PBX forwards calls accordingly.
With caller ID and the appropriate programming, the system can screen calls as
well, and forward calls from only selected users. Forwarding can be selective
depending on time of day and day of week. At the user’s option, the system can
be programmed to ring to different destinations simultaneously or sequentially in
patterns that can be changed for different days or times. This feature is available
from both TDM and IP systems, usually as an extra-cost option. It requires a central
server that typically is accessed with a browser over the Web or a private IP
network. The protocol is usually proprietary on TDM systems and either proprietary
or SIP in IP systems. The most effective products link the application to an
electronic calendar.
When a user is available via e-mail, some applications can download voice
mail as e-mail file attachments that the user can play back on a laptop computer.
This is usually a feature of unified messaging (UM), which is discussed briefly
later in this chapter and in more detail in Chapter 29.
Call Detail Recording (CDR)
This feature, sometimes known as station message detail recording (SMDR), in
combination with a call accounting system provides the equivalent of a detailed
toll statement for PBX users. Many businesses require call detail to control
long distance usage and to spread costs among the user departments. The CDR
port is a serial connection that outputs the raw call details in ASCII using a
proprietary field format. A call accounting system, which is discussed later in
this chapter, connected to the serial port parses the detail, rates the calls, and
formats various management reports such as budgetary detail and individual
toll statements.
Voice Mail
Voice mail (see Chapter 28) is available as an optional feature of all PBXs and
hybrids and is one that is almost invariably applied. When a station is busy or
unattended, the caller can leave a message, which is stored digitally on a hard
disk. The station user can dial an access and identification code to retrieve the
message. Most voice-mail systems include automated attendant, an option that
enables callers with a DTMF dial to route their own calls within the system.
Incoming calls are greeted with an announcement that invites them to dial the
extension number if they know it or to stay on the line for an attendant. Most
voice-mail systems also support dial by name for callers who reach the automated
attendant and do not know the extension number.
Dialed Number Identification System (DNIS)
Offered by IXCs along with T1/E1-based toll-free services, DNIS provides the
equivalent of DID for toll-free calls. If multiple toll-free numbers are terminated
on the same switch, DNIS digits are sent with the call to identify the number
dialed to the PBX so it can route the call to the appropriate station or group. DNIS
enables an organization to have several toll-free numbers and to route each call to
a different station, UCD or ACD hunt group, voice mail, or any other destination
within the PBX. For example, if a company has different ACD groups for sales,
service, and order inquiry, it can assign each of these groups a different toll-free
number and use DNIS to route the calls appropriately. This alternative is often
more effective than using an auto attendant to answer the call and offer a menu of
choices.
C H A P T E R 24 Customer-Premise Switching System Features
PBX FEATURES
This section discusses the main features that most PBXs and many hybrids, both
TDM and IP, support. These features are in addition to the key system features
discussed in the previous section. Two features on the key system list, flash and
paging, are generally available on hybrids but unavailable on PBXs. Although the
features in this section are common to most PBXs, users will find operational differences
among products. There are also differences in whether the features are
standard or an extra-cost option.
Direct Inward Dialing (DID)
DID offers station users the ability to receive calls from outside the system without
going through the attendant. The LEC’s central office contains a software table
with the location of the DID trunk group. When a call for a DID number arrives,
the central office seizes an idle trunk and outpulses the extension number, usually
with DTMF tones or over a channel. DID is effective in reducing the load on PBX
attendants. It also enables users to receive calls when the switchboard is closed.
DID is provided on both analog and digital trunks. On analog trunks a separate
trunk group is required. Digital trunks may be provisioned as tie lines between
the PBX and the central office to provide two-way DID. PRI trunks offer call-bycall
service selection, enabling any trunk to be used for any purpose.
Automatic Route Selection (ARS)
Most PBXs terminate a combination of public switched and private trunks on the
system. For example, in addition to local trunks, the PBX may terminate T1/E1
lines to the IXC, FEX lines, and tie trunks to another PBX. Educating users about
which service to use is impractical, particularly as rates vary with time of day and
terminating location, and the dialing plan varies with the type of service. It is a
reasonably simple matter, however, to program route selection into the central
processor of the PBX. With ARS, sometimes called least-cost routing (LCR), the
user dials the number and the system determines the preferred route and dials the
digits to complete the call over the appropriate trunk group.
The most sophisticated ARS systems can screen calls on the entire dialed
number, but some simple systems, typical of hybrids, can screen on only the NPA
and prefix. The ability to screen on the entire number is important for many companies.
With it, for example, it is possible to allow users to dial some 900 numbers,
but deny others. If a company has an IP gateway that enables it to call other company
numbers in an overseas location, the ARS can route those calls to the gateway,
and domestic calls to the PSTN. ARS can also select the trunk group based
on class of service. One class could call internationally only over IP trunks, while
international calls for another class always use the PSTN.
A related issue is digit insertion and deletion. Some services, such as FEX,
may require the PBX to insert or delete an area code for correct routing. Telephone
service is easiest for users if they always dial the same way regardless of the route
the call takes. For example, if the PBX has FEX trunks to another area code, the
user would dial the area code, but the PBX would strip it off before passing the
digits forward to the FEX trunks if the central office does not require all 10 digits.
Users cannot be expected to understand the logic of this arrangement.
Networking Options
Most PBXs offer networking options, which allow multiple PBXs to operate as a
single system. Networking is available on most PBXs, but it is rare in hybrids.
Call-processing messages pass between PBXs over a separate data channel using
IP messages or some form of common channel signaling. With the networking
option, call-processing information such as a station’s identification and class of
service travel across the network to permit features to operate in a distant PBX as
they do in the local system. This feature is known as traveling class mark.
The objective of networking is to provide complete feature transparency,
which is the ability of users to have the same calling features across the network
as they have at the main PBX. For example, users want to be able to camp on a
busy station, regardless of whether it is in their PBX or in a distant system, and
they want to share a voice-mail system across the network. Some features do not
work across a network in some products. Call pickup, for example, enables a user
who hears a ringing telephone to press a button and bring the call to his or her
telephone. The lack of this feature across a network is usually unimportant since
users are normally in separate locations and cannot hear the bell. Some companies,
however, start with separate systems in separate locations and later merge
them. The PBXs are collocated in the same equipment room and remain networked
together. If features such as call pickup do not work across the network,
users in one work group must be assigned to the same switch, which often
requires moving people from one PBX to another and possibly changing numbers.
TDM and IP PBXs have significant differences in the way they implement
networking. Figure 24-1 illustrates some of the differences. In the top half of
the figure three TDM switches are used. Each switch has a unique number range
from a different central office and a local trunk group to that office. The database
in each switch contains the details on each number range in its domain. Its
ARS knows which trunk group to use to reach an extension in either of the other
switches, but it does not contain the translations for the stations in the other
switches.
The IP configuration, by contrast, has three servers, each with an identical
database. If one of the servers fails, the other servers can support its stations,
which are attached directly to the LAN behind the routers. Each server has direct
access into the IP network, but the connectionless nature of IP enables any switch
to set up a path to either of the other switches. This configuration provides survivability
that the TDM model lacks. It also enables users to move between
switches while retaining their telephone numbers
This section discusses the main features that most PBXs and many hybrids, both
TDM and IP, support. These features are in addition to the key system features
discussed in the previous section. Two features on the key system list, flash and
paging, are generally available on hybrids but unavailable on PBXs. Although the
features in this section are common to most PBXs, users will find operational differences
among products. There are also differences in whether the features are
standard or an extra-cost option.
Direct Inward Dialing (DID)
DID offers station users the ability to receive calls from outside the system without
going through the attendant. The LEC’s central office contains a software table
with the location of the DID trunk group. When a call for a DID number arrives,
the central office seizes an idle trunk and outpulses the extension number, usually
with DTMF tones or over a channel. DID is effective in reducing the load on PBX
attendants. It also enables users to receive calls when the switchboard is closed.
DID is provided on both analog and digital trunks. On analog trunks a separate
trunk group is required. Digital trunks may be provisioned as tie lines between
the PBX and the central office to provide two-way DID. PRI trunks offer call-bycall
service selection, enabling any trunk to be used for any purpose.
Automatic Route Selection (ARS)
Most PBXs terminate a combination of public switched and private trunks on the
system. For example, in addition to local trunks, the PBX may terminate T1/E1
lines to the IXC, FEX lines, and tie trunks to another PBX. Educating users about
which service to use is impractical, particularly as rates vary with time of day and
terminating location, and the dialing plan varies with the type of service. It is a
reasonably simple matter, however, to program route selection into the central
processor of the PBX. With ARS, sometimes called least-cost routing (LCR), the
user dials the number and the system determines the preferred route and dials the
digits to complete the call over the appropriate trunk group.
The most sophisticated ARS systems can screen calls on the entire dialed
number, but some simple systems, typical of hybrids, can screen on only the NPA
and prefix. The ability to screen on the entire number is important for many companies.
With it, for example, it is possible to allow users to dial some 900 numbers,
but deny others. If a company has an IP gateway that enables it to call other company
numbers in an overseas location, the ARS can route those calls to the gateway,
and domestic calls to the PSTN. ARS can also select the trunk group based
on class of service. One class could call internationally only over IP trunks, while
international calls for another class always use the PSTN.
A related issue is digit insertion and deletion. Some services, such as FEX,
may require the PBX to insert or delete an area code for correct routing. Telephone
service is easiest for users if they always dial the same way regardless of the route
the call takes. For example, if the PBX has FEX trunks to another area code, the
user would dial the area code, but the PBX would strip it off before passing the
digits forward to the FEX trunks if the central office does not require all 10 digits.
Users cannot be expected to understand the logic of this arrangement.
Networking Options
Most PBXs offer networking options, which allow multiple PBXs to operate as a
single system. Networking is available on most PBXs, but it is rare in hybrids.
Call-processing messages pass between PBXs over a separate data channel using
IP messages or some form of common channel signaling. With the networking
option, call-processing information such as a station’s identification and class of
service travel across the network to permit features to operate in a distant PBX as
they do in the local system. This feature is known as traveling class mark.
The objective of networking is to provide complete feature transparency,
which is the ability of users to have the same calling features across the network
as they have at the main PBX. For example, users want to be able to camp on a
busy station, regardless of whether it is in their PBX or in a distant system, and
they want to share a voice-mail system across the network. Some features do not
work across a network in some products. Call pickup, for example, enables a user
who hears a ringing telephone to press a button and bring the call to his or her
telephone. The lack of this feature across a network is usually unimportant since
users are normally in separate locations and cannot hear the bell. Some companies,
however, start with separate systems in separate locations and later merge
them. The PBXs are collocated in the same equipment room and remain networked
together. If features such as call pickup do not work across the network,
users in one work group must be assigned to the same switch, which often
requires moving people from one PBX to another and possibly changing numbers.
TDM and IP PBXs have significant differences in the way they implement
networking. Figure 24-1 illustrates some of the differences. In the top half of
the figure three TDM switches are used. Each switch has a unique number range
from a different central office and a local trunk group to that office. The database
in each switch contains the details on each number range in its domain. Its
ARS knows which trunk group to use to reach an extension in either of the other
switches, but it does not contain the translations for the stations in the other
switches.
The IP configuration, by contrast, has three servers, each with an identical
database. If one of the servers fails, the other servers can support its stations,
which are attached directly to the LAN behind the routers. Each server has direct
access into the IP network, but the connectionless nature of IP enables any switch
to set up a path to either of the other switches. This configuration provides survivability
that the TDM model lacks. It also enables users to move between
switches while retaining their telephone numbers
C H A P T E R 24 Customer-Premise Switching System Features
_ Do not disturb: Users can press a button that silences the bell and
prevents intercom calls from reaching the station.
_ Forward all calls: Users can redirect all calls to another station or
destination.
_ Forward on busy or no answer: Users can redirect calls to another station
or destination if the line is busy or does not answer.
_ Held-line reminder: After a call has been left on hold for a specified
period, the telephone emits a warning tone.
_ Missed-call indicator: A list of unanswered calls is displayed
on the telephone.
_ Music on hold: While a call is on hold, music or a promotional
announcement is played.
_ Mute: A mute button on the telephone disables the microphone.
_ Paging: Stations can page over the telephone speaker.
_ Privacy: Prevents other stations from picking up a line that is in use.
In some systems privacy is automatic unless the user presses a privacy
release key.
_ Station restriction: Stations can be assigned to different classes of service
for restricting long distance calls.
_ Voice call: A user can place a call directly to the speaker of another user’s
telephone.
_ Volume control: The volume of the handset, speaker, and ringer
can be adjusted.
Caller ID can be provided in some key systems using one of two methods.
With analog lines, callers are identified with the ADSI protocol. Some key systems
support BRI ISDN, which is also capable of caller ID. In either case, the call identification
extends to the station display when the call is transferred.
Voice mail is readily available in key systems. Since DID is not a key system
feature, calls are either transferred to the user’s voice mail manually, or an auto
attendant prompts the caller to dial by name or extension number.
Key systems are often used across Centrex lines to provide features that
require dial-access codes on POTS telephones. Many LECs offer lines with
Centrex-like features such as transfer, conferencing, and third-party add-on. These
features are activated by sending a momentary on/off hook flash toward the central
office to signal for second dial tone. Flashing the switch hook of a key system
telephone signals the KSU to return second dial tone from the key system, not
from the central office. Therefore, most key systems provide a feature button, typically
called “flash” or “link” to flash the PSTN line. This feature is essential in any
key system that is intended to work behind Centrex. PBXs rarely provide this feature
because they are intended to interface trunk-side connections in the central
office, and these do not respond to switch hook flashes.
prevents intercom calls from reaching the station.
_ Forward all calls: Users can redirect all calls to another station or
destination.
_ Forward on busy or no answer: Users can redirect calls to another station
or destination if the line is busy or does not answer.
_ Held-line reminder: After a call has been left on hold for a specified
period, the telephone emits a warning tone.
_ Missed-call indicator: A list of unanswered calls is displayed
on the telephone.
_ Music on hold: While a call is on hold, music or a promotional
announcement is played.
_ Mute: A mute button on the telephone disables the microphone.
_ Paging: Stations can page over the telephone speaker.
_ Privacy: Prevents other stations from picking up a line that is in use.
In some systems privacy is automatic unless the user presses a privacy
release key.
_ Station restriction: Stations can be assigned to different classes of service
for restricting long distance calls.
_ Voice call: A user can place a call directly to the speaker of another user’s
telephone.
_ Volume control: The volume of the handset, speaker, and ringer
can be adjusted.
Caller ID can be provided in some key systems using one of two methods.
With analog lines, callers are identified with the ADSI protocol. Some key systems
support BRI ISDN, which is also capable of caller ID. In either case, the call identification
extends to the station display when the call is transferred.
Voice mail is readily available in key systems. Since DID is not a key system
feature, calls are either transferred to the user’s voice mail manually, or an auto
attendant prompts the caller to dial by name or extension number.
Key systems are often used across Centrex lines to provide features that
require dial-access codes on POTS telephones. Many LECs offer lines with
Centrex-like features such as transfer, conferencing, and third-party add-on. These
features are activated by sending a momentary on/off hook flash toward the central
office to signal for second dial tone. Flashing the switch hook of a key system
telephone signals the KSU to return second dial tone from the key system, not
from the central office. Therefore, most key systems provide a feature button, typically
called “flash” or “link” to flash the PSTN line. This feature is essential in any
key system that is intended to work behind Centrex. PBXs rarely provide this feature
because they are intended to interface trunk-side connections in the central
office, and these do not respond to switch hook flashes.
C H A P T E R 24 Customer-Premise Switching System Features
Key Telephone System Features
At one time key systems were electromechanical devices that had a limited set of
features. New electronic systems have far surpassed these limitations, but the features
of the last generation of electromechanical key systems defined the way users expect key telephones to work. The principal features are designated with
illuminated buttons. A dark button indicates an idle line and a solid light denotes
that the line is in use. A slow flash indicates an incoming call and a fast flash
shows that the line is on hold. These buttons and lamps define the following
features, which are common to all key systems:
_ Call pickup: Any station can access a line by pressing a line button.
_ Call hold: A hold button (usually red) can be pressed to hold the line
in the central unit. By contrast, the hold button on a POTS phone holds
the line in the telephone so the line cannot be used for another call.
_ Intercom: Acommon path shared by all telephones is used to announce calls.
_ Supervisory signals: Lamps show when a line is ringing, in use, or on hold.
_ Common bell: A bell common to all lines signals an incoming call. A slow
lamp flash shows which line is ringing.
The central control unit is known as a key system unit or KSU. Electronic
KSUs can support many additional features that are characteristic of most key systems.
The list below is in addition to the telephone set features such as last number
redial, message-waiting lamps, speakerphone, call logging, speed dial, etc.
In addition to the flashing lamp call indications, call status information may be
displayed on the telephone.
_ Automatic line selection: When the user picks up the phone, an outgoing
line is selected automatically.
_ Bridged call appearance: The same extension number can be terminated
on multiple phone sets.
_ Call drop: A call can be terminated without hanging up the receiver.
_ Call forwarding: Users can forward their calls to another station
in the system.
_ Call park: This feature places a call in a parking orbit so it can be
retrieved from any telephone in the system.
_ Call transfer: An incoming or outgoing call can be transferred
to another user.
_ Callback: If someone transfers a call to an extension that does not answer
after a set number of rings, the call returns to the original station.
_ Camp-on: Users or the attendant can send an external call to another
telephone even if it is busy. The callee hears a faint camp-on tone.
When the user hangs up, the camped-on call rings at the station.
_ Conferencing: Stations can bridge two or more lines together for
a multiparty conversation.
_ Distinctive ringing: Different ringing tones enable users to distinguish
between internal and PSTN calls.
At one time key systems were electromechanical devices that had a limited set of
features. New electronic systems have far surpassed these limitations, but the features
of the last generation of electromechanical key systems defined the way users expect key telephones to work. The principal features are designated with
illuminated buttons. A dark button indicates an idle line and a solid light denotes
that the line is in use. A slow flash indicates an incoming call and a fast flash
shows that the line is on hold. These buttons and lamps define the following
features, which are common to all key systems:
_ Call pickup: Any station can access a line by pressing a line button.
_ Call hold: A hold button (usually red) can be pressed to hold the line
in the central unit. By contrast, the hold button on a POTS phone holds
the line in the telephone so the line cannot be used for another call.
_ Intercom: Acommon path shared by all telephones is used to announce calls.
_ Supervisory signals: Lamps show when a line is ringing, in use, or on hold.
_ Common bell: A bell common to all lines signals an incoming call. A slow
lamp flash shows which line is ringing.
The central control unit is known as a key system unit or KSU. Electronic
KSUs can support many additional features that are characteristic of most key systems.
The list below is in addition to the telephone set features such as last number
redial, message-waiting lamps, speakerphone, call logging, speed dial, etc.
In addition to the flashing lamp call indications, call status information may be
displayed on the telephone.
_ Automatic line selection: When the user picks up the phone, an outgoing
line is selected automatically.
_ Bridged call appearance: The same extension number can be terminated
on multiple phone sets.
_ Call drop: A call can be terminated without hanging up the receiver.
_ Call forwarding: Users can forward their calls to another station
in the system.
_ Call park: This feature places a call in a parking orbit so it can be
retrieved from any telephone in the system.
_ Call transfer: An incoming or outgoing call can be transferred
to another user.
_ Callback: If someone transfers a call to an extension that does not answer
after a set number of rings, the call returns to the original station.
_ Camp-on: Users or the attendant can send an external call to another
telephone even if it is busy. The callee hears a faint camp-on tone.
When the user hangs up, the camped-on call rings at the station.
_ Conferencing: Stations can bridge two or more lines together for
a multiparty conversation.
_ Distinctive ringing: Different ringing tones enable users to distinguish
between internal and PSTN calls.
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