The optical carrier CSI

The DS carrier service infrastructure gave us two important building blocks that were used to further extend the  capacity for supporting VoIP networks. First and  foremost, the  DS network established that analog signals could be regenerated in digital  format. Second, the  DS network established that digital  signals could be aggregated with other digitally regenerated signals in the  form of DS0 channels. Thus,  the  capability to channelize digital  bandwidth evolved. Dedicated channels have  proven that they  can support VoIP with
the  same if not  better quality of service that we have  come to expect with
POTS over  the  PSTN.

The DS CSI

In 1964, the  carriers began channelizing and  aggregating analog inbound tele- phone calls onto digital,  high-bandwidth transports. The digital service (DS) carrier service infrastructure was born. The type  of wiring used for DS trans- ports was copper, like the  PSTN transport lines  in the PSTN. The DS transport lines,  however, were of a thicker gauge  and  were capable of sustaining higher bandwidth capacities. The carrier often referred to DS type  transport lines  as “high-cap” T1 lines  to distinguish them from other types of copper transport lines  in the  PSTN. Today most T1 transport lines  are provided using  fiber- optic lines.

The PSTN CSI

The public switched telephone network (PSTN) is the  oldest CSI, actually beginning with the work of Alexander Graham  Bell in the late 1800s.

Over the years, carriers have been installing, expanding, improving, and inter- connecting various parts of the  PSTN. Compared to all other carrier service infrastructures, the  PSTN is the  largest.

The Five Golden Rings of CSI

It’s useful  to understand from which CSI a network transport service comes. The CSI to which a transport service belongs affects the  way in which that transport service is implemented. sider the  case of VoIP. An integral part of VoIP is the  concept of transmitting voice  packets over  the  transport. But each CSI, each transport, and  the  VoIP transport service provided therein support voice packets differently. This is because each CSI operates with different underlying protocols.

Choosing a Transport

CSIs are made up of network transports (lines or channels) and  the carrier’s equipment used to terminate these lines.  Each CSI has  multiple carrier trans- ports to offer any customer.

A transport is a physical or wireless channel (or aggregate of contiguous channels) that supports the  transmission of electrical, optical data, telemet- ric data, voice,  or video  signals.

Road Map to VoIP Transports and Services

Let’s face it — telecommunications can be daunting to those who have not given much thought as to how their  voice gets from their  phone’s handset to their  Aunt Matilda  in Dubuque. Be that  as it may, the technology between you and Aunt Matilda  is simply amazing.

This chapter introduces you to the  wonderful world of networks, transports, and transport services. Here you discover what a CSI is (besides a great family of television shows) and  why you should even care. Before you are finished with this  chapter, you’ll have  a good  grasp of things you didn’t  even  know you needed to grasp. (Spooky,  huh?)

Applying VoIP to your situation

The moral of this  story?  If your  company has  facilities distributed across more  than one local calling  area  within the  same LATA, look at your intralata toll charges. If all your  sites switched to VoIP, you’d eliminate most, if not  all, intralata charges you currently incur.

The VoIP solution

I designed a VoIP network that provided a dedicated access line between the  two main locations. These sites were seven miles apart but  in different coun- ties  and  therefore different LATAs. We put  in digital subscriber line (DSL) access at the  other nine locations.

Analyzing the client’s usage

When I came  on the  scene, I analyzed their monthly billings  for the  past three months. I found that their total average monthly billings  for intralata services came to just  under a whopping $11,000 per  month, or a projected annualized billing of $130,000. Intralata recurring charges were about 63 percent of their total monthly telecommunications bill.

VoIP Savings: A Case Study

One of my clients in the  Pittsburgh area  has  eleven locations distributed across several local calling  areas within two Pittsburgh LATAs. Five locations are in the  city itself. The other six are in the  South Hills, with two inside Allegheny County but  outside the  city, and  four located to the south across the  county line in Washington County.  The client spent enormous amounts of money on phone service because an interoffice call between locations often  crossed intralata boundaries.

Add-on recurring costs

Tallying  the  costs of traditional phone service is like adding up the  cost of sending your  kid through college. There just  doesn’t seem to be an end in sight  to the  charges. As if the  access line costs and  recurring carrier service charges weren’t enough, you must deal  with other monthly costs and  regula- tory fees. These payments, which go to various government entities rather than to your  LEC, are based on a percentage of each line’s monthly access cost. Examples include the  Federal line surcharge and the 911 fee.

Saving with VoIP

If you’ve read the  chapter up to this  point, you’re  a much more  savvy POTS- PSTN customer. You now know exactly how your  carrier makes money at your  expense. You also  know how the  five regulated service categories can combine to increase your  monthly and  annual telephony costs and  therefore reduce your  revenue. Something that increases costs and  reduces revenue is something you need to control or change. VoIP can help  you do exactly that. So, how will you fare under a VoIP system?

Summing up carrier services

Several  layers of costs exist  whenever you make a phone call due  to regula- tions and  the  categories of carrier service. All five categories combine to form the  bulk of the  monthly recurring charges for customers (both residen- tial and  corporate) under the  existing PSTN model of telephony.

Interstate carrier service

Like Dante’s  circles of hell, phone carrier services just  keep  spreading out. The next  circle  of charges is interstate. Interstate includes calls to a destina- tion outside the  local calling  area’s state but  still inside the United  States. Interstate is sometimes referred to as calling  across state lines  or state-to- state calling.

Intrastate service rates

The next  service category is intrastate,  which involves carrier services for calls outside the  LATA but  inside the  boundaries of the  state where your local access line is installed.

As with intralata, if you tell your  LEC nothing about which intrastate carrier you want  to use when  you begin  the  lease of your  local access line, you auto- matically inherit the  LEC as your  intrastate carrier. Intrastate services are basically the  same as intralata services except they cover a much larger  geo- graphic area.  Intrastate is sometimes called interlata because several LATAs are situated in any given state.

Going the distance with intralata rates

Intralata refers to calls that terminate outside the  local calling  area  but within your  local access and  transport area  (LATA). Unfortunately, most people don’t  know the  boundaries of either their local calling  area  or their LATA. If they  did know, they  could manage their intralata calls and charges much better.

Paying the local piper

So you went  out  and  got yourself a local-access line for your  home or a slew of access lines  for your  business. Just  how much do you pay for the local calls  you place on those lines? Figuring  out  those costs is a little complicated.

Service Categories Cost You Big Time

If you’ve ever  tried to read your  monthly phone bill, you know that the system of charges for traditional phone services is virtually incomprehensi- ble to the  average person. One of the  big benefits of VoIP is that it makes the  POTS-PSTN model, together with its complicated billing structure and  weird  terminology, just  go away.

Everything You Need to Know About Charges

In the  old days  of making  calls with a telephone (remember, just  last year), you paid  for a phone line. Your company may have  had  one network to handle dozens or even  hundreds of phone lines  coming into your  business and  another network to handle computers. Now companies can converge both networks into one. By using  VoIP over  a private data network, your com- pany  can bypass the  older,  more  expensive way of using  the  public circuit-  switched network.

Converging Networks

Note the  absence of any POTS lines or pri- vate  telephone systems (KTS or PBX) under the  DS carrier service network cloud. All telephone calls are originating on the  company’s computer network using  VoIP. Only calls destined for the  PSTN are diverted off the  company’s network.

Private Systems versus VoIP

A private telephone system approach can’t begin  to compare to a VoIP model in terms of savings. Your guide  should be “How much telephone calling  traf- fic, across all five regulated PSTN charging categories, do you or your  com- pany  have  each month?” If your  monthly call volume, which is charged by the  minute for each line across each charging category, is substantial, a private telephone system model reduces your  recurring charges because you use fewer lines.  However, VoIP can reduce your recurring charges even  further, as you’ll discover in the  next  chapter.

The KTS and PBX models

The other two system models are private telephone systems installed on the  company’s premises. Low-volume customers often  use a key telephone system, or KTS. High-volume, larger  companies often  use a private branch exchange, or PBX. These two are a departure from the POTS-line model, where a line is run to each phone on the  premises. As such, they  are also  a departure from the  Centrex model, which uses the same type  of access line as POTS.

The Centrex model

The second model is the  central office exchange service, or Centrex, model. Centrex is physically set  up the  same as the  POTS access line model. Like POTS, Centrex uses the  same physical twisted-pair copper lines.

Private Telephone Systems Reduce POTS Line Costs

Computer data networks and  circuit-switched voice  networks are completely separate, with individual staffing,  billing, maintenance, and accounting sys- tems. Although the  maintenance costs of computer networks are affordable for most companies, the  recurring charges for traditional forms of telephony are huge  for small,  medium, and  large multilocation companies. VoIP is designed to converge (integrate) a company’s voice  needs onto the  com- pany’s  existing computer network. If a company does this,  they  can eliminate most (if not  all) recurring circuit-switched telephony charges.

War Breaks Out Between Circuits and Packets

The corporate sector’s thirst for leasing dedicated DS lines  was unquench- able.  Soon a dilemma emerged as to how to distinguish the circuit-switched network and  the  newer dedicated network, which used packet-switching technology. Since its inception, the  circuit-switched network was a public car- rier  services network. The DS network was being called dedicated, or private, because no one but  the  customer paying for the  DS lines  was permitted to use  them.

The digital services carrier network

The new types of digital  lines  installed by the  carriers began to form a new physical carrier services network. The lines  did not  cross-connect or inter- sect with any of the  millions of circuit-switched lines  that are in place and continue to be installed by the  carriers. At the  carrier company’s facilities, newer types of fully digital  equipment terminated these digital  lines.

The circuit-switched network gets organized

As circuit-switched networks continued to evolve, other technologies were developed that  helped the carriers manage their  telephony operations. Carriers began  offering more  types of POTS access and POTS carrier services.

Digital Telephony Invades PSTN Territory

When digital  networks were implemented back  in the  1960s, the telephone carrier companies began using  a technique that permitted them to accept analog telephone calls coming into their switching facilities and  convert those signals into digital  form for transmission on their shiny new networks. They  had  not  yet made the  leap  into packetizing telephone calls,  which is what  we have  today with VoIP. At the time,  they  thought it best to keep  the  circuit-switched telephone carrier network physically separate from the  evolving packet-switched computer network.

Combining Analog and Digital

When digital  networks were introduced, the phone companies wanted to use them right away because they provided a more  efficient  means of transmitting signals all over the place.  (Digital networks could  carry  data much  faster than analog  networks.) The phone companies were presented with a problem, how- ever: how to make existing  analog  phones work with a digital  network.

Telephony Goes Digital

Scientists, never content with two tin cans and  a string, looked for different ways to transmit sounds over  long distances. The pioneering work of Harry Nyquist in the  1920s gave us the  basics of sampling theorem. In the  1940s, Claude  Shannon would  mathematically prove Nyquist’s sampling theorem. Their work is the foundation for what we now call digital networking.

Analog Telephone Circuits

As mentioned, phone technology originally was analog,  from start to finish. Analog modulation is the technique used to convert sounds (such as your voice)  into an electromagnetic form. The analog  circuitry of the POTS tele- phone transmitter converts the voice patterns coming  from the caller’s mouth into continuous electromagnetic signal patterns. These  patterns are carried on a telephone line circuit, sometimes called  a trunk line, where they are carried to the terminating end of the circuit. There,  analog  circuitry converts the signal back into audible sounds so they can be understood by humans.

VoIP: Not Your Father’s Telephone Service (chapter 2)

Voice  over  IP represents a significant change from the  traditional way that telephone calls have  been handled until recently. Even so, the genesis
of VoIP is rooted in the  history of networks, specifically, the  history of the  circuit-switched phone network.

Looking at the TCP/IP Model

Many people marvel  at the  very thought that the  POTS method of placing telephone calls can be replaced by a technology that essentially runs on the  computer network. They  are also  startled by the  many  new and exciting fea- tures that come with VoIP. However, people also  question how VoIP can possi-  bly work and  are a bit suspicious about whether VoIP can really live up to all the  claims.

Gaining Flexibility with VoIP

VoIP is not  just  about making  and  receiving telephone calls; it’s about a whole  new way of communicating. Sure, it includes telephone calls,  but there is so much more  to the  VoIP telephony picture. VoIP integrates most if not  all other forms of communication. You can even  run videoconferencing to your  desktop.

Making internal calls | Making external calls (chapter 1)

Making internal calls

When you want  to call a coworker at your  same location, you dial the phone number corresponding to the  person’s name. The signals are packetized and sent to the  managing server, where the  packet picks  up the MAC address of the  person you’re  calling.  Next, the  packet is forwarded to the  switch, then to a particular port on that switch, and finally to the  IP telephone connected to the  port.  The coworker’s telephone rings.  When the  coworker picks  up the  receiver or answers the call, a virtual connection is established between the  coworker and yourself for the  life of the  call. IP telephony does all this  at lightning speed.

Getting Down to Business with VoIP (Chapter 1)

Technological innovation is hurling itself upon us once again.  This time, it’s coming in the  form of improving the  way we communicate, bringingtelephone  call. with it new capabilities that change the  meaning of the  phrase VoIP (often pronounced “voyp”)  is the  name of this  new communications technology.

MSF VoIP Work Plan (part 6)

 The MSF is committed to an aggressive technical program to specify and prove a solution for a full scale PSTN replacement network over next generation IP infrastructure. This technical program will attack head on the major issues that have so far prevented the next generation network (NGN) from being fit for purpose, specifically it aims to provide for a network that is capable of offering Quality of Service (QoS) and security in a way that will scale to the many billions of busy hour calls that a typical PSTN must handle.  The MSF work program will follow the established and proven MSF approach to problem solving by: 

Fax, Modem and TTY support. (part 5)

 The PSTN reliably supports fax, modem and TTY calls. Calls connect on almost every attempt and rarely fail. A VoIP network must provide a similarly reliable service. However, fax, modem and TTY traffic imposes some additional constraints beyond voice traffic.  Compared to voice traffic, fax, modem and TTY traffic is much more sensitive to packet loss but less sensitive to overall delay. In addition, lower-bit-rate codecs are optimized for voice traffic and cannot transport non-voice traffic. 

Migration Path (part 5)

While the eventual goal is an end-to-end Next-Generation Network, it will be decades before legacy networks disappear. On the access side, this means that ongoing support for POTS telephone lines and DLCs may be a requirement; in the backbone network, interconnection with SS7 signaling and TDM trunks, 911 and operator services, databases for 1-800 and local number portability and CALEA, are all essential.

Firewall and NAT traversal (part 5)

For equipment that is resident at customer premises, such as IP phones and Subscriber Gateways it is likely that there will be a firewall at the edge of the customer premises. In addition, Network Address Translation (NAT) may be used to convert internal IP addresses to external IP addresses.  Therefore it is important that both the RTP media traffic and the signaling flows (SIP, H.248, MGCP) can negotiate both NAT and the firewall. For the firewall to be effective it needs to ensure that only authorized flows enter or leave the networks.  There are working groups within the IETF, including Midcom and NSIS, who are addressing the issue of communications with firewalls and network address translators.

7 Emergency and Operator Services (part 5)

The PSTN supports extensive Emergency and Operator Services. Subscribers can dial 911 or the local equivalent and reach Emergency Services under almost any conditions. A Next-Generation VoIP Network needs to provide similar support leading to the following requirements:  
• Support for legacy Emergency and Operator Services Interfaces, for example MF and SS7.

• Support for lifeline support where this is a regulatory requirement.

• Provision of location information so that a caller’s physical location can be determined.

Reliability / Availability (part 5)

The PSTN achieves five-nines reliability, equivalent to fewer than five minutes per year downtime, and it handles millions of simultaneous calls. A VoIP network needs to achieve similar levels of reliability and scalability.  The required reliability and scalability can be achieved in a VoIP network by using redundant and loadsharing equipment and networks. The call agent, access gateway, trunk gateway, signaling gateway and media server need to be fault tolerant. The types of functionality often used to achieve fault tolerance include: 
• Redundant hardware 

• Redundant network connections

• Hot-swap capability
 • No single point of failure 

• Software and firmware that can be upgraded without loss of service.

Quality of Service (part 5)

One of the key requirements for the widespread deployment of VoIP is the ability to offer a toll quality service equivalent to the existing PSTN. Indeed some carriers are even looking for Next-Generation Networks as a means for delivering much higher voice quality as a service.  Perceived Voice quality is very sensitive to three key performance criteria in a packet network, in particular: 

• Delay
• Jitter 
• Packet loss

Security (part 5)

The PSTN has been very resistant to security attacks and has not suffered from significant problems since the introduction of SS7 out-of-band signaling. A VoIP Next-Generation network is much more susceptible to security attacks and must address three key security issues. 
• Denial of service
• Theft of service
• Invasion of privacy
Security is seen as a priority for MSF, and will be addressed in our 2003 work program. See section 6 for more details.

Issues in a VoIP Network (part 5)

There are several issues that need to be addressed in order to provide a toll-quality, PSTN equivalent end-toend VoIP network. These include: 

• Service set to be offered, and the types of end user terminal supported.
• Choice of signaling protocol(s).
• Security.
• Quality of Service (QoS).
• Reliability / availability.
• Regulatory Issues
• Lawful Interception
• Emergency and Operator Services

Nework Components 3 (part 4)

Signaling Gateway

The Signaling Gateway is located in the service provider’s network and acts as a gateway between the call agent signaling and the SS7-based PSTN. It can also be used as a signaling gateway between different packetbased carrier domains. It may provide signaling translation, for example between SIP and SS7 or simply signaling transport conversion e.g. SS7 over IP to SS7 over TDM.

Nework Components 2 (part 4)

Service Broker

The service broker is located on the edge of the service providers service network and provides the service distribution, coordination, and control between application servers, media servers, call agents, and services that may exist on alternate technologies (i.e. Parlay Gateways and SCP s).   The service broker allows a consistent repeatable approach for controlling applications in conjunction with their service data and media resources to enable services to allow services to be reused with other services to create new value added services.

Network Components (part 4)

This section describes the function of the network components listed in Figure 1. Depending upon the particular network architecture some of these network components may be combined into a single solution, for example a combined signaling and trunking gateway.

Call Agent/SIP Server/SIP Client 

The Call Agent/SIP Server/SIP Client is located in the service provider’s network and provides call logic and call control functions, typically maintaining call state for every call in the network. Many call agents include service logic for supplementary services, e.g. Caller ID, Call Waiting, and

VoIP Next-Generation Network Architecture (part 3)

VoIP can be deployed in many different network segments. To date, it has been mostly deployed in the backbone and enterprise networks. Deploying VoIP as an end-to-end Next-Generation Network solution introduces additional constraints and issues discussed in section 5.  

Key Benefits and Requirements for VoIP (part 2)

 For service providers examining the business case for VoIP, the ubiquity of IP as a networking technology at the customer premises opens the possibility of deploying a vast range of innovative converged voice and data services that simply cannot be cost effectively supported over today’s PSTN infrastructure. 

• IP-based internet applications, such as email and unified messaging, may be seamlessly integrated with voice applications  

Overview of VoIP (part 1)

 At its simplest, Voice over IP is the transport of voice using the Internet Protocol (IP), however this broad term hides a multitude of deployments and functionality and it is useful to look in more detail at what VoIP is being used for today. Currently the following types of VoIP applications are in use: 

• Private users who are using voice over IP for end to end phone calls over the public internet. These users typically trade quality, features and reliability for the fact that the service is very low cost and are generally happy with the service. Although globally the numbers of users taking advantage of this technology is large the density of such users is very low and when compared with the PSTN the call volumes are negligible.