Week Twelve Agenda

2. Week Twelve Agenda Attendance Announcements December 1, lab classroom in Phillips Hall 222 has been requested, but not official. Mimic Simulator Lab Assignment 4-1-3 Review Week Eleven Information Current Week Information Upcoming Assignments

3. Week Eleven Topics Review Week Ten Information • Analog to digital signaling • PBX and PSTN • Definitions • Trunk capacity Current Week Information • VoIP • Codec • WLAN

4. Analog and Digital Signaling • The human voice generates sound waves • The telephone converts the sound waves into an analog signal. • To obtain clear voice connections, the PSTN switches convert analog speech to a digital format and send it over the digital network. • At the other end of the connection, the digital signal is converted back to analog and to the normal sound waves that the ear can hear. • Digital signals don’t pick up the noise levels as analog signals, and doesn’t induce any additional noise when amplifiing signals. • Digital signals hold their original form better than analog signals over greater distances, regeneration, coded, and decoded translations.

5. Analog and Digital Signaling The human range for speech is approximately 400 to 4000 hertz (hz). Higher frequencies are filtered. Sampling is the method used on analog signals to formalize the digitizing process. A voltage level corresponds to the amplitude of the signal.

6. Analog and Digital Signaling Pulse Code Modulation (PCM) is a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a numeric (usually binary) code. The standard code word size is 8 bits.

7. Analog and Digital Signaling There are several steps involved in converting an analog signal into PCM digital format, as shown in the figure

8. Companding • Signal is compressed for more efficient transmission, and less noise • Two common methods: The A-law standard is used in Europe, Mu-law is used in North America and Japan • The methods are similar—but they are not compatible

9. Analog and Digital Signaling • Filter analog signal – remove frequencies > 4000 hertz • Sample – rate at least twice the highest frequency according to Nyquist Theorem. Samples the filtered input signal at a constant frequency using Pulse Amplitude Modulation (PAM). • Digitize – occurs prior to transmission over the telephone network (PCM process)

10. Analog and Digital Signaling 4. Quantization and coding – A process that converts each analog sample value into a discrete value to which a unique digital code word is assigned. 5. Companding – A process in which compression is followed by expansion; often used for noise reduction in equipment, in which case compression is applied before noise exposure and expansion after exposure. A process in which the dynamic range of a signal is reduced for recording purposes and then expanded to its original value for reproduction or playback.

11. Companding • A signal is compressed for more efficient transmission, and less noise • Two common methods: The A-law standard is used in Europe, Mu-law is used in North America and Japan • The methods are similar—but they are not compatible

12. Public Switched Telephone Network (PSTN) • Telephones connect to a CO (Central Office) through the local loop • The local loop is an analog connection • All analog signals are converted to digital at the CO • Except for the local loop the entire phone system is a modern digital network

13. Public Switched Telephone Network (PSTN)

14. Trunk Lines Trunk Lines carry traffic between Central Offices Each trunk line carries many simultaneous conversations This is accomplished through Time Division Multiplexing

15. Time Division Multiplexing

16. What is a Private Branch Exchange (PBX)? PBX is a private telephone network used within a company. The users of the PBX phone system share a number of outside lines for making external phone calls. A PBX connects the internal telephones within a business and also connects them to the public switched telephone network (PSTN).

17. PBX Features • A PBX is a business telephone system that provides business features such as call hold, call transfer, call forward, follow-me, call park, conference calls, music on hold, call history, and voice mail. • Most of these features are not available in traditional PSTN switches. • A PBX switch often connects to the PSTN through one or more T1 digital circuits. • A PBX supports end-to-end digital transmission, employs PCM switching technology, and supports both analog and digital proprietary telephones

18. PBX Features Analog PBXs were phased out about twenty years ago. Today, most all vendors manufacture digital PBXs.

19. PBXs and PSTN Switches

20. PBXs and PSTN Switches

21. Trunk Line Capacity In this diagram, 7 telephones connect to the CO in Neighborhood A and 6 connect to the CO in Neighborhood B How many simultaneous conversations should this trunk line carry?

22. Trunk Line Capacity The science of Traffic Engineering answers this question

23. What is Traffic Engineering? • Voice traffic engineering is the science of selecting the correct number of lines and the proper types of service to accommodate users. • Detailed capacity planning of all network resources should be considered to minimize degraded voice service in integrated networks. • We can calculate the bandwidth required to support a number of voice calls with a given probability that the call will go through

24. Terminology • Blocking probability • Grade of Service (GoS) • Erlang • Centum Call Second (CCS) • Busy hour • Busy Hour Traffic (BHT) • Call Detail Record (CDR)

25. Definitions • The blockingprobability value describes the calls that cannot be completed because insufficient lines have been provided. For example, a blocking probability value of 0.01 means that 1 percent of calls would be blocked. • GoS is the probability that a voice gateway will block a call while attempting to allocate circuits during the busiest hour. GoS is written as a blocking factor, Pxx, where xx is the percentage of calls that are blocked for a traffic system. For example, traffic facilities that require P01 GoS define a 1 percent probability of callers being blocked.

26. Definitions • One Erlang equals one full hour, or 3600 seconds, of telephone conversation • The busy hour is the 60-minute period in a given 24-hour period during which the maximum total traffic load occurs. The busy hour is sometimes called the peak hour. • The BHT, in Erlang’s or CCSs, is the number of hours of traffic transported across a trunk group during the busy hour (the busiest hour of operation). • A CDR is a record containing information about recent system usage, such as the identities of sources (points of origin), the identities of destinations (endpoints), the duration of each call, etc

27. Trunk Capacity Calculation • For example, one hour of conversation (one Erlang might be ten 6-minute calls or 15 4-minute calls. Receiving 100 calls, with an average length of 6 minutes, in one hour is equivalent to ten Erlangs • For example, if you know from your call logger that 350 calls are made on a trunk group in the busiest hour and that the average call duration is 180 seconds, you can calculate the BHT as follows: • BHT = Average call duration (seconds) * calls per hour/3600 • BHT = 180 * 350/3600 • BHT = 17.5 Erlangs

28. Capacity Information • There are years of data on the number and duration of a phone conversation • This historical data can be used to calculate the capacity or number of trunk lines needed in a telephone system • Erlang Tables are used for this calculation

29. What is an Erlang Table? • Erlang tables show the amount of traffic potential (the BHT) for specified numbers of circuits for given probabilities of receiving a busy signal (the GoS) • The BHT calculation results are stated in Erlangs • Erlang tables combine offered traffic (the BHT), number of circuits, and GoS in the following traffic models:

30. What is an Erlang Table? • Erlang B: This is the most common traffic model, which is used to calculate how many lines are required if the traffic (in Erlangs) during the busiest hour is known. The model assumes that all blocked calls are cleared immediately. • Extended Erlang B: This model is similar to ErlangB, but it takes into account the additional traffic load caused by blocked callers who immediately try to call again. The retry percentage can be specified. • Erlang C: This model assumes that all blocked calls stay in the system until they can be handled. This model can be applied to the design of call center staffing arrangements in which calls that cannot be answered immediately enter a queue

31. What is an Erlang Table? • Erlang C: This model assumes that all blocked calls stay in the system until they can be handled. This model can be applied to the design of call center staffing arrangements in which calls that cannot be answered immediately enter a queue

32. Trunk Capacity Calculation • The network design is based on a star topology that connects each branch office directly to the main office. • There are approximately 15 people per branch office. • The bidirectional voice and fax call volume totals about 2.5 hours per person per day (in each branch office). • Approximately 20 percent of the total call volume is between the headquarters and each branch office. • The busy-hour loading factor is 17 percent. In other words, the BHT is 17% of the total traffic. • One 64-kbps circuit supports one call. • The acceptable GoS is P05

33. Trunk Capacity Calculation • 2.5 hours call volume per user per day * 15 users = 37.5 hours daily call volume per office • 37.5 hours * 17 percent (busy-hour load) = 6.375 hours of traffic in the busy hour • 6.375 hours * 60 minutes per hour = 382.5 minutes of traffic per busy hour • 382.5 minutes per busy hour * 1 Erlang/60 minutes per busy hour = 6.375 Erlangs • 6.375 Erlangs* 20 percent of traffic to headquarters = 1.275 Erlangs volume proposed

34. Final Calculation • To determine the appropriate number of trunks required to transport the traffic, the next step is to consult the Erlangtable, given the desired GoS • This organization chose a P05 GoS. Using the 1.275 Erlangsand GoS= P05, as well as the ErlangB table: http://www.erlang.com/calculator/erlb/ • four circuits are required for communication between each branch office and the headquarters office

35. What do the terms FXS and FXO mean? FXS and FXO are the name of ports used by Analog phone lines (also known as POTS -Plain Old Telephone Service) or phones. FXS -Foreign eXchange Subscriber interface is the port that actually delivers the analog line to the subscriber. In other words it is the ‘plug on the wall’ that delivers a dial tone, battery current and ring voltage.

36. What do the terms FXS and FXO mean? FXO -Foreign eXchange Office interface is the port that receives the analog line. It is the plug on the phone or fax machine, or the plug(s) on your analog phone system. It delivers an on-hook/off-hook indication (loop closure). Since the FXO port is attached to a device, such as a fax or phone, the device is often called the ‘FXO device’. FXO and FXS are always paired, i.e similar to a male / female plug. Without a PBX, a phone is connected directly to the FXS port provided by a telephone company

37. FXS and FXO

38. Connecting a Traditional PBX to the PSTN • If you have a PBX, then you connect the lines provided by the telephone company to the PBX and then the phones to the PBX. • Therefore, the PBX must have both FXO ports (to connect to the FXS ports provided by the telephone company) and FXS ports (to connect the phone or fax devices to).

39. Connecting a Traditional PBX to the PSTN

40. Telephone Signaling In a telephony system, a signaling mechanism is required for establishing and disconnecting telephone communications.

41. Three Types of Signaling Used To Make a Phone Call • Supervision signaling: Typically characterized as on-hook, off-hook, and ringing, supervision signaling alerts the CO switch to the state of the telephone on each local loop. Supervision signaling is used, for example, to initiate a telephone call request on a line or trunk and to hold or release an established connection. • Address signaling: Used to pass dialed digits (pulse or DTMF) to a PBX or PSTN switch. These dialed digits provide the switch with a connection path to another telephone or customer premises equipment. • Informational signaling: Includes dial tone, busy tone, reorder tone, and tones indicating that a receiver is off-hook or that no such number exists, such as those used with call progress indicators

42. Analog Telephony Signaling • Loop start: Loop start is the simplest and least intelligent signaling protocol, and the most common form of local-loop signaling. Only for residential use. • Ground start: Also called reverse battery, ground start is a modification of loop start that provides positive recognition of connects and disconnects (off-hook and on-hook)., PBXs typically use this type of signaling. • E&M: E&M is a common trunk signaling technique used between PBXs.

43. Digital Telephone Signaling • CAS • CCS • DPNSS • ISDN • QSIG Digital Signaling –standards based protocol to allow different vendor’s PBXs to communicate • SS7 Digital Signaling -used within the PSTN for signaling between PSTN switches

44. Traditional Voice and Data Networks

45. Integrated Voice and Data Networks

46. Why Integrate Voice and Data Networks? • Integrating data, voice, and video in a network enables vendors to introduce new features • The unified communications network model enables distributed call routing, control, and application functions based on industry standards • Enterprises can mix and match equipment from multiple vendors and geographically deploy these systems wherever they are needed • Only one network to maintain

47. VoIP or IP Telephony? • Cisco distinguishes between the two • Most technical discussions don’t • VoIP –analog phones and/or analog PBXs are still used, but the analog signals are converted to IP packets with a Voice Enabled router • IP Telephony –IP phones are used; the system is completely IP. Specialized call processing software replaces the PBX –this may be called an IP PBX

48. VoIP Connection To setup a VoIP communication we need the do the following: • The ADC (Analog to Digital Converter) converts analog voice to digital signals (bits) • The voice data is compressed to send the fewest number of bits while still retaining the original information (Codec) • Voice packets are sent using a real-time protocol (typically RTP over UDP over IP) • We need a signaling protocol to call users: ITU-T H323 or SIP • At the receiver we have to disassemble packets, extract data, then convert them to analog voice signals and send them to sound card (or phone) • All that must be done in a real time fashion cause we cannot waiting for too long for a vocal answer! (QOS)

49. VoIP Technology • VoIP is an “Overlay” technology • VoIP is applied on top of an IP Network • If the IP network is not working properly VoIP will simply be one more thing that is broken • Make sure the IP network is working correctly FIRST--then implement VoIP

50. VoIP