Conversation Products Catalog
As far as handling more calls with less bandwidth, this might require a few words of explanation. More details on this key VoIP and voice in general concept will be examined later. For now, it is enough to point out that traditional voice digitization techniques require 64 kb s (thousands of bits per second) to function properly. Business access lines are often digital in nature as soon as they leave the customer premises, but residential access lines most often carry analog voice conversation. Voice digitization is the topic of the next chapter, so only the briefest mention of this analog digital voice dichotomy is made here. However, even in the case where the voice line in question leads to a residential user, the analog voice is digitized as soon as it reaches the major PSTN component, the voice switch, and sometimes even before. Voice digitized at 64 kb s is most often called 64-kb s PCM voice or just PCM voice. PCM stands for pulse code modulation. The point is that PCM voice...
Enhanced services are those which add value to the basic voice conversations supported by simple telephony. Since enhanced services add value to the conversation, there is usually an extra charge involved. There are numerous examples, and enhanced services cover a wide range in terms of features and capabilities.
Fax services more closely resemble voice services than data services. That is, with few exceptions, a fax is sent over the same telephone lines as those used for analog voice conversations. Many fax machines have a regular telephone handset attached and can be used to make voice telephone calls. Telephone companies have no easy way, and even few sophisticated ways, to distinguish a call made to send a fax from a call made to speak to someone. A fax call looks just like a voice call to the PSTN.
The OSI reference model partitions the overall communication process into functions that are carried out by various layers. In each layer a process on one machine carries out a conversation with a peer process on the other machine, as shown in Figure 2.3. In OSI terminology the processes at layer n are referred to as layer n entities. Layer n entities communicate by exchanging protocol data units (PDUs). Each PDU contains a header, which contains protocol control information, and usually user information in the form of a service data unit (SDU). The behavior of the layer n entities is governed by a set of rules or conventions called the layer n protocol. In the HTTP example the HTTP client and server applications acted as peer processes. The processes that carry out the transmitter and receiver functions of TCP also constitute peer processes at the layer below.
Satellite microwave systems were introduced in the 1960s and have been used extensively on transoceanic routes. A disadvantage of satellite transmission is the long signal propagation time (in excess of 250 ms from ground station to ground station), which can cause problems in telephone conversations.
Common, everyday telephony never was included in the vast array of content available on the Web. People cruised the Web to find information but still picked up the telephone to have a real conversation. Why not The Web addressed limitations in the way that information was handled and presented by the Internet. There were few perceived limitations in the way that voice was handled and presented by the PSTN.
Home networks carry phone conversations, TV programs and MP3 music programs, link computers and peripherals, electronic mail (e-mail), distribute data and entertainment programs, Internet access, remote interactive services and control of home appliances, lights, temperature and so forth. The most important remote interactive services include remote metering, home shopping, medical support, financial transactions, interactive TV, video telephony, online games, voice-over Internet Protocol (VoIP) and so forth. Home applications based on multimedia require Internet connections and higher data transfer rates. For example, video programs compressed to MPEG-2 standards require a 2-4 Mbps transfer rate DVD
The concept of an integrated network that provides the user with a variety of services over a single interface has a number of advantages for both the users and the operators. This is well reflected in the developments of both circuit switched narrowband ISDN (N-ISDN) and further packet switched broadband ISDN (B-ISDN) networks. However, a homogeneous, 'end-to-end Asynchronous Transfer Mode (ATM)' concept of a B-ISDN had some obvious difficulties in taking off, and the Internet as a heterogeneous collection of subnetworks has become increasingly dominant. The Internet provides an abundance of bandwidth owing to its core optical fiber network. This factor makes a good case for an integrated network. However, the inherent design principles of the Internet provide only a 'best effort' service that is typically not optimized for conversational services. To provide a variety of services over the Internet, a few improvements in traffic management and handling within the network are...
Internet Relay Chat (IRC) is sort of like Citizens Band (CB) radio on the Internet it has its own little culture involving lots of people talking at each other. Users access IRC via dedicated IRC clients, or by using Telnet to access a site that provides public IRC client service. IRC servers provide hundreds (sometimes thousands) of named channels for users to join. These channels come and go (anyone can create a new channel, and a channel survives as long as there's anyone on it), although some popular channels are more or less permanent. Unlike talk, which is limited to a pair of users, any number of people can participate on an IRC channel simultaneously. Some IRC clients allow a user to participate in multiple channels simultaneously (sort of like taking part in two different conversations at once at a party). More complicated systems allow richer conversations. As high-speed network connections become common, full-fledged video conferencing systems have become popular, even...
When is registered for three-way calling service, he can call S2 and then add a call to S3. Having established the call to S2, subscriber S, flashes. The local exchange returns dial tone, and Sj enters S3's number. The exchange then sets up the call to S3 while maintaining the call to S2. If S3 is busy or does not answer, Sj can end the new call (but remain in conversation with S2), by flashing twice.
The telephone operates in a full-duplex fashion, meaning both people can talk at the same time. Electricity had no problem traveling in both directions at the same time in these early analog systems, and the electrical waveforms just looked different when two people were speaking. The talker heard his or her own words over the telephone line, in a form of echo, but this turned out to be all right. When the receiver was placed over one ear, half of a person's hearing was impaired. The person reacted by speaking louder to compensate, and early telephone conversations quickly turned into shouting matches (which distorts what is said). The speaker echo in early telephone systems provided a cure for this hearing loss, and as long as the echo was not noticeably delayed compared with the speaker's voice, which could be very annoying, everything was fine. Later systems added circuitry to try to eliminate echo (analog echo suppressors and later digital echo cancelers) and added sidetone to the...
A dramatic change happened concerning the validity of these laws when telephone networks were used not only for voice conversations but also for FAX transmissions and Internet access. The statistical characteristics of these services are significantly different from voice calls. Especially, the call durations become much longer and more variable compared to classical voice calls. As the popularity of the Internet increased due to the success of Web, more and more people started to use the classical telephone networks for Internet access. These changes call for reviewing the old laws and present a challenge for today's teletraffic researchers.
If an attacker is somewhere on the network between the destination and the forged source, the attacker may be able to see the reply and carry on a conversation indefinitely. This is the basis of hijacking attacks, which are discussed in more detail later. Figure 4.6 shows an attacker using a forgery this way.
On digital links, however, silence is just coded up and sent as a string of 0s and 1s, just like active voice. Thus the silence power level is the same as the active speech power level when digital voice is used. This means that noise has less of a chance of disrupting the conversation, even at relatively high levels. More will be said later about silence suppression, which eliminates periods of silence from the bit stream. For now, it is assumed that every moment of a conversation is expressed as a stream of bits, even extended periods of silence at one or both ends of the conversation.
Carrier and radio systems require that oppositely directed portions of a single conversation occur over separate transmission channels or paths (or use mutually exclusive time periods). Thus we have two wires for the transmit path and two wires for the receive path, or a total of four wires for a full-duplex (two-way) telephone conversation. For almost all telephone systems, the end instrument (i.e., the telephone subset) is connected to its local serving exchange on a two-wire basis. In other words, the subscriber loop is two-wire.
Now, that statement may seem to be rather trivial and nonessential when you think about it. Of course the warehouse should only be built with the needs of the business in mind And of course a successful project will be one that has the business' needs in mind before you start building it. But how many books or lectures about data warehousing and data mining have you seen where this is the first and most important part of the conversation Hardly any.
Conversely, it also won't do you any good to block only half a connection. Many attacks can be carried out if attackers can get packets into your network, even if the attackers can't get any responses back. This can be possible for several reasons. For instance, attackers may only be interested in issuing a particular command which does not require a response (like shut down your network interface for a denial of service attack, using an SNMP set command). Or, the responses may be predictable enough to allow attackers to carry on their side of the conversation without having to actually see the responses at all.
The obvious starting point is the telephone, which is now a digital telephone. Instead of the telephone conversation being analog from the handset to the central office where it becomes digitized, the conversation can be digitized directly at the source and passed digitally all the way through the network to the other end.
In many circumstances - particularly those involving TCP connections - the real machine (that the attacker is pretending to be) will react to your packets (packets that are attempting to carry on a conversation it knows nothing about) by trying to reset the bogus connection. Obviously, the attacker doesn't want this to happen. Therefore, the attack must complete before the real machine gets the packets you're sending, or before you get the reset packets from the real machine. There are a number of ways to ensure this - for example The man in the middle forgery attack depends on being able to carry out a complete conversation while claiming to be the trusted host. In order to do this, the attacking machine needs to be able to not only send you packets, but also intercept the packets you reply with. To do this, the attacker needs to do one of the following
In the first part of this chapter, we explained how TDM could replace multiple physical lines by a single high-speed line. In TDM a slot within a frame corresponds to a single connection. The time-slot interchange (TSI) technique replaces the crosspoints in a space switch with the reading and writing of a slot into a memory. Suppose we have a number of pairs of speakers in conversation. The speech of each speaker is digitized to produce a sequence of 8000 bytes second. Suppose that the bytes from all the speakers are placed into a T-1 carrier, as shown in Figure 4.25. Suppose also that the first pair of speakers has been assigned slots 1 and 23. For the speakers to hear each other, we need to route slots 1 and 23 in the incoming frames to slots 23 and 1 in the outgoing frames. Similarly, if the second pair of speakers is assigned slots 2 and 24, we need to interchange incoming slots 2 and 4 with outgoing slots 24 and 2, respectively.
In North America each telephone is given a 10-digit number according to the North American numbering plan. The first three digits are the area code, the next three digits are the central office code, and the last four digits are the station number. The first six digits of the telephone number uniquely identify the central office of the destination telephone. To set up a call, the source switch uses the telephone signaling network to find a route and allocate resources across the network to the destination office.7 This step is indicated by the propagation of signaling messages in Figure 4.31. The destination office next alerts the destination user of an incoming call by ringing the phone. This ring is sent back to the source telephone, and conversation can begin when the destination phone is picked up. The call is now in the message transfer phase. The call is terminated and the resources released when the users hang up their telephones.
At this point the caller may receive call progress tones that formerly represented analog signaling tones on a trunk but now are just used to let the caller know that things are happening even though the telephone at the other end is not ringing yet. If there is no trunk available to the destination switch, which should happen only under extremely congested conditions, then the caller will receive a fast busy tone, technically called reorder. Most callers know that this fast busy means to hang up and try the call again later. Of course, the remote switch must check the destination local loop to see if a call is in progress on the destination loop. If there is already a conversation taking place, then a busy signal is given to the caller. Sometimes a congested switch may give a false busy just to get callers to try a call again later, presumably when there are fewer calls to handle.
International exchanges are also used to connect to other countries. To be able to conduct a satisfactory telephone conversation to another country it is important that both of the telephone networks comply with the appropriate technical recommendations of the ITU. Since half the telephone call is handled in the originating country and the other half in the destination country, satisfactory performance depends on both parties complying with the relevant recommendations. The most important of these cover the transmission performance and the signaling conventions.
The demonstration which carried the fame of the telephone far and wide in the United States took place at the crowded Lyceum Hall in Salem on 12 February 1877. Bell had one of the telephones in the lecture hall whereas the other one was in the hands of Watson, his assistant, 18 miles away in Boston. There was no need to lay a wire. In contrast to the telegraph, which had to be built from scratch, the telephone found a ready infrastructure. Bell and Watson could use the existing telegraph line between Boston and Salem. The demonstration started with Watson's rendering of Auld Lang Syne and Yankee Doodle followed by a conversation between Bell and Watson with others butting in. The demonstration was reported by several newspapers over the next few days. The scene was captured for posterity by the Scientific American (see Fig. 5.1). The proprietors of the Telephone, the invention of Alexander Graham Bell, for which patents have been issued in the United States and Great Britain, are now...
In the third party configuration, a call is initiated from a device that does not participate in the call. The third party or switch oriented call control system controls the switch through a separate X.25 connection using a CTI link protocol such as CSTA as shown in Figure 5.3. The call control can be performed by an API such as TS API. The TS API enabled application program is viewed as an external agent (third party) in a conversation between the calling and the called parties.
The terms digital and analogue are no longer reserved for the use of communications experts. They are freely bandied around in everyday conversation. It is easy to explain what they mean. Digital information is in the form of digits, e.g. 5 or 8 or 3.5 or 7.969 75. Some information may only exist in digital form, e.g. the number of houses in a street, the number of bedrooms in a house or the number of jokes told at a party. Examples of analogue information are length, weight, the speed of a vehicle or the current flowing in a circuit. All these, at least in principle, could be measured with arbitrary accuracy. We could for example measure the distance between two points as 12 365 m. A more accurate measurement for the same distance might yield 12 365.13 m. The point is that the distance is expressible with digits but there is no exact digital equivalence. The higher the required accuracy, the greater the number of digits needed.
System designers turned to wideband radio and coaxial cable systems where each bearer or pipe carried hundreds or thousands of simultaneous telephone conversations.12 Carrier (frequency division) multiplex techniques made this possible (see Section 4.5). Frequency division multiplex (FDM) requires separation of transmit and receive voice paths. In other words, the circuit must convert from two-wire to four-wire transmission. Figure 8.17 is a simplified block diagram of a telephone circuit with transformation from two-wire to four-wire operation at one end and conversion back to two-wire operation at the other end. This concept was introduced in Section 4.4.
It's easy for a firewall to intercept the initial connection from a client to a server. It's harder for it to intercept a return connection. With a proxy, either both ends of the conversation have to be aware of the existence of the proxy server, or the server needs to be able to interpret and modify the protocol to make certain the return connection is made correctly and uniquely. With plain packet filtering, the inbound connection has to be permitted all the time, which often will allow attackers access to ports used by other protocols. Stateful packet filtering, like proxying, has to be able to interpret the protocol to figure out where the return connection is going to be and open a hole for it.
An RPC-based client program that wishes to contact a particular RPC-based server on a machine first contacts the location server on that machine (which, remember, always runs on both TCP and UDP port 111 or 135). The client tells the location server the unique RPC service number for the server it wishes to access, and the location server responds with a message saying, in effect, either I'm sorry, but that service isn't available on this machine at the moment , or That service is currently running on TCP (or UDP) port n on this machine at the moment . At that point, the client contacts the server on the port number it got from the location server and continues its conversation directly with the server, without further involvement from the location server. (Figure 14.1 shows this process.)
For example, voice communication is about as flexible as life gets. Not only does the listener not know what is coming next, but sometimes the speaker doesn't either. However, go to a Web site 20 times a day, and not much will change over that period. People offer more sophisticated features than the Web too. People typically know what they do know and do not know. The Web often doesn't. Try a search one way and it comes up empty. Try it another way (perhaps by just changing a letter), and the user in suddenly inundated with information from the same set of Web sites. The telephony concept of collocation is spelled (apparently equally properly) both as colocation and collocation. However, people are usually consistent in their usage. Thus a search on one spelling of the term turns up perhaps 500 Web pages, and a search on the other spelling turns up maybe 500 different Web pages This could never happen in conversation. Ever find something useful on the Web and never, ever be able to...
One benefit implemented in ACDE is rapid development through automated code generation. As well, policy attributes define many different aspects of agent and agent system operation, including deployment, communication and security requirements. While the automatically generated code must be modified to provide application-specific agent functionality, the process effectively speeds agent system development. To simplify agent system deployment, the automatic code generating facility also uses policy attributes to generate a System Deployment Agent. In ACDE, the ASDO contains elements that modulate agent and agent system operation. Authorized agents may alter certain policy attributes modulating system performance in an interactive manner. For example, an agent may turn on or off its requirement for secure communication, based upon the need for private conversations. This approach reduces overall communication latency by not requiring computationally intensive encryption activity for...
Although PPTP is an encrypted protocol, not all the parts of the conversation are encrypted. Before the PPTP server starts accepting the GRE packets, a negotiation takes place over TCP. PPTP encryption protects the information being tunneled but not the negotiation involved in setting up the tunnel. The negotiation is done in cleartext and includes client and server IP addresses, the name and software version information about the client, the username, and sometimes the hashed password used for authentication. All of this information is exposed to eavesdropping. This negotiation is also done before the client has to authenticate, which makes the server particularly vulnerable to hostile clients. An attacker doesn't have to be able to authenticate in order to engage the server in negotiation, tying up resources and potentially confusing the server.
Any of the transactions described above can include one or more pairs of conversation (C) packages. In the example of Fig. 16.3-2, SCP has received a query package with an info_collected message, and returns a conversation package with a send_to_resource_interaction (STR-I) message. After connecting the calling line or incoming trunk to an INSC, playing the specified announcement, and collecting the caller's DTMF digits, SSP returns a conversation package with a resource_clear (RES-CLR) message, which includes the collected digits. Figure 16.3-2 Call-related transaction with caller interaction. Q Query package. C Conversation package. R Response package.
Quality of service (QoS) is defined in ITU-T E.800 as 'The collective effect of service performance, which determines the degree of satisfaction of a user of the service.' Unfortunately this is a very subjective evaluation, but there are several techniques for evaluating QoS such as the conversation opinion test outlined in ITU-T P.800. This test
Electrical noise is caused mainly by the random motion of the electrons. There is no way of persuading electrons to move in a nice uniform manner. In fact, a certain amount of noise is added from each point of the line, and quite a lot of noise from each amplifier. Thus from the moment an analogue voice signal is generated in San Francisco to the moment it is demodulated in London, noise will keep on conspiring to corrupt the signal. Does this mean that a San Francisco-London telephone conversation is bound to be of low quality Well, some lines are better than others but on the whole, yes, the conversation tends to be of low quality. Can it be improved Yes, by using digital techniques any conversation between any two points on Earth may sound as good as a local call. How it is done is the subject of the present chapter.
Conventional voice telephony is transported in a full duplex mode on PSTN circuits optimized for voice. By the full duplex mode we mean that there are actually two circuits, one for send and one for receive, to support a normal telephone conversation between two parties. Today, once we depart the local area, all of these circuits are digital. The descriptive word digital may seem ambiguous to some. Let's say that 10 years ago we looked ahead to our present time. All the circuits would consist of 8-bit words, which represent voltage samples of analog voice conversations in a PCM format. This is often characterized in the literature as G.711 service (i.e., ITU-T Rec. G.711). Data are also commonly transported in 8-bit sets called bytes, but more properly called octets. It is comparatively simple to replace 8-bit voltage samples of voice with 8-bit octets of data. However, there remained essential philosophical differences between voice in 8-bit octets and data transmission. A voice...
ANSI does not have separate DEs for originating and destination transaction IDs. In query and response packages, the transaction-ID (TID) value field holds one 32-bit integer, which represents the originating or destination TID value. In conversation packages, the TID field holds two 32-bit integers. The first integer represents the originating TID, and the second one represents the destination TID.
Conventional voice telephony is transported in a full duplex mode on PSTN circuits optimized for voice. By the full-duplex mode we mean that there are actually two circuits, one for send and one for receive to support a normal telephone conversation between two parties. With a few exceptions in the local area, all these circuits are digital. The descriptive word digital may seem ambiguous to some. When we say digital in this context, we mean that all circuits would carry 8-bit words (timeslots), where each word represents an 8-bit voltage sample ostensibly of an analog voice conversation in a PCM format. This is often characterized in the literature as G.711 service (i.e., ITU-T Rec. G.711). Data are also commonly transported in 8-bit sets called bytes, but more properly called octets from our vantage point. It is comparatively easy to replace 8-bit voltage samples of voice with 8-bit octets of data. is set up by a signaling routine. The distant subscriber has a telephone address...
The telephone made great strides ahead in a very short time. By 1911, as mentioned in Chapter 5, it was possible to make a call from New York to Denver, 4300 km away, but obviously that was not far enough. Ambitions were much higher firstly, to span the continent from New York to San Francisco, secondly to establish telephone connections between the US and Europe, and finally to enable people to conduct conversations between any two points on Earth. What were the difficulties in achieving these aims There were two main difficulties attenuation (the telephone signals declining with distance) and the need to have a pair of wires for each conversation. The first one was a technological difficulty. In order to overcome that, amplifiers had to be invented. The second problem was perceived as an economic one. At what stage does it become unprofitable to erect additional lines The requirement of a separate line for each conversation was a serious impediment to inter-urban communications. It...
In case (b), SSP finds that the line is not in the specified state. It then returns a conversation package with a monitor_success message, and keeps monitoring the line. When the line changes to the specified state, SSP ends the transaction, sending a response package with a status_reported message.
An MGC can control numerous gateways. However, to improve reliability and availability several MGCs may be employed in separate locations with function duplication on the gateways they control. Thus if one MGC fails others can take over its functions. We must keep in mind that the basic topic of Section 4 of this chapter is signaling. That is the establishing (setting up) of telephone connectivities, maintaining that connectivity and the taking down of the circuit when the users are finished with conversation. There is a basic discussion of signaling in Chapter 4 of this text.
The main problem with this methodology is that the number of calls in a system simulation is typically several thousand for each operating condition and each call is typically long enough to include two or more segments. The large number of segments that would need to be judged makes it impractical to do a listening test based on all calls. Furthermore, listening-only tests do not evaluate the delay component of a conversation and it would therefore be quite easy to get good scores simply by applying a longer jitter buffering time. To properly evaluate both the quality of the sound and the impairments introduced by long delay one would have to do a conversation test 100 . In a conversation test, two persons are involved in a conversation and judge the quality in real time. With this methodology, one would have to use real-time VoIP applications to encode the speech, send the speech packets through the system simulator to another VoIP application which then decodes the received packets...
A new form, SPam over Internet Telephony (SPIT), is emerging and can be practiced in different forms. It could for example be automated advertising voice messages or words injected into an ongoing conversation. SPIT may be experienced as more intrusive than email spam, as SPIT disturbs the receiver in real time.
Unified messaging (UM) combines voice mail, fax and e-mail into a single application for storing and retrieving an entire suite of message types (for example, WAV files for voice mail and TIFF files for fax). Through a common interface, users can access these unified services using PC, phone, or mobile device. Unified communications takes UM services and extends them. Unified messaging is an asynchronous, store-and-forward type of service. A unified communication service integrates UM with real-time VoIP communications and presence capabilities. For example, an e-mail, page, or fax message retrieved via PC or phone can be replied to with a real-time VoIP call to the sending party. Other capabilities such as Internet call waiting and number mailbox consolidation are often included in unified communications offerings. Unified communications as an enhanced service delivers cost-savings, convenience and increased productivity. Service provider infrastructure costs are reduced, as there is...
All packets contain authenticators and integrity checksums. In order to start a conversation with a Kerberos server, you need to have previously arranged to share a secret with the server. This means that it is likely that you will be able to detect attempts by foreign machines to obtain Kerberos credentials. On the other hand, it means that you will need to have a set of processes and procedures so that you can add and remove systems from your Kerberos realms.
One way of getting around the problems with proxying H.323 is to use what the standard calls a Multipoint Control Unit (MCU) and place it in a publicly accessible part of your network. These systems are designed primarily to control many-to-many connections, but they do it by having each person in the conference connect to them. It means that if you put one on a bastion-host network, you can allow both internal and external callers to connect to it, and only to it, and still get conferencing going. If this machine is well configured, it is relatively safe. However, it's not a true proxy. The external users have to be able to connect directly to the multipoint control unit one multipoint control unit will not connect to another. The end result is that two sites that both use this workaround can't talk to each other. It works only if exactly one site in the conversation uses it. Several systems are available that provide this functionality, under various names.
In any real-time service, one of the key service aspects is the interactive quality and the experienced responsiveness of the service. The goal for any real-time communication service is to produce a service experience which gives the users a sense of immediacy and nearness turning a long geographical distance into a short communication path. To what extent a service is able to fulfill that ambition is measured as the conversational quality. Measuring the conversational quality of a service usually requires extensive conversational tests using real users in a controlled environment. There are, however, some objective indications that can be used to get an idea of what conversational quality potential the service will have. These indications are intrinsic media quality and end-to-end delay. Intrinsic media quality denotes the quality produced by the media codecs. It can be measured by subjective listening viewing tests and does not take into account any interactivity factor. End-to-end...
On a trip to Germany a couple of years ago, I sat next to a PictureTel videophone development manager. He had a videophone at work and at home. He wanted not to travel, but rather to video telephone his contacts in Germany. They insisted that he come, so he was on the plane with me. During the course of our conversation, he said that his son would video telephone him in the afternoon at work and ask questions like How are things going, Dad , When will you be home , and Can we play ball tonight I thought to myself at the time this kid lives in the future. He may never know a world without videophones. This is like children born today never knowing what life was like without a planet filled with PCs to serve them.
In Fig. 17.4-l(a), adjacent cells X and Y are controlled by respectively MSC-A and MSC-B. It is assumed that the MS is in a conversation, using voice channel (VC-U) of base station (BS-X). The connection occupies trunk T (to an exchange in the PSTN network), trunk T2, and its associated voice channel (VC-U).
Proceeding message that processing of its connection request has been started, the MSC reserves a channel for the conversation and assigns it to the MS (assign). The connection request is signaled to the remote network exchange through the signaling system SS 7 with the ISDN User Part (ISUP) message iam 7 . When the remote exchange answers (acm), the delivery of the call can be indicated to the mobile station (alert). Finally, when the called partner goes off-hook, the connection can be switched through (connect, ans, connect acknowledge).
Hence the corresponding frequency is 32 kHz. The conclusion is that PCM needs about 8 times higher bandwidth than single sideband amplitude modulation. Is this a high price to pay for reduced noise It all depends on the bandwidth available. If it is plentiful, go for PCM. If bandwidth is at premium (e.g. with microwave links) then it may make sense to stick to old-fashioned analogue signals. It is better to have a noisy telephone conversation than wait for ages for a connection to be established.
In Fig. 19.4-1(a), adjacent cells X and Y are controlled by MSC-A and MSC-B, respectively. It is assumed that the MS is in a conversation, using voice channel (VC-U) of base station (BS-X). The connection occupies trunk Ti (to an exchange in the PSTN network), trunk T2, and its associated voice channel (VC-U).
Working with Terashima from 1993 on the application of HR to education, Tiffin and Rajasingham coined the concept schemata HyperClass, HyperSchool, HyperCollege and HyperUniversity (2001) to describe an educational environment in which physically real students, teachers and subject matter could seamlessly interact with virtual students, teachers and subject matter, and AI and HI could interact in the teaching learning process. What makes this possible is a coaction field, which provides a common site for objects and inhabitants from PR and VR and serves as a workplace or activity area within which they interact (Terashima, 2001, p. 9). Coaction takes place in the context of a specific domain of integrated knowledge. So a coaction field could be a game played between real and virtual people, or a real salesperson selling a car to virtual customers (who and what is real and who and what is virtual depends on the kind of perspective of self that exists in a telephone conversation). A...
The third way to handle this is to use a PC-to-PC connection, which is the worst of the three scenarios. This one introduces the use of the PC substandard technology on both ends of the connection, compounding the problems of echo, delay, clipping, and chipping. This is not the best way to experiment with the technology. Moreover, I have seen people using PC-to-PC communications, whereby the caller places a telephone call over the PSTN to the party desired and tells the recipient to turn on their PC so that they can be called. This is ludicrous If the IP system does not know whether the called party is active and powered up, then you have to place a telephone call to tell the person to turn on their PC. Why not just have the conversation while you have the called party on the line This is just an aberration of the technology, but it highlights the fact that VoIP is not yet ready for prime time. There are several tools that IP telephony can bring to the table, enabling the use of VoIP...
The impact of the delay on the voice service is slightly more controversial than the impact of the frame erasures. In general, very low delay is not perceptible, but there is a limit after which the delay starts to be annoying. In Figure 7.2 we show the delay impairment of the ITU-T E-model 110 . In the ITU-T model the conversational quality is degraded already from about 100 ms, even if this is not visible in the figure. The delay becomes noticeable when the one-way delay exceeds 150-170 ms. When the delay exceeds roughly 280 ms, the quality is no longer good, i.e. the MOS score is no longer above 4. This delay value can then be used as the requirement for the maximum allowed one-way delay. Another way to define a requirement is to use the fact that the MOS scores in most subjective tests have a 95 Figure 7.2 The effect of one-way delay on the conversational quality based on ITU-T E-model 110 . Estimated Conversation Quality, G.711, Bursty errors Estimated Conversation Quality,...
For a local telephone conversation one line is sufficient the voice can be carried in both directions. But if the other caller is far away, say at the other side of the Atlantic, then the line must have amplifiers in it which usually work in one direction only. Therefore two lines are needed one to carry my voice from here to America and the other one to carry my friend's voice from America to here. Now I want to ask a simple question are two lines really necessary Surely, in any telephone conversation only one person talks at a time, so the Oxford to New York line could be used by someone else while I am listening. It is a question of firstly recognizing whether I am talking or not and secondly switching the line to another pair of callers. It is worth doing
(2) To make the system economically feasible, it will have to handle a large number of telephone conversations simultaneously. This means that the number of channels in the base station must be increased from one to many and interference between the channels must be avoided. The MSTO is the nerve center of the cellular system. It controls every parameter of the conversation from when a subscriber requests service to when she terminates the call. When a cellular telephone is turned off, the system has no way to reach the subscriber. If the telephone is on but not yet engaged in a conversation (idle), the When a call comes into the MTSO, it broadcasts the mobile identification number (MIN) of the called mobile (a 10-digit number derived from the telephone number assigned to the device) over all the forward control channels in every base station. The mobile recognizes its MIN and responds by identifying itself via the reverse control channel which it has been monitoring all along. The...
If mail can be sent electronically, why not voice Well, e-mail is sent via computer networks by means of packet switching. That means that the message is likely to arrive in an order different from the one in which it was sent. Do we really want to conduct a telephone conversation when there is a good chance that our voice will arrive all jumbled up Clearly not. Hence we can conclude that packet switching is not the right method for voice transmission. The logic leading to this conclusion is impeccable but happens to be wrong. There are ways of asking for priority treatment. The headings in the packet which carry the information about the address, etc. may also carry a priority label which means 'please, can I jump the queue ' at some or all the nodes. That's great, but what if lots of voice packets queue up They cannot all jump the queue so some of the packets will surely arrive out of order. And what about those packets which get lost altogether As it turns out, neither of these...
3.Once the first call is over, another pop-up window signals the presence of a VoIP call coming in over the Internet. Again, this is useful information because this customer may be looking at the Web page at that very moment. A few clicks of the mouse, and the agent can now see the page (if any) that the customer currently sees and can even put a frame on the customer side of the conversation that includes the agent's name and even photo. If the customer has filled up a shopping cart, the agent sees the contents.
By modern times all the telecommunications cables in the cities had been put under the streets, using a string of manholes for their maintenance and for inserting new cables in the ducts. Unfortunately, after a time the ducts inevitably filled up with cables. The only way to add new capacity was to dig up the city streets and lay new ducts. It was a prospect no telephone company (or, in the UK, the Post Office) contemplated with pleasure. It was too expensive. This is when digitalization came to the rescue. The proposed solution was to convert a single analogue line to a digital line capable of carrying 24 conversations. The question that comes to mind is why resort to digital signals in order to carry multiple channels Surely, a single cable can carry many analogue channels with the aid of frequency division multiplex, a technique known and used since the 1930s (see Chapter 7). Unfortunately, frequency division multiplex needed expensive, high quality cables and expensive, high...
Georganas, 2002).Utilizing advanced interfaces, people could gather and interact in public and private spaces, own and share objects, and spend lots of time online (Adler & Christopher, 2003). Interactive characters in virtual worlds will play supportive and helpful roles interacting with their users or other members of the community through natural forms of conversation and gesture, keeping at the same time track of relationships and preferences in a personalized database which they will be updating constantly (Elliott & Brzezinski, 1998). This functionality can be accomplished with several artificial intelligence techniques, but arguably the most viable means for achieving it is via the application of software agents. Through their learning, autonomy, cooperation, and flexibility capabilities, software agents hold the potential and will eventually become a significant part of every virtual world of 3D representations with which agents can examine, interact, and use.
Leisure time communities are communities of people that are using computers for relaxation, fun and social interaction. They appear, at the third level, in the form of communities of relationship and communities of fantasy. Communities of relationship center on intense personal experiences and generally adhere to masking identities and anonymity (Armstrong & Hagel, 1997). Here, participants discuss the personal issues associated with these experiences and exchange information about support institutions. The Well (http www.well.com), is a pioneering online community of relationship known for engaging conversation and intelligent debate. It regularly features more than 260 conferences ranging
Set up between two human beings is occupied from the beginning to the end of the conversation. If you hang up in the middle you will very likely offend your co-conversationist. Computers are less touchy. They do not mind if the information exchange is interrupted at any time. Whatever they have to say to each other is in the form of digital signals and takes place in small bursts. It is a bit like two people playing chess on the phone. One says 'Knight cs' and then hangs up. The other one might call two minutes later and say 'Queen f2, check.' If establishing the connection takes considerably less time than the interval between the calls, then it is worthwhile hanging up and calling again. Since computers talk to each other in small bursts it makes good sense to cut off the communications after each burst. How efficient would that be It would certainly be more efficient than occupying the line all the time, but still not very efficient if setting up the call takes much longer (say 10...
Web-based channels are used to access an agent via a WWW WAP page, and from specialized ACE applications. These channels use personal, semi-automatic formatting of both contexts and presentation of the data to be sent. To this goal, XSL-T technology (The Extensible Stylesheet Language Family, 2002) was adopted with XSL transformations defined in a personal manner and stored in private agent variables (Rykowski & Juszkiewicz, 2003). In a case of a conversation with a human, automatic detection of an end-user device may be used, thus restricting the communication. For example, a small textual message is sent to a mobile phone using WAP connection a similar message with the same contents but with some additional formatting is sent to a PDA device and a full text and graphic message is sent to a stationary PC.
Sometime during the first 30 s or so of the pitch, the telemarketer has to make a decision about where to go next in the script. The conversation can take a nasty turn, so there are menu picks for hostile and silent as well as wants to hear more. Based on the agent's choice, the call can now branch off in a number of different directions, sometimes even adjusting the product offered. For example, a remark about a small kitchen can lead to home improvements instead of the pitch about roofing that was intended. Naturally, the sponsoring company must be able to offer this variety of services.
The range of the cordless telephone is limited to less than 200 m because its radiated RF power is limited to less than 0.75 mW The distance limitation is not necessarily a disadvantage since the frequencies on which it operates can be reused by other cordless telephone users outside the 200 meter radius with minimal or no interference. The real disadvantage of the system is that, because it uses a radio link with no attempts at encryption, anyone with an FM radio capable of operating in the frequency bands assigned to cordless telephones can tune in and listen to the conversation. As mentioned earlier a new frequency band at 2.4 GHz has been assigned to cordless telephones in North America. These cordless telephones are digital, have a longer range (400 m), and use Spread Spectrum technology to ensure that private conversations remain private.
For a voice user, handoff results in an audible click interrupting the conversation for each handoff, and because of handoff, data users may lose packets and unnecessary congestion control measures may degrade the signal level however, it is a random process, and simple decision mechanisms such as those based on signal strength measurements result in the ping-pong effect. The ping-pong effect refers to several handoffs that occur back and forth between two BSs. This takes a severe toll on both the user's quality perception and the network load. One way of eliminating the ping-pong effect is to persist with a BS for as long as possible. However, if handoff is delayed, weak signal reception persists unnecessarily, resulting in lower voice quality, increasing the probability of call drops and or degradation of quality of service (QoS). Consequently, more complex algorithms are needed to decide on the optimal time for handoff.
Gaining access to private telephone conversations is relatively easy. Wiretapping extends beyond copper wire phone line interception to include all types of communications media. All communications technologies including satellite, microwave, infrared, and even fiber optics can be intercepted. Wiretapping and other electronic equipment needed to intercept and interpret telephone conversations is inexpensive. The technical knowledge is provided by electronics hobbyist organizations and Internet publications. There are few if any cost-effective, commercially available products to detect electronic eavesdropping on satellite communications links.
This enhanced service allows subscribers to record voice conversations, whether they are point-point calls or conference calls. Ability to record a call is given to the call initiator or conference host and is available via both phone and PC. Playback of the voice session is accessible by any phone and PC user, if given access to the recording by the initiator. Record and playback, especially as an option for conferencing services, provides participants the ability to review previous discussions. For online education, it gives students who have missed a session an ability to listen at a more convenient time. For call centers, record and playback capabilities can ensure agent service quality or accuracy of customer transactions.
NetMeeting provides a typical set of videoconferencing, or video teleconferencing, features. Such features permit using a PC and the Internet to hold face-to-face conversations with friends and family, or to collaborate with co-workers around the world. Such NetMeeting features include Chat Chat supports real-time text conversations with many people simultaneously. Text messages are typed to communicate with other people in a chat room, either one person or a group of people. A whisper mode sends private messages to another person during a group chat session. Chat session contents can be saved for future reference.
It would theoretically be possible to write one. Because talk involves internal and external clients simultaneously, it would almost have to be a modified-procedure proxy server. (No generic server would handle it, in any case, because it involves both TCP and UDP.) Given the considerable difficulty of writing a talk proxy, and the extreme fragility of the process, it's unlikely that one will become available. talk has been almost altogether abandoned for cross-Internet conversations, in favor of things like IRC and ICQ, described previously.
Normally, ICQ clients will attempt to send messages directly to each other. If you are using a proxy server incoming connections will presumably fail, even when outgoing ones succeed, since the initiating client doesn't know that it should contact the proxy server. Therefore, if you tell your ICQ client that you are using a proxy server, it will route conversations through the ICQ server (via the proxy server) instead of directly to the other client.
When a larger portion of users use a comparatively smaller number of available channels, there is always a finite probability that the number of conversations requiring service will exceed the number of channels providing that service. This competition causes an impairment called competitive clipping,'' where the initial portion of a speech spurt is clipped. The percentage of time that speech is lost due to such competition is called percentage of freeze-out or freeze-out fraction. In the design of a TASI or DSI system, the fraction of speech lost must be acceptably small. Reference 6 gives a freeze-out fraction of 0.5 for TASI systems.
Could we combine the cellular system as it exists today with the services which can be provided by geostationary satellites Not without difficulties. Those satellites are too far away. Consequently, to make contact the mobile phones must be provided with a lot of power, which means lots of batteries which means a lot of weight and lack of portability. If the aim is easy access then the solution is to have orbits nearer to Earth. The official jargon calls them Low Earth Orbits. The satellites of the planned Iridium system3 which will orbit the Earth at a height of only 780 km fall into this category. They are about fifty times closer to Earth than geostationary satellites, which means that they can be accessed with 2500 times less power and that makes a light hand-held set a practical possibility. It will use Time Division Multiple Access with an average power of about 0.34 watt.4 If everything goes well by the turn of the millennium, or perhaps even earlier, conversations could be...
An electronic switching system went into use at the Brown engineering company at cocoa beach, florida in November 1963. It was a small installation, serving only a single company. On May 1965, first commercial electric central office was put into service at succasunna, new jercy. It served only 200 subscribers initially. It had some features of speed, provision for three party conversations and automatic transfer of incoming calls.
On the other hand, long delay is undesirable for real-time services because it is hard to have a fluent conversation if the users have to wait for a long time for the response from the other user. Long delays thus reduce the conversational quality see for example 109 . The delay is thereby a compromise between channel coding performance and perceived conversational quality.
For video conversation in 3G mobile devices, H.263 120 and MPEG-4 Visual 93 are the most important video coding standards. They are used in 3G-324M, which is the mobile circuit switched video telephony standard defined by 3GPP in 6 and 7 based on the ITU-T Recommendation H.324 119 for video conferencing. The baseline version of H.263 was approved in 1995 and has been adopted by 3GPP as the common mandatory video codec that all video-enabled 3G mobile phones must support. There are two extended versions of H.263 published in 1998 (Version 2) and 2000 (Version 3), respectively. All versions are called H.263, although the 1998 and 2000 versions are also referred to as H.263+ and H.263++, respectively. H.263 targets video communication at low bit rates up to 64 kbps but there is no upper limit and it works well up to 1 Mbps or even higher.
Microwave signals propagate in one direction at a time, which means that two frequencies are necessary for two-way communication such as a telephone conversation. One frequency is reserved for transmission in one direction and the other for transmission in the other. Each frequency requires its own transmitter and receiver. Today, both pieces of equipment usually are combined in a single piece of equipment called a transceiver, which allows a single antenna to serve both frequencies and functions.
But there are differences compared to PoC, one main difference being that IM does not use anything similar to the talk burst control of PoC to control the media transfer. Therefore, the controlling IM function can be optional in one-to-one IM sessions. Further, the IM Server can implement two other roles called the deferred messaging IM function and the conversation history function. The conversation history function, as the deferred messaging IM function, stores IM messages in the network. The difference is that the conversation history function stores messages from an active IM conversation that an IM user participates in. The IM server performs the conversation history function when an IM user requests the storage of an IM conversation. This can be done by an IM service setting to always store conversations for the IM user or an IM client initiated request to store a particular conversation. It should be noted that it is only the IM user that requested the storage of the...
An MSC has two more interfaces besides the A and B interfaces, namely the C and E interfaces. Charging information can be sent over the C interface to the HLR. Besides this, the MSC must be able to request routing information from the HLR during call setup, for calls from the mobile network as well as for calls from the fixed network. In the case of a call from the fixed network, if the fixed network's switch cannot interrogate the HLR directly, initially it routes the call to a gateway MSC (GMSC), which then interrogates the HLR. If the mobile subscriber changes during a conversation from one MSC area to another, a handover needs to be performed between these two MSCs, which occurs across the E interface.
The Ericsson WeShare solution is a multimedia person-to-person service that provides the end-users with an instant way of sharing an image, video stream or some other type of stored media while speaking in a CS telephony call conversation. Two users involved in a voice conversation can at any time simply add an image, add a live video or share any multimedia file stored on the mobile terminal. The users can also share a whiteboard session during the CS telephony session. This way enriches and enhances the CS voice call by not limiting them to the capabilities of the current form of communication. This is available with a few simple keystrokes on the terminal. The following sections present the end-user features the Ericsson IMS WeShare service provides.
In last section, three levels of controls of hardware architectures were discussed for a general digital switching system. For effective processing of a call, to perform various functions of subsystems and to interface with the other subsystems, softwares plays a vital role. The software programs enables any digital switching system input data, to give outputs in a fraction of seconds, concurrent processing of many calls in real time and performs many features other than simple pathset between subscribers for conversation.
The second limiting factor is voltage drop. If the battery voltage is kept constant with increase in length, the effectiveness of the signalling and conversation will be limited. This is due to IR drop of the line. The IR drop of the line varies with resistances of the battery used in the system, telephone set resistance and the allowable resistance of the subscriber loop.
Various combinations of the features in Table 9.3 appear in different applications. A simple example such as voice can require different combinations of features in different contexts. Voice-based applications are almost always stream oriented. In many situations these applications are error tolerant. However, the degree of error tolerance is different when the stream is carrying voice than when it is carrying modem signals that carry data or fax. Furthermore, as the bit rate of voice signals decreases with compression, the degree of error tolerance also decreases. If silence suppression is used, the stream becomes variable bit rate rather than constant bit rate. Telephone conversations between humans requires real-time transfer with low delay and jitter. But voice mail and voice response applications are much more tolerant of delay. Finally, multiplexing may be important in voice applications where bandwidth costs are significant, but not relevant where bandwidth is cheap as in LANs.
Consider The Synchronous Multiplexing In Fig 4.17. Explain How The Pointers In The Outgoing Sts-1 Signal Are Determined
Let's consider an approach for providing fiber-to-the-home connectivity from the central office to the user. The telephone conversations of users are time-division multiplexed at the telephone office and broadcast over a passive optical network'' that operates as follows. The TDM signal is broadcast on a fiber up to a passive optical splitter'' that transfers the optical signal to N optical fibers that are connected to N users. Each user receives the entire TDM signal and retrieves its own signal. b. Next he calls home in Chicago to inform his wife that he forgot to take out his son's hockey gear from the trunk of his car and to give her the parking spot where he left the car in the airport ( somewhere on the third level of parking lot A''). Sketch the sequence of events that take place to set up his call. (Don't concern yourself with the specifics of the conversation.)
It may happen that, during a conversation, the mobile phone moves from one cell to another. When it does, the signal may become weak. To solve this problem, the MTSO monitors the level of the signal every few seconds. If the strength of the signal diminishes, the MTSO seeks a new cell that can accommodate the communication better. The MTSO then changes the channel carrying the call (hands the signal off from the old channel to a new one). Handoffs are performed so smoothly that most of the time they are not observed by the users.
Instant Messaging (IM) allows users to exchange information with others in near real time. The interactive nature of an IM service encourages the IM users to engage in conversations. Typically, small text messages are exchanged in IM conversations but the OMA Instant Messaging communication service supports other content types than text like images, video clips, etc., making it a multimedia IM service enabler. The OMA Instant Messaging communication service enables a user to send messages to another individual or to a group of users in which everyone can see what everyone else is sending. The service enabler supports two modes of IM communication. They are called the pager mode and IM sessions. The pager mode communication method is designed to handle brief message exchanges, while the latter is similar to a conference hosted by the network (i.e. the service can provide 'chat rooms') where the IM users join and leave the IM communication sessions over time. The OMA Instant Messaging...
Initially, the trunk is on-hook at both ends. Exchange A seizes trunk T, and sends a forward off-hook (seizure signal). Exchange then connects a digit receiver to the trunk, and sends a (backward) wink signal. After receipt of the wink, exchange A sends the digits of the called number. When the call is answered, exchange B sends an off-hook (answer signal). During the conversation, both exchanges are sending off-hook.
Most scientists agree that what makes human beings radically different from all other mammals is that we talk with each other. There has not been enough time for evolution to make human brains that much different from apes, and DNA research has reinforced this belief. Human speech, however, learned rapidly after birth, appears to drive many of the differences in brain organization that allow people to carry on a conversation with their parents but not with their dogs. This came as a surprise to researchers, since conventional wisdom attributed speech to human brain organization and not the other way around. People got bigger and better brains because they needed to express bigger and better thoughts vocally. Between the ages of 5 and 11 years, the average person learns about two or three new words a day, because it is necessary to do so. One further aspect of human voice should be discussed before moving on to the invention of the telephone. This is the fact that silence plays a large...
The discontinuous transmission mode takes advantage of the fact, that during a normal telephone conversation, both parties rarely speak at the same time, and thus each directional transmission path has to transport speech data only half the time. In DTX mode, the transmitter is only activated when the current frame indeed carries speech information. This decision is based on the VAD signal of speech pause recognition. The DTX mode can reduce the power consumption and hence prolong the battery life. In addition, the reduction of transmitted energy also reduces the level of interference and thus improves the spectral efficiency of the GSM system. The missing speech frames are replaced at the receiver by a synthetic background noise signal called Comfort Noise (Figure 6.3). The parameters for the Comfort Noise Synthesizer are transmitted in a special Silence Descriptor (SID) frame.
Frequency Division Multiple Access (FDMA) is one of the most common multiple access procedures. The frequency band is divided into channels of equal bandwidth such that each conversation is carried on a different frequency (Figure 2.6). Best suited to analog mobile radio, FDMA systems include the C-Netz in Germany, TACS in the UK, and AMPS in the USA. In the C-Netz, two frequency bands of 4.44 MHz each are subdivided into 222 individual communication channels at 20 kHz bandwidth. The effort in the base station to realize a frequency division multiple access system is very high. Even though the required hardware components are relatively simple, each channel needs its own transceiving unit. Furthermore, the tolerance requirements for the high-frequency networks and the linearity of the amplifiers in the transmitter stages of the base station are quite high, since a large number of channels need to be amplified and transmitted together 15,54 . One also needs a duplexing unit with...
When the called subscriber lifts his handset, the line is looped and ringing is removed. Once the conversation started, the exchange completes the connections between the subscribers. Normally, once the conversation is over, the exchange will be at idle state. But in general, there are two types difficulties arises.
In its simplest form, the switch consists of a patch cord panel and a human operator as shown in Figure 1.2-5a. The originating user picks up the telephone and in the process activates a signal in the circuit that connects it to the telephone office. The signal alerts the operator that a connection is requested. The operator takes the requested name and checks to see whether the desired user is available. If so, the operator establishes a connection by inserting the two ends of a cord into the sockets that terminate the lines of the two users as shown in Figure 1.2-5b. This connection allows electrical current, and the associated voice signal, to now between the two users. This end-to-end connection is maintained for the duration of the call. When the users are done with their conversation, they hang up their telephones, which generates a signal indicating that the call is complete. The two telephone lines are then available to make new connections. We say that telephone networks are...
When exchange R receives an answer signal from S2, it cuts through the path between T2 and the called subscriber line and sends an ANM message. When the message reaches exchange P, it cuts through its forward transmission path and the conversation or data transmission can start.
The basic call control is divided into three phases call set-up, the data conversation phase, and call cleardown (takedown). Messages on the signaling link are used to establish and terminate different phases of a call. Standard in-band supervisory tones and or recorded announcements are returned to the caller on appropriate connection types to provide information on call progress. Calls originating from ISDN terminals may be supplied with more detailed call progress information by means of additional messages in the access protocol supported by a range of messages in the network. Two signaling methods are used with ISUP
In-band signaling was the first type of signaling deployed in the telephone network. In-band signaling is defined as voice and signaling information sharing the same communications path. Think of this When you are talking on your analog home phone and your fingers hit keys on the telephone's numeric keypad, what do you hear You hear the DTMF beeps in the middle of your conversation. That is because analog telephones use in-band signaling. The signaling information (beeps) is carried in the main and only communications channel.
Out-of-band signaling is signaling that does not take place in the same path as the conversation. We are used to thinking of signaling as being in-band. We hear dial tone, dial digits, and hear ringing over the same channel on the same pair of wires. When the call connects, we talk over the same path that was used for the signaling. Traditional telephony used to work this way as well. The signals that set up a call between one switch and another always took place over the same trunk that would eventually carry the call. In early days, the out-of-band signaling was used in the 4 kHz voice grade channel (see Figure 8-2). The telephone companies used band pass filters on their wiring to contain the voice conversation within the 4 kHz channel. The band pass filters were placed at 300 Hz (the low pass) and at 3,300 Hz (the high pass). The range of frequencies above the actual filter was 700 Hz (4,000 - 3,300 700). In this additional spectrum, in-band signaling was sent down the wires...
The switching centers receives the control signals, messages or conversations and forwards to the required destination, after necessary modification (link amplifications) if necessary. A switching system is a collection of switching elements arranged and controlled in such a way as to setup a communication path between any two distant points. A switching center of a telephone network comprising a switching network and its control and support equipment is called a central office.
A major problem with in-band signaling is the possibility of talk-down, which refers to the premature activation or deactivation of supervisory equipment by an inadvertent sequence of voice tones through the normal use of the channel. Such tones could simulate the SF tone, forcing a channel dropout (i.e., the supervisory equipment would return the channel to the idle state). Chances of simulating a 2VF tone set are much less likely. To avoid the possibility of talk-down on SF circuits, a time-delay circuit or slot filters to by-pass signaling tones may be used. Such filters do offer some degradation to speech unless they are switched out during conversation. They must be switched out if the circuit is going to be used for data transmission (Ref. 2). In the short run, out-of-band signaling is attractive in terms of both economy and design. One drawback is that when channel patching is required, signaling leads have to be patched as well. In the long run, the signaling equipment...
Connection-oriented communication is where the sender first establishes contact with the recipient before sending the data. Because of the connection between the two parties, it is easy to obtain acknowledgment of receipt. This method is typical of a normal telephone conversation and in terms of networking protocols this is performed by the TCP IP protocol suite's transmission control protocol (TCP).
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