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Digital transmission. 

Digital transmission. 

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Conference Paper
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From the user perspective In developed countries bandwidth now a days is a commodity. Users hardly worry about bandwidth limitations anymore, bandwidth is an afterthought. In the rest of the world it is not the same case, especially in places where the digital gap is enormous; bandwidth is still a precious resource. Service providers everywhere sti...

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... BW = bandwidth; f1 = the highest frequency; f2 = the lowest frequency, so it clearly is the difference between the highest and lowest frequencies[1]. Vowels and consonants provide a perfect example of bandwidth. Vowels occupy higher frequencies and consonants occupy lower frequencies. The human ear can detect frequencies in the range of 40 to 18,000 Hz. Early telecommunications transmission devices did not have the capacity to transmit in the higher frequencies, fortunately this was not a major problem as the human brain has the capacity to reconstruct sound characteristics and the intention of a message[2]. Initially, for voice transmission, a 4000Hz channel was defined as the optimal one. This meant that the channel could transmit frequencies ranging from 0 to 4KHz. The first telephone companies connected only a handful of subscribers as each subscriber received a pair of copper cables at 4 KHz. It was a great disadvantage having to provision a cable to each of the subscriber’s home[1]. There’s a direct relationship between bandwidth and transmission speed (bits per second), the more bandwidth allocated to a channel the more transmission capacity there will be. Transmission capacity (R) is expressed by R = 2 f1. In copper cables it is possible to transmit beyond 100,000 Hz, so a method for transmitting beyond the 4,000 Hz per user was needed and here’s where FDM plays the important role of Multiplexing. FDM stands for Frequency Division Multiplexing. With FDM up to 30 voice channels can be transmitted simultaneously; the first channel is transmitted in that range from 0 to 4 KHz as previously explained, the second channel is transformed to transmit in the range between 4 KHz to 8 KHz, the third channel transmits from 8 KHz to 12 KHz and so on, until completing the 30 channels. Figure 2 explains this concept in greater detail. The advantage with FDM is that it reduced considerably the cable requirements to connect subscribers. Noise. Cables, as antennas, receive airwave signals, but in cables these signals interfere with the original information being sent, this interference is called noise, unwelcomed noise. Noise is generated mostly form sources of electrical radiation such as electric engines, fluorescent lights and even from other telephone lines [1]. Unfortunately in analog transmission systems noise cannot be separated from the original signal. As a transmitted voice signal progresses throughout a cable it starts to weaken and it could fade away before it reaches its destination; to avoid this the signal needs to be amplified; along the signal path noise has been incorporating to the original signal so when the signal is amplified the incorporated noise is amplified as well resulting in a corrupted signal at the receiving end. Figure 3 highlights this phenomenon. Digital Systems. There have been two major developments that changed telephony networks: Digital transmission and Common Channel Signaling [3]. Digital telephony transmit voice signals as bit strings, these strings maintain very low levels of noise facilitating signal switching and transmitting different signals in the same telephone line. Common Channel Signaling (CCS) allows control information to optimize the deployment of digital services. Figure 4 highlights a digital signal. Voice digitalization tackles the noise problem very well. The main difference between analog and digital systems is that digital systems already “know” what they will be receiving from the transmitter. An analog signal could be any value in the signal curve, therefore any small variation caused by noise is difficult to detect and impossible to remove. A digital signal on the other hand can only have few values (-1, 0, +1), so any variation represents noise; during the transmission the digital signal is regenerated periodically restoring the original signal transmitted (Figure 5). The transmitter and digital logic lead to another technology: Time Division Multiplexing (TDM). TDM assigns time intervals to transmission channels (Generally 30 or 24 channels, depending of European or North American standards), and it rotates those channels. The most common voice digitalization method is Pulse Code Modulation, PCM. To convert analog voice into a digital signal it goes through a CODEC process (Codification-Decodification). The conversion takes place in two steps: 1.- Modulation by pulse amplitude. Based on the mathematical theorem by Harry Nyquist, a signal representing the voice variations entering the modulator is sampled 8,000 times per second, the modulator utilizes the resulting sample by transmitting a narrow wave pulse per each individual sample, which voltage (height) is the same as in the analog signal [3]. 2.- Digital Codification. The pulse height is converted into a digital value; the output is an octet (byte) that represents the pulse voltage, i.e. the sampled voice. The two steps process converts the analog signal into a 64,000 bit string. (8,000 samples x 8 bits), see figure 6. The 64 KBPS (Kilo Bits Per Second) is commonly known as a DS-0 (Digital Signal level 0). Bandwidth Requirements. With the above description we can define now bandwidth in the most trivial way and it is the amount of information that it can be transmitted in any form of connection in a given time. Usually it is measured in bits per second. Thanks to the Internet and all the services being offered over this technology, the per-capita bandwidth has been increasing worldwide as new services such as streaming video, Voice over IP (VoIP), conferencing, etc. are demanding more and more bandwidth, and bandwidth has been becoming more accessible to the end user, mostly in developed countries. Bandwidth availability has not been equal throughout the world. In underdeveloped countries prices are still prohibited. Table 1 highlights the current scenario in 30 countries: As you can appreciate from the table above the bandwidth availability and price difference between South Korea and Mexico is huge. Even though bandwidth has become more affordable, providers are still continuing to economize the bandwidth, that’s why we have different rates for uploading and downloading and that’s the reason why voice and video compression algorithms exist, to economize on bandwidth resources. VoIP telephony providers require great lengths of bandwidth for servicing their end customers, for example let’s estimate the bandwidth requirements for market standard Media Gateways. A typical Media Gateway that supports up to 16 E1s, which is the equivalent of 480 concurrent voice ports (calls), are normally 4 rack units (RU) tall therefore a conventional 19 inch rack could host 7 Media Gateways. This means that a full rack would serve 3,360 users simultaneously. If we acknowledge that G7.11 is the best CODEC then we need to reserve at least 64 KBPS per voice port for this CODEC, that’s 215 MBPS of bandwidth to transmit voice packets over a data network, and this requirement is for voice media only, it is not considering the packet’s that transmit the informational headers and trailers data. And this requirement is minimum in comparison to physically smaller equipments with higher capacity, there are 2RU Media Gateways with DS-3 (Digital Signal, Level 3) interfaces out there. This means that a single one of these equipments can support up to 4,042 simultaneous voice calls. Being 2 RU tall means that 14 of these units would fit a typical 19 inch rack. Using a higher compression CODEC, this time, G.729, the amount of bandwidth needed for a fully loaded rack is 452 MBPS. That’s a lot of bandwidth!!! Normally tier 1 carriers have more than one fully loaded rack in their location; they have several. So regardless of how inexpensive the Bandwidth is every provider will still try to economize on their Bandwidth resources. There are several data network types out there, worldwide, that can be used for transmitting packet voice such as Frame Relay or MPLS (Multi Protocol Label Switching); to save bandwidth, voice compression algorithms are still needed. Compression Algorithms. We have loosely calculated bandwidth requirements and we spoken vaguely about compression algorithms. I mentioned the G.711and G.729 compression methods, in ...

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