vocable full report
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22-01-2010, 04:11 PM

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Voice (and fax) service over cable networks is known as cable-based Internet Protocol (IP) telephony. Cable based IP telephony holds the promise of simplified and consolidated communication services provided by a single carrier at a lower cost than consumers currently to pay to separate Internet, television and telephony service providers. Cable operators have already worked through the technical challenges of providing Internet service and optimizing the existing bandwidth in their cable plants to deliver high speed Internet access. Now, cable operators have turned their efforts to the delivery of integrated Internet and voice service using that same cable spectrum.

Voice (and fax) service over cable networks is known as cable-based Internet Protocol (IP) telephony. Cable based IP telephony holds the promise of simplified and consolidated communication services provided by a single carrier at a lower cost than consumers currently to pay to separate Internet, television and telephony service providers. Cable operators have already worked through the technical challenges of providing Internet service and optimizing the existing bandwidth in their cable plants to deliver high speed Internet access. Now, cable operators have turned their efforts to the delivery of integrated Internet and voice service using that same cable spectrum.
Cable based IP telephony falls under the broad umbrella of voice over IP (VoIP), meaning that many of the challenges that telecom carriers facing cable operators are the same challenges that telecom carriers face as they work to deliver voice over ATM (VoATM) and frame-relay networks. However, ATM and frame-relay services are targeted primarily at the enterprise, a decision driven by economics and the need for service providers to recoup their initial investments in a reasonable amount of time. Cable, on the other hand, is targeted primarily at home. Unlike most businesses, the overwhelming majority of homes in the United States is passed by cable, reducing the required up-front infrastructure investment significantly.
Cable is not without competition in the consumer market, for digital subscriber line (xDSL) has emerged as the leading alternative to broadband cable. However, cable operators are well positioned to capitalize on the convergence trend if they are able to overcome the remaining technical hurdles and deliver telephony service that is comparable to the public switched telephone system.
In the case of cable TV, each television signal is given a 6-megahertz (MHz, millions of cycles per second) channel on the cable. The coaxial cable used to carry cable television can carry hundreds of megahertz of signals -- all the channels we could want to watch and more.
In a cable TV system, signals from the various channels are each given a 6-MHz slice of the cable's available bandwidth and then sent down the cable to your house. In some systems, coaxial cable is the only medium used for distributing signals. In other systems, fibre-optic cable goes from the cable company to different neighborhoods or areas. Then the fiber is terminated and the signals move onto coaxial cable for distribution to individual houses.
When a cable company offers Internet access over the cable, Internet information can use the same cables because the cable modem system puts downstream data -- data sent from the Internet to an individual computer -- into a 6-MHz channel. On the cable, the data looks just like a TV channel. So Internet downstream data takes up the same amount of cable space as any single channel of programming. Upstream data -- information sent from an individual back to the Internet -- requires even less of the cable's bandwidth, just 2 MHz, since the assumption is that most people download far more information than they upload.
Putting both upstream and downstream data on the cable television system requires two types of equipment: a cable modem on the customer end and a cable modem termination system (CMTS) at the cable provider's end. Between these two types of equipment, all the computer networking, security and management of Internet access over cable television is put into place.
Inside the Cable Modem
Cable modems can be either internal or external to the computer. In some cases, the cable modem can be part of a set-top cable box, requiring that only a keyboard and mouse be added for Internet access. In fact, if the cable system has upgraded to digital cable, the new set-top box the cable company provides will be capable of connecting to the Internet, whether or not we receive Internet access through our CATV connection. Regardless of their outward appearance, all cable modems contain certain key components:
¢ A tuner
¢ A demodulator
¢ A modulator
¢ A media access control (MAC) device
¢ A microprocessor

¢ Tuner
The tuner connects to the cable outlet, sometimes with the addition of a splitter that separates the Internet data channel from normal CATV programming. Since the Internet data comes through an otherwise unused cable channel, the tuner simply receives the modulated digital signal and passes it to the demodulator.
In some cases, the tuner will contain a diplexer, which allows the tuner to make use of one set of frequencies (generally between 42 and 850 MHz) for downstream traffic, and another set of frequencies (between 5 and 42 MHz) for the upstream data. Other systems, most often those with more limited capacity for channels, will use the cable modem tuner for downstream data and a dial-up telephone modem for upstream traffic. In either case, after the tuner receives a signal, it is passed to the demodulator.
¢ Demodulator
The most common demodulators have four functions. A quadrature amplitude modulation (QAM) demodulator takes a radio-frequency signal that has had information encoded in it by varying both the amplitude and phase of the wave, and turns it into a simple signal that can be processed by the analog-to-digital (A/D) converter. The A/D converter takes the signal, which varies in voltage, and turns it into a series of digital 1s and 0s. An error correction module then checks the received information against a known standard, so that problems in transmission can be found and fixed. In most cases, the network frames, or groups of data, are in MPEG format, so an MPEG synchronizer is used to make sure the data groups stay in line and in order.
¢ Modulator
In cable modems that use the cable system for upstream traffic, a modulator is used to convert the digital computer network data into radio-frequency signals for transmission. This component is sometimes called a burst modulator, because of the irregular nature of most traffic between a user and the Internet, and consists of three parts:
¢ A section to insert information used for error correction on the receiving end
¢ A QAM modulator
¢ A digital-to-analog (D/A) converter
¢ Media Access Control (MAC)
The MAC sits between the upstream and downstream portions of the cable modem, and acts as the interface between the hardware and software portions of the various network protocols. All computer network devices have MACs, but in the case of a cable modem the tasks are more complex than those of a normal network interface card. For this reason, in most cases, some of the MAC functions will be assigned to a central processing unit (CPU) -- either the CPU in the cable modem or the CPU of the user's system.
¢ Microprocessor
The microprocessor's job depends somewhat on whether the cable modem is designed to be part of a larger computer system or to provide Internet access with no additional computer support. In situations calling for an attached computer, the internal microprocessor still picks up much of the MAC function from the dedicated MAC module. In systems where the cable modem is the sole unit required for Internet access, the microprocessor picks up MAC slack and much more.
Cable Modem Termination System
The CMTS takes the traffic coming in from a group of customers on a single channel and routes it to an Internet service provider (ISP) for connection to the Internet. At the head-end, the cable providers will have, or lease space for a third-party ISP to have, servers for accounting and logging, Dynamic Host Configuration Protocol (DHCP) for assigning and administering the IP addresses of all the cable system's users, and control servers for a protocol called CableLabs Certified Cable Modems -- formerly Data Over Cable Service Interface Specifications (DOCSIS), the major standard used by U.S. cable systems in providing Internet access to users.
The downstream information flows to all connected users. It's up to the individual network connection to decide whether a particular block of data is intended for it or not. On the upstream side, information is sent from the user to the CMTS -- other users don't see that data at all. The narrower upstream bandwidth is divided into slices of time, measured in milliseconds, in which users can transmit one "burst" at a time to the Internet. The division by time works well for the very short commands, queries and addresses that form the bulk of most users' traffic back to the Internet.
A CMTS will enable as many as 1,000 users to connect to the Internet through a single 6-MHz channel. Since a single channel is capable of 30 to 40 megabits per second (Mbps) of total throughput, this means that users may see far better performance than is available with standard dial-up modems. The single channel aspect, though, can also lead to one of the issues some users experience with cable modems.
If we are one of the first users to connect to the Internet through a particular cable channel, then we may have nearly the entire bandwidth of the channel available for your use. As new users, especially heavy-access users, are connected to the channel, we will have to share that bandwidth, and may see our performance degrade as a result. It is possible that, in times of heavy usage with many connected users, performance will be far below the theoretical maximums. The good news is that this particular performance issue can be resolved by the cable company adding a new channel and splitting the base of users.
Another benefit of the cable modem for Internet access is that, its performance doesn't depend on distance from the central cable office. A digital CATV system is designed to provide digital signals at a particular quality to customer households. On the upstream side, the burst modulator in cable modems is programmed with the distance from the head-end, and provides the proper signal strength for accurate transmission.
For almost a century, we have taken for granted the nearly unfailing service provided by the public telephone network, often referred to as plain old telephone service (POTS). If the cable is to emerge as a legitimate alternative, many technical issues must be addressed. Perhaps the most fundamental of these is the evolution of the nationâ„¢s cable infrastructure from a one-way, broadcast medium to a two-way personal communication medium.
Half-Duplex versus Full-Duplex Cable Infrastructure
Cable was first introduced in the United States in the late 1950s. For the next 30 years, nearly every mile of buried cable was half-duplex and therefore capable of broadband transmission in the downstream direction (i.e., from the head end to the subscriber, but not in the upstream direction). Communication from the subscriber back to the head end was possible only by means of a telephone line.
This makes half-duplex lines cumbersome even for premium TV services, such as pay-per-view, that require upstream communication. It also makes half-duplex lines extremely inconvenient for Internet service, as a result of the fact that outbound e-mail messages and hypertext transfer protocol (HTTP) requests must be sent via the phone. Furthermore, it renders half-duplex lines completely useless for voice, as such service requires packets to be sent up- and downstream continuously.
In recent years, cable operators have heavily investigated possibilities for upgrading their buried cable from half to full duplex as a necessary first step toward capitalizing on the demand for integrated data and voice services. While upstream transmissions still are not as fast as downstream (typically 1.5 to 3 Mbps downstream and 500 kbps to 2.5 Mbps upstream), full duplex lines offer sufficient throughput to support cable-based IP telephony. As cable operators compete for subscribers with xDSL providers, the speed with which the cable operators replace older lines with full duplex lines will be critical to their ultimate success.
Telephony Service across a Broadcast Media
Unlike POTS, which was developed from the outset as a point-to-point communication technology, cable networks were originally designed to broadcast one signal to many recipients. There was no concept of dedicated circuits, and there was no need to parcel out bandwidth to individual subscribers. To enable cable-based IP telephony, modifications must be made to the way bandwidth is allocated and packets are delivered. This must be done without using the bulk of the cable spectrum, because most of the bandwidth will continue to be used for TV broadcasts.
Direct connect
Callers must be able to send and receive only their own voice packets, and these packets must be given priority over data packets to ensure that callers experience smooth, uninterrupted conversations. The first step in this process was addressed by the Data over Cable Service Interface Specification (DOCSIS). DOCSIS established universal ground rules for the transmission packets across cable networks, ensuring that packets will not be routed incorrectly.
DOCSIS was later enhanced (in version 1.1) with QoS and security features necessary for voice communication. DOCSIS 1.1 also enables the prioritization of packet traffic. This allows cable operators to give certain packets (i.e., voice) the right of way and allows the other traffic to be sent with a best-effort priority, as determined by bandwidth availability. However, even this second-generation DOCSIS standard was not intended to address all of the technical issues associated with cable-based voice service.
To fill in the gaps left by DOCSIS, CableLabs created the packet cable specification known as the network based call signaling (NCS) protocol for signaling voice calls over cable networks. NCS leverages the existing media gateway controlling protocol (MGCP), and the protocol is thus sometimes known as MGCP NCS. NCS uses network-based call agents to negotiate cable-based IP telephony calls. Call agents, ensure that voice packets traverse the network and are audible only at the two conversation end points.
While POTS is considered an extremely secure service, cable-based IP telephony is not. Much like cellular telephony, cable-based conversations are susceptible to illegal wire tapping and inadvertent chat conditions. To address this untenable situation, DOCSIS and NCS support multiple security services.
NCS currently supports the secure version of IP (IPsec) authentication specification. Adequate protection of telephony connections can be achieved if the telephony gateway accepts only packets that have been authenticated by IPsec. DOCSIS supports an encapsulation protocol for encrypting packet data across the cable network. The encapsulation protocol defines the frame format for carrying encrypted packet data, the set of supported data-encryption and authentication algorithms, and rules for applying the cryptographic algorithms to packet data.

DOCSIS currently employs the cipher block chaining (CBC) mode of the U.S. data encryption standard (DES) to encrypt packet data. The protocols are extensible, can support multiple encryption algorithms, and will, in all likelihood, be extended to support the new advanced encryption standard (AES) once it is in place.
Power Consumption
As we all know, traditional telephones draw all the power they need from POTS lines. Because the public telephone system has evolved to such a reliable state and is essentially immune from the effects of power outages, it is exceptionally rare that service is lost. Electrical utilities in most areas do not, however, offer this degree of unfailing reliability. Therefore, head-end and customer premises cable equipment that relies solely on the local electric company for power puts users at a risk of losing phone service should a power outage occur.
To address this issue, lifeline service requirements are being implemented across the country that require IP phones, such as those that connect to cable lines, to provide at least 4 hours of battery backup. To meet this requirement, equipment manufactures must develop phones that can be powered by as little as 3 watts. A key to achieving this is a telephony chipset that minimizes idle processing cycles and offers sufficient onboard memory to handle all signal processing.
The ideal cable-based, IP telephony system is typically built with a reduced instruction set computing (RISC) microprocessor to handle the signaling functions and digit collection. The necessary telephony peripherals, such as a local area network (LAN) controller and universal serial bus (USB), are on a single chip to conserve power, and dedicated hardware should be used for the cable-communications protocol. Several megabytes of high-speed random access memory is needed to store the telephony application. The nonvolatile memory should be electrically reprogrammable, like a FLASH memory, to enable on-line software updates.
A high-performance, low-power digital signal processor (DSP) is needed to support the analog functionality (e.g., codecs), noise reduction, and echo cancellation. A programmable DSP can greatly reduce application-development time for solution providers. Texas Instrumentsâ„¢ TMS320C54x DSP is one such chip.
Telephony billing is an extremely complex process. Most cable TV customers receive the same bill each month. Aside from pay-per-view requests, there is no need to meter or monitor customer usage. Telephone billing is quite different. A typical bill includes recurring monthly service fees, international and long-distance charges that vary based on time and day, and premium services, such as *69 and directory assistance, that are billed on a per-use basis.
To enable timely accurate billing, call agents or broadband telephony interfaces (BTI) must collect all relevant usage data. The BTI is the cable equivalent to the phone box that is outside every home. In the absence of a BTI, cable-based IP telephony can also be delivered using voice-enabled cable modems inside a customerâ„¢s home. If the call agents collect the billing data, the BTI or cable modem need not be involved. Otherwise, the software inside the BTI or cable modem must provide application programming interfaces (APIs) so that the billing system can access the relevant data. Depending on each cable operatorâ„¢s implementation, the data may be contained in standard management
Information base (MIB) or in unique files set up specifically for telephony metering.
For cable operators, choosing which standard to support and preparing their infrastructures to support voice is only the beginning of the technological obstacle course. What remains is the QoS challenge inherent in all VoIP implementations. Among the most significant QoS hurdles are transmission latency, echo, jitter, and lost packets. These QoS factors are relatively harmless for data transmissions but must be dealt with aggressively to provide acceptable voice quality.
Latency, or delay, causes two problems: echo and talker overlap. Echo is caused by the signal reflections of the speakerâ„¢s voice. Echo becomes a significant problem when delay is greater than 50 milliseconds. Because echo is a significant quality problem, equipment providers must implement echo cancellation. Talker overlap becomes significant if one-way delay is greater is greater than 250 milliseconds, so every effort must be made to minimize delay. The sources of delay in a VoIP implementation include the following:
Accumulation Delay (also called algorithmic delay)
This delay is caused by the need to collect a frame of voice samples to be processed by the voice coder. It varies from a single sample time (.125 microseconds) to many milliseconds.
Processing Delay
This delay is caused by the actual process of encoding and collecting samples into a packet for transmission. The encoding delay is a function of both the processor execution time and the type of algorithm used. Often, multiple voice-coder frames will be collected in a single packet to reduce overhead. For example, three frames of G.729 code words, equaling 30 milliseconds of speech, may be collected into a single packet. This process of encapsulating several small packets into a single larger frame is known as concatenation.
Network Delay
Network delay is a function of the processing that occurs as packets are sent across a network. This delay is caused by the physical medium and the protocols used to transmit the voice data as well as by the buffers used to remove the packet jitter on the receive side. The jitter buffers add additional delay that is used to smooth the jitter created by the varying times at which each packet arrives. This delay can be significant part of the overall delay, as it can be as high as 70 to 100 milliseconds.
Polling Delay
Cable-based IP telephony creates an additional latency that other packet networks do not because of the way head-end systems collect packets from callers. The head end polls the BTI at each customer location. Because the head end does not maintain a continuous connection with each BTI, there is a transmission delay while voice packets wait for the next poll. Therefore, it is important that cable-based IP telephony medium equipment minimize this delay by anticipating when the next poll will arrive (a process called grant synchronization), so that the packets are queued and ready to go.
Echo is present even in a conventional POTS network. However, it is acceptable because delay is less than 50 milliseconds, and the echo is masked by the normal side tone that every telephone generates. Echo becomes a problem in VoIP networks because the delay is almost always greater than 50 milliseconds. Thus, echo-cancellation techniques must be used.

Echo is generated toward the packet network from the telephone network. The echo canceller compares the voice data received from the packet network with voice data being transmitted to the packet network. The echo from the telephone network is removed by a digital filter on the transmit path into the packet network.
The delay problem is compounded by the need to remove jitter-a variable interpacket timing caused by the fact that packets do not all cross the network at the same speed. Removing jitter requires collecting packets and holding them long enough to allow the slowest packets to arrive and be played in the correct sequence. This causes significant delay. The conflicting goals of minimizing delay and removing jitter have led to various schemes aimed at optimizing the size of the jitter buffer to minimize its impact on latency.

A common approach in the cable-based IP telephony is to count the number of packets that arrive late and create a ratio of these packets to the number of packets that are successfully processed. This ratio is then used to adjust the jitter buffer to target a specific late-packet ratio.
Lost Packets
In todayâ„¢s IP networks, voice frames are treated exactly like data. Under peak loads and congestion, voice frames will be dropped at the same rate as data frames. The data frames, however, are not time-sensitive, and dropped packets can be corrected through retransmission. Lost voice packets cannot be handled in the same manner. Some techniques used by VoIP software to address this problem include the following:
¢ Interpolation - Interpolate for lost speech packets by replaying the last packet received during the interval when the lost packet was supposed to be played out. This scheme is a simple method that fills the time between noncontiguous speech frames. It works well when the incidence of lost frames is infrequent. It does not, however, work very well when the incidence of lost frames is infrequent. It does not, however, work very well if there are a number of consecutive lost packets or a burst of lost packets.
¢ Redundancy - Send redundant information at the expense of bandwidth utilization. The basic approach replicates and sends the nth packet along with the (n+1)th packet. This method has the advantage of being able to correct for the lost packet exactly. However, this approach uses more bandwidth and also creates greater delay.
¢ Voice coder - A hybrid approach uses a lower-bandwidth voice coder to provide redundant information carried along in the (n+1)th packet. This reduces the problem of bandwidth consumption but does not solve the problem of delay.
The challenges of implementing fax-over-cable networks are similar to those of voice. The two most significant issues are timing and lost packets. The delay of fax packets through a packet network causes the precise timing that is required for the fax protocol to be skewed and can result in the loss of the call. The fax-over-packet protocol compensates for the skewed timing of message so calls are not dropped, and the accuracy of faxed images is not compromised.
Lost packets can be an even more serious problem for IP fax systems than for IP voice systems. In a VoIP application, the loss of packets can be addressed by replaying last packets and using other methods of interpolation. Fax-over-IP applications, however, have more severe constraints on the loss of data because the fax protocol can fail if the information is lost. The severity of the problem varies depending on the type of fax machine and whether or not error correction mode is enabled.
CableLabs is working on specific fax services that will be added to NCS to standardize the implementation of fax-over-cable networks, and there is currently an optional fax relay service in the NCS protocol.
Texas Instruments and its subsidiary, Telogy Networks, have developed one implementation of embedded VoIP software for cable-based IP telephony. The software supports cable modems and BTIs (up to four ports), as well as the telephony gateway (up to several thousand ports), at the cable head end. The software supports MGCP (Media Gateway Control Protocol) as well as the session protocol (SIP). The basic purpose of the two protocols, which is to process packetized voice traffic, is the same. However, the software supports both because standard bodies are divided as to the relative merits of each. This software interfaces to both streams of information from the telephony network and converts them to a single stream of packets transmitted to the packet network.
MGCP is a centralized call-processing system in which intelligence resides primarily at the head end. The cable modems and BTIs are similar to dumb client, and the system relies on call agents to negotiate the call through the network. SIP is a distributed system in which the intelligence resides in the BTI, and the head end is mainly a gateway to the public telephone network.
The greatest benefit of MGCP implementations is simplified, efficient management and administration. Fault detection and isolation are typically limited to the head end. Furthermore, there is no need to distribute software upgrades and patches to customers, and, thus, there is also no concern about software-version synchronization among all BTIs.
The supporters of SIP, on the other hand, argue that it is a more scalable and reliable system. The case for scalability relies on the fact that the head end, acting mainly as a gateway, is unlikely to bottleneck subscriber capacity. Traffic load, rather than the processing time, would be the only potential bottleneck. Supporters also claim that SIP is more reliable, because a SIP-based network architecture does not have a single point of failure.
The MGCP-and SIP-compliant architecture processes voice packets similarly using either protocol. The software is broken down into two parts: the DSP and microprocessor components. The DSP processes voice data and passes voice packets to the microprocessor with generic voice headers. The microprocessor component is responsible for moving voice packets and adapting the generic voice headers to the NCS protocol. The microprocessor also processes signaling information and converts it from a telephony signaling protocol to IP.
This partitioning of functionality between the DSP and microprocessor provides a clean interface between the generic processing functions (such as compression, echo cancellation, and voice-activity detection) and the application-specific signaling and protocol processing.
DSP Component (or Voice-Processing Module)
This software prepares voice samples for transmission over the packet network. Its components perform echo cancellation, voice compression (to conserve cable bandwidth), voice activity detection, jitter removal, clock synchronization, and voice packetization. This unique software, along with TIâ„¢s programmable DSP technology, provides a comprehensive yet flexible foundation that allows equipment providers to shave months off of typical development schedules, resulting in tremendous cost savings and a critical time-to-market boost.
Microprocessor Component
In NCS-based products, the microprocessor component handles detection of various events to call agents, dual tone multifrequency (DTMF) digit collection and reporting, application of signals, and forwarding of audio packets. The microprocessor component in NCS-based products is comprised of the following software modules:
¢ XGCP signaling module (XGCM)
¢ Digit collection module (DCM)
¢ DSP interface module (DIM)
PCM Interface (Pulse Code Modulation Interface)
This interface receives pulse code modulation (PCM) samples from the digital interface and forwards them to the appropriate DSP software modules for processing. It also forwards processed PCM samples received from DSP software modules to the digital interface and performs continuous resampling of output samples to avoid sample slips.
Tone Generator
This generates DTMF tones and call-progress tones under command of the host (e.g., telephone, modem, private branch exchange [PBX], or telephone switch). It supports U.S. and international tones.
Echo Canceller
This performs echo cancellation on sampled, full-duplex voice signals. It has a programmable range of tail lengths.
Voice Activation Detector
This monitors the received signal for voice activity. When no activity is detected for a specific period of time, the software informs the IP. This prevents the encoder output from being transported across the network when there is silence so as to save the bandwidth. This software also measures the idle noise characteristics of the telephony interface. It reports this information to the IP in order to relay this information to the head end for noise generation when no voice is present.
Tone Detector
This detects the reception of DTMF tones and performs voice and fax discrimination. Detected tones are reported to the host so that the appropriate speech or fax functions are activated.
Voice Codec Software
This software compresses the voice data for transmission over the packet network. It is capable of numerous compression ratios through its modular architecture. A compression ratio of 8 to 1 is achievable with the G.729 voice codec.
Fax Software
This software performs a fax relay function by demodulating PCM data, extracting the relevant information, and packing the fax-line scan data into frames for transmission.
Voice Playout Unit
This buffers voice packets received from the packet network and sends them to the voice codec for playout. The following features are supported:
¢ First in, first out (FIFO) buffer that stores voice codewords before playout to remove timing jitter from the incoming packet sequence.
¢ Continuous-phase resampler that removes timing-frequency offset without causing packet slips or loss of data.
¢ Timing-jitter measurement, which allows adaptive control of FIFO dealy.
The voice packetization protocols use a sequence number field in the transmit-packet stream to maintain temporal integrity of voice during playout. Using this approach, the transmitter inserts the contents of a free-running, modulo-16 packet counter into each transmitted packet, allowing the receiver to detect lost packets and to reproduce silent intervals during playout.
Packet Voice Protocol
This encapsulates compressed voice and fax data for end-to-end transmission over a backbone network between two ports.
Control Interface Software
This coordinates the exchange of monitor and control information between the DSP and host via a mailbox mechanism. Information exchanged includes software downline load, configuration data, and status reporting.
Real-Time Portability Environment
This provides the operating environment for the software residing on the DSP. It also provides synchronization functions, task management, memory management, and timer management.
Unsolicited Grant Service(UGS)
Cable networks are asymmetric, i.e., the downstream data received is streaming while the upstream data transmitted is either transmitted on a collision time fragment or must get a time slot or grant. Because requesting a grant can cause significant delay, UGS ensures that cable modems will be contacted at regular intervals without having to make separate requests. The concatenation process mentioned earlier can lighten UGS requirements and increase the efficient bandwidth.
UGS with Activity Detection (UGS-AD)
Upon detection of voice inactivity, UGS-AD enables network resources to be diverted to other cable modems and data flows, maximizing the efficiency of all data transmissions.

The DIM is responsible for the interface to the DSP software. The microprocessor communicates with the DSP through a shared memory arrangement, the mechanics of which are hidden by the DIM. The DIM shields the rest of the microprocessor software from the complexities of the DSP interface.
This module is responsible for providing MGCP embedded client functionality. It parses and processes each message received from an MGCP call agent. It reports detected events to the call agent, generates signals requested by the call agent, reports detected DTMF digits, and sets up connections requested by the call agent. This module is also responsible for forwarding audio packets received from the DSP to packet network interface and forwarding audio packets received from the packet network interface to the DSP.
This module is responsible for processing dialed digits received from the XGCP module. It accumulates all the dialed digits and matches them against the digit map. It reports the results along with the accumulated digits to the XGCP module.
Network Management Module
This module is responsible for providing the management interface to configure and maintain the other modules of the software. A sample module is provided, but the customer may replace the sample with a custom module. A proprietary voice packet MIB is supported because no standard MIB exists.


Fundamental to any communications system is the ability to discover, isolate and remedy problems as quickly as possible to minimize or eliminate the degree to which users are impacted. The software offers a rich set of diagnostic features to accomplish this.
Loop-Back Capability
PCM loop-back is used for diagnosing problems related to the telephony side. Packet-send loop-back diagnoses problems related to the DSP software performance. Packet-receive loop-back diagnoses problems related to the packet network.
Signal-Level Measurement
The software can perform signal-level measurements on the telephony-side interface for a particular channel. It reports instant and mean values for signal power in both directions.
Packet Network Statistics
The software generates an extensive set of performance statistics for each channel. The statistics include number of transmitted and received packets, minimum and maximum packet interarrival times (i.e., jitter measurement), number of invalid packet headers, and number of lost packets. In addition, the voice playout unit reports number of lost voice frames, number of idle frames, number of dropped frames, and average packet jitter.
PCM Sample Trace
The DSP software can provide 10 milliseconds of PCM samples (approximately 80 samples).
Memory Trace
The DSP software can provide 40 memory values, starting from the requested location. Memory locations can be from data memory or program memory.
Fax Processing Debug
The software provides a stream of debug information, tracing the performance of fax operations.
Echo Canceller Statistics
The software offers a detailed set of echo-canceller statistics.
With the merging of telecom carriers, cable operators, and Internet service providers (ISPs), most experts agree that convergence is not merely a trend but an inevitability. The potential cost savings, consolidated billing, streamlined network management, and overall convenience are too compelling for service providers and customers to ignore. With buried cable passing hundreds of homes worldwide, it is logical to assume that cable will be front and center as convergence becomes a mainstream.
The technical challenges will be overcome as innovation and experience combine to provide cable-based IP-telephony solutions that are equals of the public telephone system. The software architecture has been field tested and is designed to provide equipment manufacturers with a repeatable, core starting point to help them develop unique, value-added telephony solutions and bring those solutions to market as quickly and cost-effectively as possible.
1. iec.org
2. itpapers.com
3. altavista.com
4. howstuffworks.com
5. cable-modems.org

I express my sincere gratitude to Dr. Agnisarman Namboodiri, Head of Department of Information Technology and Computer Science , for his guidance and support to shape this paper in a systematic way.
I am also greatly indebted to Mr. Saheer H.B. and Ms. S.S. Deepa, Department of IT for their valuable suggestions in the preparation of the paper.
In addition I would like to thank all staff members of IT department and all my friends of S7 IT for their suggestions and constrictive criticism.
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