Today the Information and Communication Technology becoming more broaden & advanced, and provides all the users with a broad perspective on the nature of technology, how to use and apply a variety of technologies, and the impact of information and communication technologies on themselves and on society. Adopting the facility that has been given by the technology is not intended to stand alone, but rather to be adopting and extending all over the world on the public level should be far better. What technology is using people from different part of the world is different on their understanding because of their level of difference in country development and technology implementation. VoIP (Voice over IP) is the top most popular and burning communication technology for the upcoming decades. So, let’s talk about it.
Sunday
VoIP Gateways:
An Overview
Gateways have become a central, yet complex, component in most state-of-the-art VoIP systems. Although they’ve been around for years, VoIP gateways remain something of a mystery. What, exactly, are these devices gateways to? Do they lead the way into a data network, a voice network, telephones, network management or outright confusion? In a way, they actually open the door to all of these areas. That's because VoIP gateways have become a central, yet complex, component in most state-of-the-art VoIP systems.
VoIP gateways act as VoIP network translators and mediators. Perhaps most importantly, they translate calls placed through the public switched telephone network (PSTN) - the "regular" telephone system - into digital data packets that are compatible with an enterprise's VoIP system. VoIP gateways can also help direct VoIP calls to specific users with the assistance of built-in routing tables. Additionally, the units can translate between different VoIP protocols, such as H.323 and SIP, enabling compatibility between various VoIP systems and devices.
Given all of these benefits, it's easy to see why VoIP gateways are highly recommended for virtually any VoIP implementation. Yet this hasn't always been the case. In VoIP's early days, system designers often "VoIP-enabled" switches and routers to handle key gateway tasks. But as VoIP networks grew larger and more sophisticated, and as end users began demanding higher quality and more reliable service, most designers began specifying standalone VoIP gateways for their systems.
Gateways have become a central, yet complex, component in most state-of-the-art VoIP systems. Although they’ve been around for years, VoIP gateways remain something of a mystery. What, exactly, are these devices gateways to? Do they lead the way into a data network, a voice network, telephones, network management or outright confusion? In a way, they actually open the door to all of these areas. That's because VoIP gateways have become a central, yet complex, component in most state-of-the-art VoIP systems.
VoIP gateways act as VoIP network translators and mediators. Perhaps most importantly, they translate calls placed through the public switched telephone network (PSTN) - the "regular" telephone system - into digital data packets that are compatible with an enterprise's VoIP system. VoIP gateways can also help direct VoIP calls to specific users with the assistance of built-in routing tables. Additionally, the units can translate between different VoIP protocols, such as H.323 and SIP, enabling compatibility between various VoIP systems and devices.
Given all of these benefits, it's easy to see why VoIP gateways are highly recommended for virtually any VoIP implementation. Yet this hasn't always been the case. In VoIP's early days, system designers often "VoIP-enabled" switches and routers to handle key gateway tasks. But as VoIP networks grew larger and more sophisticated, and as end users began demanding higher quality and more reliable service, most designers began specifying standalone VoIP gateways for their systems.
Various Vendors
With VoIP technology steadily gaining momentum, VoIP gateway shoppers have an array of products to choose from. Leading VoIP gateway vendors include Cisco Systems, Mediatrix Telecom, Quintum Technologies, Stratus, Welltech Computer and Nortel Networks. VoIP gateways can be either hardware- or software-based. Hardware-based VoIP gateways - by far the most widely used approach - are available as standalone boxes, chassis cards or modules. Hardware VoIP gateways, while generally most expensive than their software counterparts, are usually preferred because they are viewed as more reliable, provide built-in interfaces and don't consume computer processing power.
In the enterprise market, VoIP gateways come in many different configurations. Buyers can select from products that offer numerous phone, fax machine, PBX and PSTN support capabilities. Additionally, for large enterprises with offices and branch operations spread around the country or world, VoIP gateways provide an effective way to extend and distribute voice communications systems.
At the market's low-end, it's possible to find a basic VoIP gateway, featuring a phone jack, Ethernet router and firewall, for under $200. A device at this price level would likely offer a minimum of three ports: a standard RJ-11 telephone jack and two RJ-45 ports - one for a broadband modem/router and one for a computer or network sharing device. Such a system would be capable of handling the voice needs of a home or small office.
A mid-level VoIP gateway, costing anywhere from $400 to $2,000, offers additional interfaces supporting a wide range of phone system and network devices. These products also include various quality of service (QoS) features, network-thrifty voice compression and built-in security capabilities, such as encryption. The primary selection criteria of these VoIP gateways is the maximum packet throughput and the number of simultaneous phone calls supported. A VoIP gateway buyer needs to know just how much capacity his or her VoIP system needs, and these figures can only be arrived at by a thorough professional analysis.
At the market's high end are Carrier Class VoIP gateways, costing several thousand dollars. Widely used by both telephone carriers and large enterprises, these devices support hundreds or even thousands of channels for advanced voice services, such as interactive voice response (IVR), a technology that allows callers to select an option from a voice menu. Other advanced functions supported by carrier-class VoIP gateways include voice recording, distributed voice announcements and conference calls.
In the enterprise market, VoIP gateways come in many different configurations. Buyers can select from products that offer numerous phone, fax machine, PBX and PSTN support capabilities. Additionally, for large enterprises with offices and branch operations spread around the country or world, VoIP gateways provide an effective way to extend and distribute voice communications systems.
At the market's low-end, it's possible to find a basic VoIP gateway, featuring a phone jack, Ethernet router and firewall, for under $200. A device at this price level would likely offer a minimum of three ports: a standard RJ-11 telephone jack and two RJ-45 ports - one for a broadband modem/router and one for a computer or network sharing device. Such a system would be capable of handling the voice needs of a home or small office.
A mid-level VoIP gateway, costing anywhere from $400 to $2,000, offers additional interfaces supporting a wide range of phone system and network devices. These products also include various quality of service (QoS) features, network-thrifty voice compression and built-in security capabilities, such as encryption. The primary selection criteria of these VoIP gateways is the maximum packet throughput and the number of simultaneous phone calls supported. A VoIP gateway buyer needs to know just how much capacity his or her VoIP system needs, and these figures can only be arrived at by a thorough professional analysis.
At the market's high end are Carrier Class VoIP gateways, costing several thousand dollars. Widely used by both telephone carriers and large enterprises, these devices support hundreds or even thousands of channels for advanced voice services, such as interactive voice response (IVR), a technology that allows callers to select an option from a voice menu. Other advanced functions supported by carrier-class VoIP gateways include voice recording, distributed voice announcements and conference calls.
Getting Smarter
Building new VoIP gateway features and functions, such as faster translations and support for emerging VoIP standards, represents a major challenge for vendors. Fortunately, many enhancements are software based, and can be delivered to customers fairly quickly and inexpensively in the form of a simple software upgrade.
Perhaps the biggest trend in VoIP gateway technology is the rapid shift toward "smarter" products. Most major vendors are developing products that work with a wider mix of VoIP products and technologies, paving the road to enhanced multi-vendor interoperability. This trend promises to allow businesses to cut costs by enabling them to purchase products from any company that offers the best features at the best rather than from a single vendor.
In the months and years ahead, VoIP gateway customers can expect more products, enhanced features and increased interoperability. These trends promise to help enterprises more easily build, maintain and upgrade VoIP networks that support both inexpensive and high-quality calls.
Perhaps the biggest trend in VoIP gateway technology is the rapid shift toward "smarter" products. Most major vendors are developing products that work with a wider mix of VoIP products and technologies, paving the road to enhanced multi-vendor interoperability. This trend promises to allow businesses to cut costs by enabling them to purchase products from any company that offers the best features at the best rather than from a single vendor.
In the months and years ahead, VoIP gateway customers can expect more products, enhanced features and increased interoperability. These trends promise to help enterprises more easily build, maintain and upgrade VoIP networks that support both inexpensive and high-quality calls.
Wednesday
Building VoIP Gateways
Building Residential VoIP Gateways
A look at the security issues surrounding residential VoIP gateways Customer Premises Equipment - IP phones and media gateways with VoIP capability - is vulnerable to many Internet attacks, such as malformed frames or packet floods, both of which lead to Denial of Service attacks (DoS). Since DoS consumes significant equipment CPU processing cycles, this results in impaired voice quality in a real-time call processing scenario. This article addresses the implementation of security in such residential voice gateways.
Sections included in this paper include:• Areas for VoIP Security • VoIP Security Performance Measurement • Encryption Protocols • Key Exchange Models • Security Association • VoIP Configuration SecurityRegister to download this white paper now.
Sections included in this paper include:• Areas for VoIP Security • VoIP Security Performance Measurement • Encryption Protocols • Key Exchange Models • Security Association • VoIP Configuration SecurityRegister to download this white paper now.
Gateway VoIP Implementation
One San Francisco hotel's experience installing a gateway-based VoIP system. White Star is a large hotel located in US’s West Coast serving guests coming from Asia, Europe, Latin America and US. With a majority of its customers being business persons, there is a large volume of long-distance and international phone calls made from the hotel which are routed in the traditional telephone network (PSTN).A feasibility study on VoIP was carried out and concluded with the following two main points:• The VoIP voice quality is indistinguishable from the traditional phone calls. • Rates for VoIP calls charged by Savytel represent a large saving, compared to the rates charged by the traditional telephone service providers.
VoIP Introduction
Voice over IP (VoIP) uses the Internet Protocol (IP) to transmit voice as packets over an IP network। So VoIP can be achieved on any data network that uses IP, like the Internet, Intranets and Local Area Networks (LAN). Here the voice signal is digitized, compressed and converted to IP packets and then transmitted over the IP network. The Voice-over-Internet Protocol (VoIP) application meets the challenges of combining legacy voice networks and packet networks by allowing both voice and signaling information to be transported over the packet networks.
Voice over IP (VoIP) uses the Internet Protocol (IP) to transmit voice as packets over an IP network। So VoIP can be achieved on any data network that uses IP, like the Internet, Intranets and Local Area Networks (LAN). Here the voice signal is digitized, compressed and converted to IP packets and then transmitted over the IP network. The Voice-over-Internet Protocol (VoIP) application meets the challenges of combining legacy voice networks and packet networks by allowing both voice and signaling information to be transported over the packet networks.
Organizations around the world are seeking to reduce rising communications costs. Consolidation of separate voice, fax, and data resources offers an opportunity for significant savings. Organizations are pursuing solutions that enable them to take advantage of excess capacity on broadband networks for voice, fax, and transmission, as well as to utilize the Internet and company Intranets as an alternative to costlier mediums. VoIP could be applied to almost any voice communications requirement, ranging from a simple inter-office intercom to complex multi-point teleconferencing/shared-screen environments. Accordingly, the challenge of integrating voice, fax, and data is becoming a rising priority for many network managers.Although there are thousands of standards and technical specifications for circuit-switched telephony, the systems themselves are generally proprietary in nature. The opportunities for third parties to develop new software applications for these systems are extremely limited. The systems generally also require extensive training to operate and manage. Unlike the Circuit-switched networks, IP architectures are open and competition friendly, thus enabling the implementation of new features more quickly. Also features can be developed and deployed in a few months. Commonly, the operating system is less tightly coupled with the hardware, and the application software is quite separate again from the operating system. This situation also enables a greater range of choices for the purchaser. IP systems tend to use distributed client-server architecture rather than large monolithic systems. This type of architecture means that there are companies that make only portions of the network solution, enabling the customer to pick those companies that are best in different areas and creating a solution that is optimum in all respects. Using more sophisticated coding algorithms, voice can be transferred at rates as low as 8 Kbps compared to 64 Kbps required by traditional telephony networks. As the transmission capacity accounts for the large initial investment and large percentage of carrier’s operational costs, these bandwidth savings can mean a big difference to the bottom line.The advantages of reduced cost and bandwidth savings of carrying voice over packet networks are associated with some QoS issues unique to packet networks। By looking at the excitement that VoIP has generated and the resources that have been applied in developing the technical solutions for VoIP, it seems like VoIP is becoming a serious alternative for voice communications, especially to the Circuit-switched telephony that has been around for decades. However to become a serious alternative, it has to be able to provide the same quality and reliability, which Circuit-switched telephony is able to provide for decades. Right now there is lot of research is going on in the VoIP field both in Industry and in Academy, which can make advancements in the overall voice quality and seamless integration with the traditional packet switching networks.
Factors affecting VoIP quality
There are several factors that profoundly impact the quality of voice over the Internet. These factors can be described in terms of their general affect on VoIP quality: Negative or Positive.
There are several factors that profoundly impact the quality of voice over the Internet. These factors can be described in terms of their general affect on VoIP quality: Negative or Positive.
Negative Factors
Of the three negative factors for VoIP performance, the first one is delay, which results in echo and talker overlap. The second one is jitter, which is essentially the variation in delay. The third problem is packet loss. These factors are explained in much more detail below.
DelayDelay results in echo and talker overlap. Echo becomes a problem when the round-trip delay becomes high. Talker overlap (the problem of one caller stepping over the other talker’s speech) becomes significant if the one-way delay becomes greater than 250 milliseconds.
JitterJitter is essentially the variation in delay. This is primarily introduced because of the variation in inter-packet arrival time.
Of the three negative factors for VoIP performance, the first one is delay, which results in echo and talker overlap. The second one is jitter, which is essentially the variation in delay. The third problem is packet loss. These factors are explained in much more detail below.
DelayDelay results in echo and talker overlap. Echo becomes a problem when the round-trip delay becomes high. Talker overlap (the problem of one caller stepping over the other talker’s speech) becomes significant if the one-way delay becomes greater than 250 milliseconds.
JitterJitter is essentially the variation in delay. This is primarily introduced because of the variation in inter-packet arrival time.
Packet Loss
Packet loss is a constant problem in packet-based networks. In a circuit-switched network, all speech in a given conversation follows the same path and is received in the order in which it is transmitted. If something is lost, the cause is a fault rather than an inherent characteristic of the system.
Apart from these factors there could be impairments caused by codecs. These impairments are due to the distortion introduced by the codec and due to the interaction of network effects with codec operation. Speech coding and compression Both speech coding and compression have been used in the traditional telephony for over two decades. With the exception of the local loop, almost all voice is carried over the PSTN in digital format. The received analog voice undergoes an analog-digital conversion at 8000 samples per second with 8 bits per sample, producing a 64 kbps digital data stream. A codec is the device that performs the conversion from analog voice into a digital format and vice-versa. The standard method used in traditional telephony is PCM (pulse code modulation) implemented by using a codec that conforms to ITU-T standard G.711. Most humans can hear sound up to about 20 KHz, but the traditional telephony uses low-pass filtering to remove everything but approximately the lower 4 KHz of the speech signal. In addition to this, voice over packet networks commonly use low bit rate codecs for compressing the received noise. These low bit rate codecs preserve the parts of the speech that are of important to the human listener taking out those parts that are not of any importance such as silence, redundantly long words. This is generally known as perceptual coding and is used in a number of other areas too, such as MPEG-2 video compression, JPEG image compression and MP3 audio. Standardized codecs have been tested with multiple speakers and multiple languages. The results can be tabulated as below.
Packet loss is a constant problem in packet-based networks. In a circuit-switched network, all speech in a given conversation follows the same path and is received in the order in which it is transmitted. If something is lost, the cause is a fault rather than an inherent characteristic of the system.
Apart from these factors there could be impairments caused by codecs. These impairments are due to the distortion introduced by the codec and due to the interaction of network effects with codec operation. Speech coding and compression Both speech coding and compression have been used in the traditional telephony for over two decades. With the exception of the local loop, almost all voice is carried over the PSTN in digital format. The received analog voice undergoes an analog-digital conversion at 8000 samples per second with 8 bits per sample, producing a 64 kbps digital data stream. A codec is the device that performs the conversion from analog voice into a digital format and vice-versa. The standard method used in traditional telephony is PCM (pulse code modulation) implemented by using a codec that conforms to ITU-T standard G.711. Most humans can hear sound up to about 20 KHz, but the traditional telephony uses low-pass filtering to remove everything but approximately the lower 4 KHz of the speech signal. In addition to this, voice over packet networks commonly use low bit rate codecs for compressing the received noise. These low bit rate codecs preserve the parts of the speech that are of important to the human listener taking out those parts that are not of any importance such as silence, redundantly long words. This is generally known as perceptual coding and is used in a number of other areas too, such as MPEG-2 video compression, JPEG image compression and MP3 audio. Standardized codecs have been tested with multiple speakers and multiple languages. The results can be tabulated as below.
Here MOS is the measurement for voice clarity. This is explained in detail later in this chapter.Positive FactorsOf the two positive factors for VoIP performance, the first one is bandwidth, which is absolutely necessary for adequate performance. The second factor is prioritization. Prioritization becomes increasingly important as the network gets congested.
Bandwidth
One of the greatest challenges of VoIP is voice quality, and one of the keys to improving voice quality to an acceptable level is bandwidth. Therefore, additional bandwidth is certainly needed, if only to support additional traffic as demand for VoIP continues to grow. While additional bandwidth is a necessity for a network that is required to support voice in addition to the data traffic that is might have carried all along, additional bandwidth is not, by itself, a complete solution for the QoS issue.
PrioritizationOnce sufficient bandwidth is available to enable high-quality voice transfer, we need to control and prioritize access to the available bandwidth. As this regulation is not exerted over the Internet presently and because the IP is designed completely for the transfer of the data from its outset, depending upon the usage of the bandwidth, voice quality over the Internet might vary from acceptable to atrocious.
PrioritizationOnce sufficient bandwidth is available to enable high-quality voice transfer, we need to control and prioritize access to the available bandwidth. As this regulation is not exerted over the Internet presently and because the IP is designed completely for the transfer of the data from its outset, depending upon the usage of the bandwidth, voice quality over the Internet might vary from acceptable to atrocious.
Voice Quality & Quality of Service (QoS)
QoS is a collective measure of the level of service delivered to a user. QoS can be considered as the level of assurance for a particular application that the network can meet its service requirements. From a technical perspective, QoS can be characterized by several performance criteria, such as uptime, throughput, connection setup time, percentage of successful transmissions, speed of fault detection and correction, etc. In an IP network, QoS can be measured in terms of bandwidth, packet loss, delay, and jitter. In order to provide a high QoS, the IP network needs to provide assurances that for a given session or set of sessions, the measurement of these characteristics will fall within certain bounds. High quality over IP networks requires the use of managed networks, QoS solutions, and service-level agreements between the providers. Given the stringent delay requirement voice imposes, one should look at the avenues to achieve quality, reliability and scalability of traditional telephone networks, if they want to make VoIP a fierce competitor to the traditional telephony.
For organizations that are interested in deploying VoIP technology on their corporation Intranets or on their other networks, the success of these technologies will depend on the performance of the network elements that carry and route the voice packets। The users of VoIP are concerned about the possible voice quality degradation when voice is carried over these packet networks, as the existing Internet protocols do not support real time traffic. Voice quality is the crucial factor in making VoIP acceptable to users, and it is important to understand the factors that affect the quality of the voice over the packet transmission networks, as well as to obtain the tools and optimize them. Although speech quality is often cited as one of the greatest challenges facing the development and market acceptance of voice over packet networks, people may in fact accept ‘sub-toll quality’ voice in exchange for some other benefits such as mobility, reduced cost and other advanced services VoIP can offer.
QoS is a collective measure of the level of service delivered to a user. QoS can be considered as the level of assurance for a particular application that the network can meet its service requirements. From a technical perspective, QoS can be characterized by several performance criteria, such as uptime, throughput, connection setup time, percentage of successful transmissions, speed of fault detection and correction, etc. In an IP network, QoS can be measured in terms of bandwidth, packet loss, delay, and jitter. In order to provide a high QoS, the IP network needs to provide assurances that for a given session or set of sessions, the measurement of these characteristics will fall within certain bounds. High quality over IP networks requires the use of managed networks, QoS solutions, and service-level agreements between the providers. Given the stringent delay requirement voice imposes, one should look at the avenues to achieve quality, reliability and scalability of traditional telephone networks, if they want to make VoIP a fierce competitor to the traditional telephony.
For organizations that are interested in deploying VoIP technology on their corporation Intranets or on their other networks, the success of these technologies will depend on the performance of the network elements that carry and route the voice packets। The users of VoIP are concerned about the possible voice quality degradation when voice is carried over these packet networks, as the existing Internet protocols do not support real time traffic. Voice quality is the crucial factor in making VoIP acceptable to users, and it is important to understand the factors that affect the quality of the voice over the packet transmission networks, as well as to obtain the tools and optimize them. Although speech quality is often cited as one of the greatest challenges facing the development and market acceptance of voice over packet networks, people may in fact accept ‘sub-toll quality’ voice in exchange for some other benefits such as mobility, reduced cost and other advanced services VoIP can offer.
Approach:
VIPER
To investigate the quality matters of Voice over IP a project named ‘Voice over Internet Protocol Environment for Research (VIPER)’ has been undertaken which could enable network-integrated, controllable, and statistically valid end-to-end measurements of VoIP quality. This system also enables specifications for vendors about ideal network configurations to obtain better voice quality over the Internet. Following sections discuss the specific motivations and architecture of the VIPER system. Results from VIPER testing are presented in subsequent chapters.
To investigate the quality matters of Voice over IP a project named ‘Voice over Internet Protocol Environment for Research (VIPER)’ has been undertaken which could enable network-integrated, controllable, and statistically valid end-to-end measurements of VoIP quality. This system also enables specifications for vendors about ideal network configurations to obtain better voice quality over the Internet. Following sections discuss the specific motivations and architecture of the VIPER system. Results from VIPER testing are presented in subsequent chapters.
Motivation
In performing this work, we would like to be able to determine the impact of design and environmental changes (e.g. network conditions, such as packet loss) on voice quality. The ability to quantify voice quality is important for a number of reasons. First, we would like to compare the quality of voice over packet networks to the PSTN, as the PSTN has become the de facto standard for what constitutes acceptable voice quality. We would also be able to test the effectiveness of various network protocols and policies that are known to support real time traffic. Lastly, from a business perspective, measurements of voice quality allow a vendor to offer better features than those of its competitors, as well as to provide the basis for voice quality service level agreements (SLA).
Voice quality could be measured using a procedure called Mean Opinion Scores (MOS). The MOS uses the Absolute Category Rating (ACR) procedure to determine the general acceptability or quality of voice communication systems or products. A MOS measurement is made by having a group of listeners rank a speech sample on a scale of 1-5, where 1 is very bad, 5 is excellent and 4 is normally considered ‘toll-quality’ (what one hears on the Public Switched Telecommunications Network (PSTN)). Obviously MOS is highly subjective and not highly reproducible. It is difficult to assemble a group of people, creation of ideal test facilities, selection of proper sound files, assembling audio devices and it is not suitable for long-term measurement
To address the shortcomings of this subjective testing a number of methods have been developed to create an objective and reproducible measurement of perceived voice quality. There are two clarity measurements currently used, the first one is PSQM (Perceptual Speech Quality Measurement) developed by KPN Research and the second one is PAMS (Perceptual Analysis/Measurement System) developed by British Telecom. Both these techniques use natural speech or speech-like samples as their inputs. The speech samples are played over the network that is setup for different configurations and the received speech sample is compared with the original speech sample using clarity algorithms.
In performing this work, we would like to be able to determine the impact of design and environmental changes (e.g. network conditions, such as packet loss) on voice quality. The ability to quantify voice quality is important for a number of reasons. First, we would like to compare the quality of voice over packet networks to the PSTN, as the PSTN has become the de facto standard for what constitutes acceptable voice quality. We would also be able to test the effectiveness of various network protocols and policies that are known to support real time traffic. Lastly, from a business perspective, measurements of voice quality allow a vendor to offer better features than those of its competitors, as well as to provide the basis for voice quality service level agreements (SLA).
Voice quality could be measured using a procedure called Mean Opinion Scores (MOS). The MOS uses the Absolute Category Rating (ACR) procedure to determine the general acceptability or quality of voice communication systems or products. A MOS measurement is made by having a group of listeners rank a speech sample on a scale of 1-5, where 1 is very bad, 5 is excellent and 4 is normally considered ‘toll-quality’ (what one hears on the Public Switched Telecommunications Network (PSTN)). Obviously MOS is highly subjective and not highly reproducible. It is difficult to assemble a group of people, creation of ideal test facilities, selection of proper sound files, assembling audio devices and it is not suitable for long-term measurement
To address the shortcomings of this subjective testing a number of methods have been developed to create an objective and reproducible measurement of perceived voice quality. There are two clarity measurements currently used, the first one is PSQM (Perceptual Speech Quality Measurement) developed by KPN Research and the second one is PAMS (Perceptual Analysis/Measurement System) developed by British Telecom. Both these techniques use natural speech or speech-like samples as their inputs. The speech samples are played over the network that is setup for different configurations and the received speech sample is compared with the original speech sample using clarity algorithms.
General architecture
VIPER includes network facilities for testing, a speech data repository that contains voice files, facilities for capturing subjective data, and scripts for the analysis of this gathered subjective data. Before the data can be collected and analyzed from VIPER, certain parameters are specified such as the architectural issues that could affect end-to-end subjective performance, such as bandwidth allocation, prioritization schemes, etc. The network facilities along with certain configuration parameters combine to create a test scenario. Untrained listeners had participated in these test scenarios.
The repeated exercise of the VIPER realization with multiple listeners can produce statistically meaningful insights into VoIP QoS. Thus, a significant benefit of the globally organized and integrated VIPER environment is, test-specific QoS parameters can be applied to the network elements. These parameters in effect transport, prioritize, and differentiate the service between the test data and/or noise traffic through the network.
VIPER includes network facilities for testing, a speech data repository that contains voice files, facilities for capturing subjective data, and scripts for the analysis of this gathered subjective data. Before the data can be collected and analyzed from VIPER, certain parameters are specified such as the architectural issues that could affect end-to-end subjective performance, such as bandwidth allocation, prioritization schemes, etc. The network facilities along with certain configuration parameters combine to create a test scenario. Untrained listeners had participated in these test scenarios.
The repeated exercise of the VIPER realization with multiple listeners can produce statistically meaningful insights into VoIP QoS. Thus, a significant benefit of the globally organized and integrated VIPER environment is, test-specific QoS parameters can be applied to the network elements. These parameters in effect transport, prioritize, and differentiate the service between the test data and/or noise traffic through the network.
User Interface
To enable the VIPER testing, a web-based user-interface was created, which includes the following features:
To enable the VIPER testing, a web-based user-interface was created, which includes the following features:
Network Setup
The network will be configured for different configurations using a tool called Expect, which can automate interactive applications such as telnet, ftp, rlogin, etc. The different network configurations used in VIPER realization are explained in the next chapter. After the quality is measured for a particular configuration, the network will be reconfigured into a default state.
The network will be configured for different configurations using a tool called Expect, which can automate interactive applications such as telnet, ftp, rlogin, etc. The different network configurations used in VIPER realization are explained in the next chapter. After the quality is measured for a particular configuration, the network will be reconfigured into a default state.
Generation of Noise
Data enough to congest the network in the form of noise is injected into the network. This noise is generated using a tool called Iperf.
Data enough to congest the network in the form of noise is injected into the network. This noise is generated using a tool called Iperf.
Playing of voice files
Vgetty is the tool used to generate the calls and to pump the voice files into the network. Vgetty is controlled through the web-based interface using the vgetty::modem, which is perl module and allows the control of voice modem through a perl script.
Vgetty is the tool used to generate the calls and to pump the voice files into the network. Vgetty is controlled through the web-based interface using the vgetty::modem, which is perl module and allows the control of voice modem through a perl script.
Voice quality rating
The test takers rate the quality of voice on a scale of 1 to 5 for the entire possible network configuration that were been designed. These results will be stored into a database for further analysis.
The test takers rate the quality of voice on a scale of 1 to 5 for the entire possible network configuration that were been designed. These results will be stored into a database for further analysis.
Technologies used in the creation of User Interface
Database (MySQL)
MySQL is one of the most popular SQL servers right now. SQL is programming language developed by IBM in 1970s and after which popularized as industry-standard language for creating, updating and querying the relational database management systems (RDBMS). MySQL is very fast, multi-threaded and multi-user SQL server. Using MySQL a database is created which contains a table to place the MOS values that are gathered from various end-users.
MySQL is one of the most popular SQL servers right now. SQL is programming language developed by IBM in 1970s and after which popularized as industry-standard language for creating, updating and querying the relational database management systems (RDBMS). MySQL is very fast, multi-threaded and multi-user SQL server. Using MySQL a database is created which contains a table to place the MOS values that are gathered from various end-users.
Web-based scripting (PHP)
PHP is used to create the dynamic web pages essential for carrying out the requirements of this project. Data collected from various users is inserted into the database using the PHP scripts. PHP enables the administrator to access and administer the database. The code for network configuration, noise generation, call generation and playing the voice files are also included into the overall PHP script. So in the whole the PHP scripts drive the whole VIPER implementation.
PHP is used to create the dynamic web pages essential for carrying out the requirements of this project. Data collected from various users is inserted into the database using the PHP scripts. PHP enables the administrator to access and administer the database. The code for network configuration, noise generation, call generation and playing the voice files are also included into the overall PHP script. So in the whole the PHP scripts drive the whole VIPER implementation.
Expect scripts for configuration
The routers are remotely telnetted using the expect script. Expect can make such things relatively easy through its powerful automation capabilities. In this project one expect script file is created for each network configuration, which virtually contains everything such as IP address of the router that needs to be accessed, password for accessing that router and router commands for that particular configuration.
The routers are remotely telnetted using the expect script. Expect can make such things relatively easy through its powerful automation capabilities. In this project one expect script file is created for each network configuration, which virtually contains everything such as IP address of the router that needs to be accessed, password for accessing that router and router commands for that particular configuration.
Vgetty
Vgetty is a Unix package, which coupled with mgetty (used to handle both the incoming and outgoing calls without any interference between them) can make the voice modem to send and receive the voice messages like an auto responder.
Vgetty is a Unix package, which coupled with mgetty (used to handle both the incoming and outgoing calls without any interference between them) can make the voice modem to send and receive the voice messages like an auto responder.
Operation
The network will initially be in a default state. As the listener proceeds through a test scenario ambient traffic will be generated through the network and the network will be configured through some scripts to represent that particular scenario. Noise will also be injected into the network, if it is needed for that particular test scenario. After that the user clicks on a hyperlink created on the PHP page, which triggers the vgetty package to play a randomly chosen voice file from the speech data repository and this will be played to that listener. As a general testing methodology, a voice file will be played over the speech path, encountering various types of impairments during encoding/decoding, packetization, and transmission. Depending upon the voice quality the listener has perceived, he/she is going to rate the quality of the voice on a scale of 1 to 5. This result will be collected into the MySQL database using the PHP script and will be placed into it across his/her name and time and date the user took the exam. This particular procedure will be repeated with many listeners for all the QoS architectures. This information will be will be analyzed for all the network configurations that are of interest.
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FAQ:
10 Questions To Ask When Buying A VoIP Gateway
Gateways are a crucial part of a successful VoIP system. Here are the questions to ask before you buy.There's no question that a gateway is a crucial component in any business VoIP system. That's because the gateway handles the fundamental task of transferring voice or fax traffic from a PBX to the IP network while supporting service levels equal to or exceeding the performance of traditional telephone technology.
Yet, with vendors offering dozens of VoIP different gateway models--each with a different feature set--finding the appropriate device for your enterprise's VoIP network can be hard work. Selecting a gateway that doesn't meet your VoIP network's needs can lead to all sorts of problems, including poor voice quality, inadequate management tools, difficult or impossible interoperability with other system devices and so on.
You can sidestep many of these issues at the outset by closely questioning each vendor about its products and policies. Here's a quick list of the key questions you should ask:
Yet, with vendors offering dozens of VoIP different gateway models--each with a different feature set--finding the appropriate device for your enterprise's VoIP network can be hard work. Selecting a gateway that doesn't meet your VoIP network's needs can lead to all sorts of problems, including poor voice quality, inadequate management tools, difficult or impossible interoperability with other system devices and so on.
You can sidestep many of these issues at the outset by closely questioning each vendor about its products and policies. Here's a quick list of the key questions you should ask:
1. How Much Does the Gateway Cost?
This should be your first question. You'll need to know the gateway's cost in order to match it against products with similar feature sets. Be sure to factor in any additional support costs.
2. Is the Gateway Hardware- or Software-based?
Most businesses buy hardware-based gateways because they're perceived to offer stronger security protection, are more reliable, don't rob computing power and provide better connectivity. The software type, on the other hand, tend to cost less and can be easier to update and modify. The choice is up to you.
3. What is the Chassis Size?
If you're considering the purchase of a hardware gateway, the unit's chassis size is crucial since it typically dictates the product's packet processing capacity. Sluggish processing leads to poor VoIP call quality, user complaints and, potentially, lost business. So be sure to purchase a gateway that can accommodate your VoIP's system's current call load as well as planned future growth. Which leads us to the next question...
4. How Many Simultaneous VoIP Calls Can the Gateway Handle?
It's important to select a product that can keep pace with the network's call load. A good rule of thumb is to purchase a gateway that can handle a call load that's at least 20 percent greater than existing traffic levels.
5. How Many Foreign Exchange Office (FXO) Ports Are Provided?
VoIP gateways convert the PSTN signal to a VoIP signal. For analog lines, an FXO port is needed. Until recently, most reasonably priced ($300 to $500) VoIP gateways had only one or two FXO ports--sufficient for home use, but too few for most small businesses and remote offices. Make sure that any gateway you're planning to buy at has at least four FXO ports.
6. What Type of IP Connectivity is Included?
Today, there are two major standards-based protocols available to establish and maintain VoIP connections: the ITU-T H.323 specification and the Session Initiation Protocol (SIP). These protocols provide the functions that allow end users to place and receive VoIP calls. Your new gateway, obviously, needs to handle whatever standards are used on your VoIP network.
7. How Are Voice Digitization and Compression Accomplished?
The key task required of any gateway is to convert the analog voice signal into the digital format that allows it to be transferred through a digital network. Usually, digitization results in a 64 kbps data rate. VoIP gateways can further compress voice call data rates to 24 kbps to 5.3 kbps per call. For maximum control over bandwidth usage and quality of service (QoS), you want as must flexibility as possible in compression rates.
8. What Are the Upgrade Options?
As your VoIP system grows, you'll probably need additional ports and other features. There's also always the possibility of new VoIP standards appearing over time, meaning your gateway will have to keep pace with the new technology.
Some gateways are more or less set in stone, and are virtually impossible to upgrade. Others products offer various levels of upgradeability. It's important to know which type of gateway you're looking at, since an upgradeable system may save you a significant amount of money in the years ahead.
Some gateways are more or less set in stone, and are virtually impossible to upgrade. Others products offer various levels of upgradeability. It's important to know which type of gateway you're looking at, since an upgradeable system may save you a significant amount of money in the years ahead.
9. Is the Gateway Compatible With My VoIP System Hardware?
he VoIP gateway needs to interoperate with a number of existing and future technologies, such as private branch exchange (PBX), automatic call director (ACD) and interactive voice response (IVR) systems. It's most important to find out if the trunk circuit port types on your PBX match those that are available on the gateway.
10. What Type of Support Do You Offer?
Beyond the gateway itself, you need to discover the level of support the vendor offers. How long is the basic warranty? Does the vendor provide phone- or e-mail-based troubleshooting? Is on-site support available? What is the cost of such services? These are all key things to know before you make your final purchase decision.
Selecting the right gateway for your VoIP system can be a confusing and time consuming process. But asking the right questions will ensure that you get the product and support you need at a price you can afford.
Selecting the right gateway for your VoIP system can be a confusing and time consuming process. But asking the right questions will ensure that you get the product and support you need at a price you can afford.
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