Veraz's I Gate SIP VOIP Gateway Deployed by 013 NetVision
Introduction
SIP gateways are devices that serve as translators for VOIP networks. They help companies in converting voice or fax calls into digital data packets that are compatible with the enterprise's VOIP system from a normal telephone. There are many VOIP providers that offer the best SIP VOIP services to their customers.
SIP gateways are also used by enterprises to direct calls to individuals with the help of built in routing tables. These gateways play a very vital role in the telecommunication industry. Recently Veraz Networks, who is leading provider of VOIP applications and optimization of bandwidth products, announced that 013 NetVision, an Israeli telecommunication industry has selected Veraz's SIP gateways to offer value added solutions to their customers.
Company Profile
Veraz Networks offers a wide range of facilities and applications to their customers. Their main objective is to manage, create and transport VOIP services in a cost effective manner to mobile and fixed service providers. They are well known for offering the best SIP VOIP applications and solutions to the top companies in the world.
NetVision on the other hand, offers value added and communication service to their business clients. Utilizing the services of NetVision business enterprises can easily reduce on their operational costs by making domestic and international calls at affordable rates. Like Veraz Networks, this company also has a good reputation for offering the best VOIP services to their customers.
What is the Veraz's I Gate 4000 Edge SIP Gateway?
The purchase decision about any product or service depends upon a detailed research about the product. Many of us may wonder what is the use of these SIP gateways and how will these benefit our company. Some of the benefits of this application are as follows:
It helps companies to reduce their operational costs and increase their revenues
These best VOIP application offer superior and high quality voice an efficient handling of fax calls
The I Gate 4000 Edge SIP Gateway is user friendly and can be easily deployed to small, medium or big enterprises
Companies by making use of these services and communicate with different clients throughout the world
Mr. Tovi Canor, the Vice President of 013 NetVision commented that partnering with Veraz has its own advantages. The offer the best VOIP services that meet with the challenges and requirements of the customers. This is an ideal tool for enterprises looking for expanding their businesses. He further stated that Veraz's aims at offering innovative and creative applications and technologies to their business customers.
These best SIP VOIP applications can easily fit into the budget of small businesses that are looking at diversifying their businesses. Making use of these services will give them an opportunity to attract more clients and also provide an edge over their competitors.
Saturday, August 15, 2009
How Contact Centers Can Utilize VoIP Call Recording Solutions
How Contact Centers Can Utilize VoIP Call Recording Solutions
Contact centers are becoming increasingly distributed and are beginning to leverage hosted VoIP services to drive efficiency, automation, and distribution into their communications infrastructure. However, due to quality management, regulatory and legal considerations, most contact centers are finding that they cannot completely outsource their communications to a hosted VoIP provider because they cannot maintain features such as call recording. Customer premise based recorders require complicated networking changes for the customer and are not able to handle the increasingly distributed nature of the contact center.
VoIP communications relies on the transmission of "packets" that contain content of the call. One VoIP call can contain numerous packets and in the course of transmission from one origination to terminating destination, the packets are generally received in random order. However, each packet contains information in a header, payload, and trailer section relating to the source & destination, the actual content of the call (data), and information identifying the order in which the packets should appear so that the transmission makes sense.
The information contained in the packets that identify the order in which they are to appear, or the "instructions", are also known as protocols. In very simple forms, protocols can appear individually. However, more commonly, protocols appear as stacks, multiple layers of protocols that work together to form the resulting outcome.
The most widely used protocols for VoIP communications today are H.323 and SIP. Although, all voice and video communication is generally performed over a separate protocol known as RTP (Real-Time Transport Protocol). RTP defines the standardized packet format for delivering the aforementioned type of communications over the internet.
In the past, when customers replaced their PBXs every 3-7 years or so, the expense of new physical equipment was a given. But now, with the entrance of VoIP onto the scene, it is possible to deliver all the functionality of hardware-based, key systems for much cheaper-using software. In order to compete in this new marketplace, service providers need to be able to offer hosted solutions to their customers-solutions that require little or no up-front investment, but instead are paid for through an ongoing contract. It is also important, to compete in the VoIP marketplace, to continue to provide customers with services that were once important in traditional telephony in the VoIP world as well.
Examples of such service are call accounting and call recording. Contact centers want applications that can record calls for quality management, regulatory and legal compliance issues. Better yet, contact centers want hosted call accounting and call recording products that meet their needs and cost them little or no upfront capital investment.
In a hosted, call recording scenario, VoIP extensions that require recording are registered on the Service Provider's switch. A VoIP call is placed by the contact center agent and a SIP invite is sent to the switch. The switch recognizes that that traffic for this extension is to be recorded and redirects the SIP invite to the call recording server. The SIP invite is then sent back to the switch and either forwarded on to the terminating VoIP address or to the PSTN (Public Switched Telephone Network) for termination to an off-network user. Once the SIP invitation is accepted, packets of data containing the call content begin to be exchanged.
This call content is exchanged in the form of RTP packets. These RTP packets are passed from the call originator, in this case the agent, to the call recording server where they are recorded and passed on to the termination point. Call content from the termination point similarly passes from the termination point to the call recording server to the originator. The call recording server transcodes the call content into an MP3 (Mpeg 2.5 layer III). These MP3 files are then sent to a file storage/web server for access via a browser-based application.
The browser-based application is able to display traffic for calls currently in progress as well as calls that are stored on the file server. A feature rich browser-based application will offer the ability to display calls based upon user permissions as well as offer features such as call descriptions and filtering. The browser application should also offer the tenant the ability to decide parameters and thresholds for call recording such as by hour, day, number, or a percentage of call traffic.
VoIP call recording options exist in several forms. The purpose of this paper is to address those platforms offered by the hosted, VoIP Service Provider. Along with packet routing call recording technologies outlined earlier, technologies exist which involve the practice of capturing or "sniffing" data packets as they travel through the Service Provider's network. In this technology, all data packets carrying signals or content are reviewed by the call recording application. As each packet is touched, the call recording application decides if the packet is to be recorded. This technology can be challenging for the hosted Service Provider due to the inherent scalability problems of touching every pack.
Additionally, situations exist for aggregators of VoIP services who only wish to offer call recording services to Service Providers who have purchased the service.
Challenges also arise in recording extension to extension calls placed within the tenant as these calls do not typically leave the customers network and are not routed through the Service Provider's switch.
Call recording is an integral part of the contact center environment as well as required for many regulatory and legal purposes. VoIP service providers offering value added services including call recording expand their reach into lucrative vertical markets and retain customers.
Contact centers are becoming increasingly distributed and are beginning to leverage hosted VoIP services to drive efficiency, automation, and distribution into their communications infrastructure. However, due to quality management, regulatory and legal considerations, most contact centers are finding that they cannot completely outsource their communications to a hosted VoIP provider because they cannot maintain features such as call recording. Customer premise based recorders require complicated networking changes for the customer and are not able to handle the increasingly distributed nature of the contact center.
VoIP communications relies on the transmission of "packets" that contain content of the call. One VoIP call can contain numerous packets and in the course of transmission from one origination to terminating destination, the packets are generally received in random order. However, each packet contains information in a header, payload, and trailer section relating to the source & destination, the actual content of the call (data), and information identifying the order in which the packets should appear so that the transmission makes sense.
The information contained in the packets that identify the order in which they are to appear, or the "instructions", are also known as protocols. In very simple forms, protocols can appear individually. However, more commonly, protocols appear as stacks, multiple layers of protocols that work together to form the resulting outcome.
The most widely used protocols for VoIP communications today are H.323 and SIP. Although, all voice and video communication is generally performed over a separate protocol known as RTP (Real-Time Transport Protocol). RTP defines the standardized packet format for delivering the aforementioned type of communications over the internet.
In the past, when customers replaced their PBXs every 3-7 years or so, the expense of new physical equipment was a given. But now, with the entrance of VoIP onto the scene, it is possible to deliver all the functionality of hardware-based, key systems for much cheaper-using software. In order to compete in this new marketplace, service providers need to be able to offer hosted solutions to their customers-solutions that require little or no up-front investment, but instead are paid for through an ongoing contract. It is also important, to compete in the VoIP marketplace, to continue to provide customers with services that were once important in traditional telephony in the VoIP world as well.
Examples of such service are call accounting and call recording. Contact centers want applications that can record calls for quality management, regulatory and legal compliance issues. Better yet, contact centers want hosted call accounting and call recording products that meet their needs and cost them little or no upfront capital investment.
In a hosted, call recording scenario, VoIP extensions that require recording are registered on the Service Provider's switch. A VoIP call is placed by the contact center agent and a SIP invite is sent to the switch. The switch recognizes that that traffic for this extension is to be recorded and redirects the SIP invite to the call recording server. The SIP invite is then sent back to the switch and either forwarded on to the terminating VoIP address or to the PSTN (Public Switched Telephone Network) for termination to an off-network user. Once the SIP invitation is accepted, packets of data containing the call content begin to be exchanged.
This call content is exchanged in the form of RTP packets. These RTP packets are passed from the call originator, in this case the agent, to the call recording server where they are recorded and passed on to the termination point. Call content from the termination point similarly passes from the termination point to the call recording server to the originator. The call recording server transcodes the call content into an MP3 (Mpeg 2.5 layer III). These MP3 files are then sent to a file storage/web server for access via a browser-based application.
The browser-based application is able to display traffic for calls currently in progress as well as calls that are stored on the file server. A feature rich browser-based application will offer the ability to display calls based upon user permissions as well as offer features such as call descriptions and filtering. The browser application should also offer the tenant the ability to decide parameters and thresholds for call recording such as by hour, day, number, or a percentage of call traffic.
VoIP call recording options exist in several forms. The purpose of this paper is to address those platforms offered by the hosted, VoIP Service Provider. Along with packet routing call recording technologies outlined earlier, technologies exist which involve the practice of capturing or "sniffing" data packets as they travel through the Service Provider's network. In this technology, all data packets carrying signals or content are reviewed by the call recording application. As each packet is touched, the call recording application decides if the packet is to be recorded. This technology can be challenging for the hosted Service Provider due to the inherent scalability problems of touching every pack.
Additionally, situations exist for aggregators of VoIP services who only wish to offer call recording services to Service Providers who have purchased the service.
Challenges also arise in recording extension to extension calls placed within the tenant as these calls do not typically leave the customers network and are not routed through the Service Provider's switch.
Call recording is an integral part of the contact center environment as well as required for many regulatory and legal purposes. VoIP service providers offering value added services including call recording expand their reach into lucrative vertical markets and retain customers.
Call India through VoIP and stop worrying about heavy bills
Call India through VoIP and stop worrying about heavy bills
Gone are those days when making calls to the distant and remote places were a kind of great trouble for everyone who has friends and families staying at these places. As they were not been able to have a tension free conversation with them, because of this many steps have been taken to provide the users some hassle free calls. And the service providers have come up with many new economical calling plans. So, now you can call India without being bothered about the lengthy phone bills that take your breath away.
This is why calling India with this brand new way is a smarter option to make calls. Within very short span of time, it has gained tremendous response from its customers as they are very much happy to find this mode of calling to distant places. This is the reason that people now feel easy to call India. It is because India is a place where there are a lot of business as well as professionals opportunities. Therefore, anybody who is interested to make calls to India always look for such plans that can cut the rate of calling and for those people VoIP technology has come up to gift us some hassle free calls.
The introduction of VoIP has changed the face of telecom industry totally. That is why now it has been treated as a tool of prime importance in connecting people across the globe. This brand new tool is playing a vital role in strengthening the relationships among people separated by the distances. With the VoIP technology, it has now become very easy to call India from Dubai without any hassles. It is basically a digital technology that uses satellite to forward calls to the end users in the form of digital packets. It works quite well for even the people staying at a remote place.
India has plenty number of call centers providing the customer care support based in India. That is why, day by day, calling India for free is earning popularity. There are number of websites available, in order to facilitate the VoIP calls to India. This specific feature is available on many websites and the messengers of many popular portals and you can call a person if he or she is online. But, the users are highly crazy about the VoIP calls. The Internet portals also offer many exciting offers to the users of VoIP.
The VoIP calls is getting popularised in India with time and it is quite obvious as the calls for which you have to spend a lot in the past, can now be forwarded without even paying a single penny from your pocket. Some of the websites charge very negligible charges which is quite affordable. Making calls without paying anything is definitely a beneficial deal for people, as it not only helps the people in their personal talks, but one can also gets benefits from it in their business purposes. More and more websites are entering in this area to be a part of this great deal as it not only helps them providing a good service to the potential users but also give the service provider as well as the web portals to earn a good share of profit.
Gone are those days when making calls to the distant and remote places were a kind of great trouble for everyone who has friends and families staying at these places. As they were not been able to have a tension free conversation with them, because of this many steps have been taken to provide the users some hassle free calls. And the service providers have come up with many new economical calling plans. So, now you can call India without being bothered about the lengthy phone bills that take your breath away.
This is why calling India with this brand new way is a smarter option to make calls. Within very short span of time, it has gained tremendous response from its customers as they are very much happy to find this mode of calling to distant places. This is the reason that people now feel easy to call India. It is because India is a place where there are a lot of business as well as professionals opportunities. Therefore, anybody who is interested to make calls to India always look for such plans that can cut the rate of calling and for those people VoIP technology has come up to gift us some hassle free calls.
The introduction of VoIP has changed the face of telecom industry totally. That is why now it has been treated as a tool of prime importance in connecting people across the globe. This brand new tool is playing a vital role in strengthening the relationships among people separated by the distances. With the VoIP technology, it has now become very easy to call India from Dubai without any hassles. It is basically a digital technology that uses satellite to forward calls to the end users in the form of digital packets. It works quite well for even the people staying at a remote place.
India has plenty number of call centers providing the customer care support based in India. That is why, day by day, calling India for free is earning popularity. There are number of websites available, in order to facilitate the VoIP calls to India. This specific feature is available on many websites and the messengers of many popular portals and you can call a person if he or she is online. But, the users are highly crazy about the VoIP calls. The Internet portals also offer many exciting offers to the users of VoIP.
The VoIP calls is getting popularised in India with time and it is quite obvious as the calls for which you have to spend a lot in the past, can now be forwarded without even paying a single penny from your pocket. Some of the websites charge very negligible charges which is quite affordable. Making calls without paying anything is definitely a beneficial deal for people, as it not only helps the people in their personal talks, but one can also gets benefits from it in their business purposes. More and more websites are entering in this area to be a part of this great deal as it not only helps them providing a good service to the potential users but also give the service provider as well as the web portals to earn a good share of profit.
Tuesday, August 11, 2009
OneSuite VOIP Launched to Broadcast Audio Messages
OneSuite VOIP Launched to Broadcast Audio Messages
Want to organize a function, party or any sort of event? Tired of calling or sending multiple messages to hundreds of people? All these methods are time consuming and also expensive. Wouldn't it be great it things very a lot easier. Well you no longer have to worry about all such things because OneSuite VOIP has come out with the best VOIP application known as SuiteAdvantage.
What is SuiteAdvantage?
SuiteAdvantage is a VOIP application that allows users to broadcast audio messages without having to make multiple calls. You just need to subscribe for this wonderful service to send your messages worldwide. This is one of the best VOIP services that users can avail without any fuss or hassles.
In addition to the above this service allows users to send audio messages to around 100 of people simultaneously either to domestic or international locations at a touch of a button. What more can you ask for. This is a user friendly service that offers high quality voice calls.
How to Use SuiteAdvantage?
SuiteAdvantage is very easy to use. All you need to have is a PC and an internet connection. This best VOIP application can be used by following these simple steps:
First of all you need to create an account. Then you need to log in to your account, click on SuiteAdvantage followed by the features option
Go to the Voicemail option and click enabled
After this step you need to create your distribution groups to whom you will be sending frequent messages. The best part of this best VOIP application is that the telephone numbers may either be SuiteAdvantage, international or domestic numbers
Use your SoftPhone OneSuite application and dial 0 preceded by * (*0). This will automatically direct you to the voicemail menu
Messages can be sent by dialing 2
Record your message and when prompted, enter the numbers created in your distributed groups
The messages are then sent to your distribution groups. If the users don't pick up their calls, they are automatically directed to their voicemail inbox
By following these simple steps, multiple messages can be broadcasted to hundreds of people in a matter of minutes. This is indeed the best VOIP services for those who need to frequently organize events or occasions.
Conclusion
Using this best VOIP application can help you save loads of money. You no longer have to send repeated messages or make repeated calls. The price for this service is $2.95 per month.
Want to organize a function, party or any sort of event? Tired of calling or sending multiple messages to hundreds of people? All these methods are time consuming and also expensive. Wouldn't it be great it things very a lot easier. Well you no longer have to worry about all such things because OneSuite VOIP has come out with the best VOIP application known as SuiteAdvantage.
What is SuiteAdvantage?
SuiteAdvantage is a VOIP application that allows users to broadcast audio messages without having to make multiple calls. You just need to subscribe for this wonderful service to send your messages worldwide. This is one of the best VOIP services that users can avail without any fuss or hassles.
In addition to the above this service allows users to send audio messages to around 100 of people simultaneously either to domestic or international locations at a touch of a button. What more can you ask for. This is a user friendly service that offers high quality voice calls.
How to Use SuiteAdvantage?
SuiteAdvantage is very easy to use. All you need to have is a PC and an internet connection. This best VOIP application can be used by following these simple steps:
First of all you need to create an account. Then you need to log in to your account, click on SuiteAdvantage followed by the features option
Go to the Voicemail option and click enabled
After this step you need to create your distribution groups to whom you will be sending frequent messages. The best part of this best VOIP application is that the telephone numbers may either be SuiteAdvantage, international or domestic numbers
Use your SoftPhone OneSuite application and dial 0 preceded by * (*0). This will automatically direct you to the voicemail menu
Messages can be sent by dialing 2
Record your message and when prompted, enter the numbers created in your distributed groups
The messages are then sent to your distribution groups. If the users don't pick up their calls, they are automatically directed to their voicemail inbox
By following these simple steps, multiple messages can be broadcasted to hundreds of people in a matter of minutes. This is indeed the best VOIP services for those who need to frequently organize events or occasions.
Conclusion
Using this best VOIP application can help you save loads of money. You no longer have to send repeated messages or make repeated calls. The price for this service is $2.95 per month.
Microsoft Competes with Skype to Offer the Best VOIP Services to Their Customers
Microsoft Competes with Skype to Offer the Best VOIP Services to Their Customers
The main objective of any VOIP provider is to offer good quality and consistent services to their customers. In this world of competition it is necessary to meet with the growing needs and requirements of the customers. There are a lot of service providers who offer the best VOIP services at affordable prices. Recently Microsoft Office has signed up with Interoute to supply SIP trunking facilities to their customers.
Interoute Company Profile
Interoute is one of the top leaders in the European market. They offer a wide range of voice and data network services to more than 3 million customers worldwide. Some of their clients include government, research agencies and also to some well known universities. Many of the big shot companies choose Interoute since they offer the best SIP VOIP solutions in a cost effective manner.
What's so unique about Interoute Products or Services?
You may be wondering why Microsoft Office has opted for the services of Interoute. Well the answer is quite simple. Microsoft office is quite popular amongst customers. By combining the services of Interoute, companies can easily reduce on their operation costs and simultaneously offer best VOIP services to their clients.
The main advantage of using the services of this company is that enterprises or business firms can make roaming and international calls at incredibly low prices. In addition to this calls can be made to either landlines or any other devices. Some of the benefits of Interoute services are as follows:
By making use of these best SIP VOIP services offered by Microsoft Office can enable companies to shift to unified communications and other VOIP applications without any problems or hassles.
This also enables companies to make free VOIP calls from PC to PC. Calls to other devices are charged at affordable rates.
This is a golden opportunity for businesses to give head on competition to Skype since these best VOIP services assist firms to transfer to business unified communications
The Research Manager of European Telecoms Group Ms. Angela Salmeron confirmed that the SIP trunking solutions of Interoute have the ability to connect OCS users to PSTN. Theses services are totally flexible and can be easily adopted by firms who want to migrate over to VOIP.
Conclusion
Businesses by making use of these best SIP VOIP services can offer their customers with a rich and sophisticated communication experience. This is a great opportunity for enterprises to expand their services and increase the profits for their business.
The main objective of any VOIP provider is to offer good quality and consistent services to their customers. In this world of competition it is necessary to meet with the growing needs and requirements of the customers. There are a lot of service providers who offer the best VOIP services at affordable prices. Recently Microsoft Office has signed up with Interoute to supply SIP trunking facilities to their customers.
Interoute Company Profile
Interoute is one of the top leaders in the European market. They offer a wide range of voice and data network services to more than 3 million customers worldwide. Some of their clients include government, research agencies and also to some well known universities. Many of the big shot companies choose Interoute since they offer the best SIP VOIP solutions in a cost effective manner.
What's so unique about Interoute Products or Services?
You may be wondering why Microsoft Office has opted for the services of Interoute. Well the answer is quite simple. Microsoft office is quite popular amongst customers. By combining the services of Interoute, companies can easily reduce on their operation costs and simultaneously offer best VOIP services to their clients.
The main advantage of using the services of this company is that enterprises or business firms can make roaming and international calls at incredibly low prices. In addition to this calls can be made to either landlines or any other devices. Some of the benefits of Interoute services are as follows:
By making use of these best SIP VOIP services offered by Microsoft Office can enable companies to shift to unified communications and other VOIP applications without any problems or hassles.
This also enables companies to make free VOIP calls from PC to PC. Calls to other devices are charged at affordable rates.
This is a golden opportunity for businesses to give head on competition to Skype since these best VOIP services assist firms to transfer to business unified communications
The Research Manager of European Telecoms Group Ms. Angela Salmeron confirmed that the SIP trunking solutions of Interoute have the ability to connect OCS users to PSTN. Theses services are totally flexible and can be easily adopted by firms who want to migrate over to VOIP.
Conclusion
Businesses by making use of these best SIP VOIP services can offer their customers with a rich and sophisticated communication experience. This is a great opportunity for enterprises to expand their services and increase the profits for their business.
Sunday, August 9, 2009
Equipment Requirements at a Work Call Centre
Equipment Requirements at a Work Call Centre
You will need to be prepared to invest quite a bit of money on getting the right equipment for establishing yourself as a work-at-home call centre assistant. It is also important to be wary about buying the right sort of equipment and high quality gear at that. The equipment needs to be built to a high standard and expected to last a decent amount of time, so automatically going with the cheapest options can often turn out to be a false economy.
The most expensive and also most important requirement is to have a computer. If you want to be on the move and have a little more flexibility with regards to where you work, then a laptop can be a great option. However, price-wise, desktops are still cheaper and are generally capable of more as well as being more upgradable. With computers as cheap as they are nowadays, this is not likely to be a particularly daunting investment considering all of the other things that you can use a computer for too. Since most people will already have an adequate computer system, this is unlikely to even be an issue.
The Internet connection required can be a slightly demanding requirement in some cases. For a call centre, you will need to have an adequate amount of bandwidth so you can use the Internet telephony / VoIP programs that you will need. This basically rules out any older 56k modem connections. Cable modem and DSL are generally adequate but bear in mind, any employers you might have might not permit certain types of connections.
You will also need to be able to use the Internet as well as the telephone line at the same time. This is typically not an issue if you have a broadband connection which is already a requirement anyway. There will likely be various requirements for the type of telephony hardware that you need for the job, depending on your employer.
All of the above are probably things you have already. Since so many people these days have at least one computer in their household and broadband Internet access, these are unlikely to become extra expenditures. However, you may well find it preferable to have a dedicated computer which only you have access to. Using the primary family computer as your primary work station is generally not a good idea for security and reliability reasons.
You will also need to spend a bit of money on a decent headset. Something that has a high quality noise cancelling microphone is a must. However, these are not too hard to find and do not cost a great deal of money. In fact, there is unlikely to be any reason why you would need to spend more than about $25 on this.
Your working environment should also be taken into account. Your employer will most likely require that you have a separate room to work in and no interruptions such as noisy air conditioning units etc. Simply put, there must be minimal background noises in your work place.
You will need to be prepared to invest quite a bit of money on getting the right equipment for establishing yourself as a work-at-home call centre assistant. It is also important to be wary about buying the right sort of equipment and high quality gear at that. The equipment needs to be built to a high standard and expected to last a decent amount of time, so automatically going with the cheapest options can often turn out to be a false economy.
The most expensive and also most important requirement is to have a computer. If you want to be on the move and have a little more flexibility with regards to where you work, then a laptop can be a great option. However, price-wise, desktops are still cheaper and are generally capable of more as well as being more upgradable. With computers as cheap as they are nowadays, this is not likely to be a particularly daunting investment considering all of the other things that you can use a computer for too. Since most people will already have an adequate computer system, this is unlikely to even be an issue.
The Internet connection required can be a slightly demanding requirement in some cases. For a call centre, you will need to have an adequate amount of bandwidth so you can use the Internet telephony / VoIP programs that you will need. This basically rules out any older 56k modem connections. Cable modem and DSL are generally adequate but bear in mind, any employers you might have might not permit certain types of connections.
You will also need to be able to use the Internet as well as the telephone line at the same time. This is typically not an issue if you have a broadband connection which is already a requirement anyway. There will likely be various requirements for the type of telephony hardware that you need for the job, depending on your employer.
All of the above are probably things you have already. Since so many people these days have at least one computer in their household and broadband Internet access, these are unlikely to become extra expenditures. However, you may well find it preferable to have a dedicated computer which only you have access to. Using the primary family computer as your primary work station is generally not a good idea for security and reliability reasons.
You will also need to spend a bit of money on a decent headset. Something that has a high quality noise cancelling microphone is a must. However, these are not too hard to find and do not cost a great deal of money. In fact, there is unlikely to be any reason why you would need to spend more than about $25 on this.
Your working environment should also be taken into account. Your employer will most likely require that you have a separate room to work in and no interruptions such as noisy air conditioning units etc. Simply put, there must be minimal background noises in your work place.
Wednesday, August 5, 2009
How Contact Centers Can Utilize VoIP Call Recording Solutions
How Contact Centers Can Utilize VoIP Call Recording Solutions
Contact centers are becoming increasingly distributed and are beginning to leverage hosted VoIP services to drive efficiency, automation, and distribution into their communications infrastructure. However, due to quality management, regulatory and legal considerations, most contact centers are finding that they cannot completely outsource their communications to a hosted VoIP provider because they cannot maintain features such as call recording. Customer premise based recorders require complicated networking changes for the customer and are not able to handle the increasingly distributed nature of the contact center. VoIP communications relies on the transmission of "packets" that contain content of the call. One VoIP call can contain numerous packets and in the course of transmission from one origination to terminating destination, the packets are generally received in random order. However, each packet contains information in a header, payload, and trailer section relating to the source & destination, the actual content of the call (data), and information identifying the order in which the packets should appear so that the transmission makes sense. The information contained in the packets that identify the order in which they are to appear, or the "instructions", are also known as protocols. In very simple forms, protocols can appear individually. However, more commonly, protocols appear as stacks, multiple layers of protocols that work together to form the resulting outcome. The most widely used protocols for VoIP communications today are H.323 and SIP. Although, all voice and video communication is generally performed over a separate protocol known as RTP (Real-Time Transport Protocol). RTP defines the standardized packet format for delivering the aforementioned type of communications over the internet. In the past, when customers replaced their PBXs every 3-7 years or so, the expense of new physical equipment was a given. But now, with the entrance of VoIP onto the scene, it is possible to deliver all the functionality of hardware-based, key systems for much cheaper—using software. In order to compete in this new marketplace, service providers need to be able to offer hosted solutions to their customers—solutions that require little or no up-front investment, but instead are paid for through an ongoing contract. It is also important, to compete in the VoIP marketplace, to continue to provide customers with services that were once important in traditional telephony in the VoIP world as well. Examples of such service are call accounting and call recording. Contact centers want applications that can record calls for quality management, regulatory and legal compliance issues. Better yet, contact centers want hosted call accounting and call recording products that meet their needs and cost them little or no upfront capital investment. In a hosted, call recording scenario, VoIP extensions that require recording are registered on the Service Provider's switch. A VoIP call is placed by the contact center agent and a SIP invite is sent to the switch. The switch recognizes that that traffic for this extension is to be recorded and redirects the SIP invite to the call recording server. The SIP invite is then sent back to the switch and either forwarded on to the terminating VoIP address or to the PSTN (Public Switched Telephone Network) for termination to an off-network user. Once the SIP invitation is accepted, packets of data containing the call content begin to be exchanged. This call content is exchanged in the form of RTP packets. These RTP packets are passed from the call originator, in this case the agent, to the call recording server where they are recorded and passed on to the termination point. Call content from the termination point similarly passes from the termination point to the call recording server to the originator. The call recording server transcodes the call content into an MP3 (Mpeg 2.5 layer III). These MP3 files are then sent to a file storage/web server for access via a browser-based application. The browser-based application is able to display traffic for calls currently in progress as well as calls that are stored on the file server. A feature rich browser-based application will offer the ability to display calls based upon user permissions as well as offer features such as call descriptions and filtering. The browser application should also offer the tenant the ability to decide parameters and thresholds for call recording such as by hour, day, number, or a percentage of call traffic. VoIP call recording options exist in several forms. The purpose of this paper is to address those platforms offered by the hosted, VoIP Service Provider. Along with packet routing call recording technologies outlined earlier, technologies exist which involve the practice of capturing or "sniffing" data packets as they travel through the Service Provider's network. In this technology, all data packets carrying signals or content are reviewed by the call recording application. As each packet is touched, the call recording application decides if the packet is to be recorded. This technology can be challenging for the hosted Service Provider due to the inherent scalability problems of touching every pack. Additionally, situations exist for aggregators of VoIP services who only wish to offer call recording services to Service Providers who have purchased the service. Challenges also arise in recording extension to extension calls placed within the tenant as these calls do not typically leave the customers network and are not routed through the Service Provider's switch. Call recording is an integral part of the contact center environment as well as required for many regulatory and legal purposes. VoIP service providers offering value added services including call recording expand their reach into lucrative vertical markets and retain customers.
Contact centers are becoming increasingly distributed and are beginning to leverage hosted VoIP services to drive efficiency, automation, and distribution into their communications infrastructure. However, due to quality management, regulatory and legal considerations, most contact centers are finding that they cannot completely outsource their communications to a hosted VoIP provider because they cannot maintain features such as call recording. Customer premise based recorders require complicated networking changes for the customer and are not able to handle the increasingly distributed nature of the contact center. VoIP communications relies on the transmission of "packets" that contain content of the call. One VoIP call can contain numerous packets and in the course of transmission from one origination to terminating destination, the packets are generally received in random order. However, each packet contains information in a header, payload, and trailer section relating to the source & destination, the actual content of the call (data), and information identifying the order in which the packets should appear so that the transmission makes sense. The information contained in the packets that identify the order in which they are to appear, or the "instructions", are also known as protocols. In very simple forms, protocols can appear individually. However, more commonly, protocols appear as stacks, multiple layers of protocols that work together to form the resulting outcome. The most widely used protocols for VoIP communications today are H.323 and SIP. Although, all voice and video communication is generally performed over a separate protocol known as RTP (Real-Time Transport Protocol). RTP defines the standardized packet format for delivering the aforementioned type of communications over the internet. In the past, when customers replaced their PBXs every 3-7 years or so, the expense of new physical equipment was a given. But now, with the entrance of VoIP onto the scene, it is possible to deliver all the functionality of hardware-based, key systems for much cheaper—using software. In order to compete in this new marketplace, service providers need to be able to offer hosted solutions to their customers—solutions that require little or no up-front investment, but instead are paid for through an ongoing contract. It is also important, to compete in the VoIP marketplace, to continue to provide customers with services that were once important in traditional telephony in the VoIP world as well. Examples of such service are call accounting and call recording. Contact centers want applications that can record calls for quality management, regulatory and legal compliance issues. Better yet, contact centers want hosted call accounting and call recording products that meet their needs and cost them little or no upfront capital investment. In a hosted, call recording scenario, VoIP extensions that require recording are registered on the Service Provider's switch. A VoIP call is placed by the contact center agent and a SIP invite is sent to the switch. The switch recognizes that that traffic for this extension is to be recorded and redirects the SIP invite to the call recording server. The SIP invite is then sent back to the switch and either forwarded on to the terminating VoIP address or to the PSTN (Public Switched Telephone Network) for termination to an off-network user. Once the SIP invitation is accepted, packets of data containing the call content begin to be exchanged. This call content is exchanged in the form of RTP packets. These RTP packets are passed from the call originator, in this case the agent, to the call recording server where they are recorded and passed on to the termination point. Call content from the termination point similarly passes from the termination point to the call recording server to the originator. The call recording server transcodes the call content into an MP3 (Mpeg 2.5 layer III). These MP3 files are then sent to a file storage/web server for access via a browser-based application. The browser-based application is able to display traffic for calls currently in progress as well as calls that are stored on the file server. A feature rich browser-based application will offer the ability to display calls based upon user permissions as well as offer features such as call descriptions and filtering. The browser application should also offer the tenant the ability to decide parameters and thresholds for call recording such as by hour, day, number, or a percentage of call traffic. VoIP call recording options exist in several forms. The purpose of this paper is to address those platforms offered by the hosted, VoIP Service Provider. Along with packet routing call recording technologies outlined earlier, technologies exist which involve the practice of capturing or "sniffing" data packets as they travel through the Service Provider's network. In this technology, all data packets carrying signals or content are reviewed by the call recording application. As each packet is touched, the call recording application decides if the packet is to be recorded. This technology can be challenging for the hosted Service Provider due to the inherent scalability problems of touching every pack. Additionally, situations exist for aggregators of VoIP services who only wish to offer call recording services to Service Providers who have purchased the service. Challenges also arise in recording extension to extension calls placed within the tenant as these calls do not typically leave the customers network and are not routed through the Service Provider's switch. Call recording is an integral part of the contact center environment as well as required for many regulatory and legal purposes. VoIP service providers offering value added services including call recording expand their reach into lucrative vertical markets and retain customers.
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