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IP PBX, SIP & VOIP
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An IP PBX or VOIP phone system replaces a traditional PBX or phone system and gives employees an extension number, the ability to conference, transfer and dial other colleagues. All calls are sent via data packets over a data network instead of the traditional phone network. With the use of a VOIP gateway, you can connect existing phone lines to the IP PBX and make and receive phone calls via a regular PSTN line. Companies are switching their traditional phone systems / PBX systems to a VOIP phone system / IP PBX at a staggering rate: IP Telephony equipment sales are increasing each year by more than 50%.
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10 reasons to switch to an IP PBX
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Benefit #1: Much easier to install & configure than a proprietary phone system:
An IP PBX runs as software on a computer and can leverage the advanced processing power of the computer and user interface as well as Windows’ features. Anyone proficient in networking and computers can install and maintain an IP PBX. By contrast a proprietary phone system often requires an installer trained on that particular proprietary system!
Benefit #2: Easier to manage because of web/GUI based configuration interface:
An IP PBX can be managed via a web-based configuration interface or a GUI, allowing you to easily maintain and fine tune your phone system. Proprietary phone systems have difficult-to-use interfaces which are often designed to be used only by the phone technicians.
Benefit #3: Significant cost savings using VOIP providers:
With an IP PBX you can easily use a VOIP service provider for long distance and international calls. The monthly savings are significant. If you have branch offices, you can easily connect phone systems between branches and make free phone calls.
Benefit #4 Eliminate phone wiring!
An IP Telephone system allows you to connect hardware phones directly to a standard computer network port (which it can share with the a computer). Software phones can be installed directly onto the PC. You can now eliminate the phone wiring and make adding or moving of extensions much easier. In new offices you can completely eliminate the extra ports to be used by the office phone system!
Benefit #5: Eliminate vendor lock in!
IP PBXs are based on the open SIP standard. You can now mix and match any SIP hardware or software phone with any SIP-based IP PBX, PSTN Gateway or VOIP provider. In contrast, a proprietary phone system often requires proprietary phones to use advanced features, and proprietary extension modules to add features.
Benefit #6: Scalable
Proprietary systems are easy to outgrow: Adding more phone lines or extensions often requires expensive hardware modules. In some cases you need an entirely new phone system. Not so with an IP PBX: a standard computer can easily handle a large number of phone lines and extensions – just add more phones to your network to expand!
Benefit #7: Better customer service & productivity:
With an IP PBX you can deliver better customer service and better productivity: Since the IP telephone system is now computer-based you can integrate phone functions with business applications. For example: Bring up the customer record of the caller automatically when you receive his/her call, dramatically improving customer service and cutting cost by reducing time spent on each caller. Outbound calls can be placed directly from Outlook, removing the need for the user to type in the phone number.
Benefit #8: Twice the phone system features for half the price!
Since an IP PABX is software-based, it is easier for developers to add and improve feature sets. Most VOIP phone systems come with a rich feature set, including auto attendant, voice mail, ring groups, advanced reporting and more. These options are often very expensive in proprietary systems.
Benefit #9 Allow hot desking & roaming
Hot desking – the process of being able to easily move offices/desks based on the task at hand, has become very popular. Unfortunately traditional PBXs require extensions to be re-patched to the new location. With an IP PBX the user simply takes his phone to his new desk – No patching required!
Users can roam too – if an employee has to work from home, he/she can simply fire up their SIP software phone and are able to answer calls to their extension, just as they would in the office. Calls can be diverted anywhere in the world because of the SIP protocol characteristics!
Benefit #10 Better phone usability: SIP phones are easier to use
Employees often struggle using advanced phone features: Setting up a conference, transferring a call – On an old PBX it all requires instruction.
Not so with an IP PBX – all features are easily performed from a user friendly Windows GUI. In addition, users get a better overview of the status of other extensions and of inbound lines and call queues via the IP PBX Windows client. Proprietary systems often require expensive ‘system’ phones to get an idea what is going on on your phone system. Even then, status information is cryptic at best.
Conclusion:
Investing in a software-based IP PBX makes a lot of sense, not only for new companies buying a phone system, but also for companies who already have a PBX. An IP PBX delivers such significant savings in management, maintenance, and ongoing call costs, that upgrading to an EX4U hosted IP PBX Centrex, should be the obvious choice for any company.
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The benefits of replacing your old PBX with an IP PBX.
What is an IP PBX?
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An IP PBX is a complete telephony system that provides telephone calls over IP data networks. All conversations are sent as data packets over the network.
The technology includes advanced communication features but also provides a significant dose of worry-free scalability and robustness that all enterprises seek. The IP PBX is also able to connect to traditional PSTN lines via an optional gateway - so upgrading day-to-day business communication to this most advanced voice and data network is a breeze!
Enterprises don’t need to disrupt their current external communication infrastructure and operations. With IP PBX deployed, an enterprise can even keep its regular telephone numbers. This way, the IP PBX switches local calls over the data network inside the enterprise and allows all users to share the same external phone lines.
How it works:
An IP PBX or IP Telephone System consists of one or more SIP phones, an IP PBX server and optionally a VOIP Gateway to connect to existing PSTN lines. The IP PBX server functions in a similar manner to a proxy server: SIP clients, being either soft phones or hardware-based phones, register with the IP PBX server, and when they wish to make a call they ask the IP PBX to establish the connection. The IP PBX has a directory of all phones/users and their corresponding SIP address and thus is able to connect an internal call or route an external call via either the EX4U VOIP gateway or on the EX4U international VOIP platforms with the option of an hosted IP PBX, more details on the hosted IP PBX Centrex can be found in our VoIP section.
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What is SIP - Session Initiation Protocol?
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SIP, short for Session Initiation Protocol is an IP telephony signaling protocol used to establish, modify and terminate VOIP telephone calls. SIP was developed by the IETF and published as RFC 3261
SIP describes the communication needed to establish a phone call. The details are then further described in the SDP protocol.
SIP has taken the VOIP world by storm. The protocol resembles the HTTP protocol, is text based, and very open and flexible. It has therefore largely replaced the H323 standard.
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What is SDP - Session Description Protocol?
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SDP, short for Session Description Protocol, is a format for describing streaming media initialization parameters. It has been published by the IETF as RFC 4566. Streaming media is content that is viewed or heard while it is being delivered.
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What is H323?
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H323 is a set of standards from the ITU-T, which defines a set of protocols to provide audio and visual communication over a computer network.
H323 is a relatively old protocol and is currently being superseded by SIP – Session Initiation Protocol. One of the advantages of SIP is that its much less complex and resembles the HTTP / SMTP protocols.
Therefore most VOIP equipment available today follows the SIP standard. Older VOIP equipment though would follow H 323.
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What is ECHO cancellation?
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Echo cancellation is the process of removing echo from a voice communication in order to improve the voice call quality. Echo cancellation is often needed because speech compression techniques and packet processing delays generate echo. There are 2 types of echo: acoustic echo and hybrid echo.
Echo cancellation not only improves quality but it also reduces bandwidth consumption because of its silence suppression technique.
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What is RTP - Real Time Transport Protocol?
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RTP - short for Real Time Transport Protocol defines a standard packet format for delivering audio and video over the internet. It is defined in RFC 1889. It was developed by the Audio Video Transport Working group and was first published in 1996.
RTP and RTCP are closely linked – RTP delivers the actual data and RTCP is used for feedback on quality of service.
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What is RTCP - Real Time Transport Protocol?
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RTCP stands for Real Time Transport Protocol and is defined in RFC 3550. RTCP works hand in hand with RTP. RTP does the delivery of the actual data, where as RTCP is used to send control packets to participants in a call. The primary function is to provide feedback on the quality of service being provided by RTP.
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What is a SIP-URI?
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A SIP URI is the SIP addressing schema to call another person via SIP. In other words, a SIP URI is a user’s SIP phone number. The SIP URI resembles an e-mail address and is written in the following format:
SIP URI = sip:x@y:Port
Where x=Username and y=host (domain or IP)
Examples:
sip:john@sip.ex4u.org
sip:support@sip.ex4u.org
sip:123456789@sip.ex4u.org
The SIP URI standard has been defined in the RFC 3261 standard.
Links:
RFC 2396 - Uniform Resource Identifiers (URI): Generic Syntax.
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What are SIP Methods / Requests and Responses?
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SIP uses Methods / Requests and corresponding Responses to establish a call session.
SIP Requests:
There are six basic request / method types:
INVITE = Establishes a session
ACK = Confirms an INVITE request
BYE = Ends a session
CANCEL = Cancels establishing of a session
REGISTER = Communicates user location (host name, IP)
OPTIONS = Communicates information about the capabilities of the calling and receiving SIP phones
SIP responses:
SIP Requests are answered with SIP responses, of which there are 6 classes:
1xx = informational responses, such as 180, which means ringing
2xx = success responses
3xx = redirection responses
4xx = request failures
5xx = server errors
6xx = global failures
Note the similarity with HTTP – the beauty of SIP is in its clarity and simplicity
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Can you list all known SIP responses?
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1xx = informational responses
100 Trying
180 Ringing
181 Call Is Being Forwarded
182 Queued
183 Session Progress
2xx = success responses
200 OK
202 accepted: Used for referrals
3xx = redirection responses
300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
305 Use Proxy
380 Alternative Service
4xx = request failures
400 Bad Request
401 Unauthorized: Used only by registrars. Proxys should use proxy authorization 407
402 Payment Required (Reserved for future use)
403 Forbidden
404 Not Found: User not found
405 Method Not Allowed
406 Not Acceptable
407 Proxy Authentication Required
408 Request Timeout: Couldn't find the user in time
410 Gone: The user existed once, but is not available here any more.
413 Request Entity Too Large
414 Request-URI Too Long
415 Unsupported Media Type
416 Unsupported URI Scheme
420 Bad Extension: Bad SIP Protocol Extension used, not understood by the server
421 Extension Required
423 Interval Too Brief
480 Temporarily Unavailable
481 Call/Transaction Does Not Exist
482 Loop Detected
483 Too Many Hops
484 Address Incomplete
485 Ambiguous
486 Busy Here
487 Request Terminated
488 Not Acceptable Here
491 Request Pending
493 Undecipherable: Could not decrypt S/MIME body part
5xx = server errors
500 Server Internal Error
501 Not Implemented: The SIP request method is not implemented here
502 Bad Gateway
503 Service Unavailable
504 Server Time-out
505 Version Not Supported: The server does not support this version of the SIP protocol
513 Message Too Large
6xx = global failures
600 Busy Everywhere
603 Decline
604 Does Not Exist Anywhere
606 Not Acceptable
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Example of SIP Call session between 2 phones
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A sip call session between 2 phones is established as follows:
The calling phone sends out an invite
The called phone sends an information response 100 – Trying – back.
When the called phone starts ringing a response 180 – Ringing – is sent back
When the caller picks up the phone, the called phone sends a response 200 – OK
The calling phone responds with ACK – acknowledgement
Now the actual conversation is transmitted as data via RTP
When the person calling hangs up, a BYE request is sent to the calling phone
The calling phone responds with a 200 – OK.
It’s as simple as that! The SIP protocol is easy to understand and logical.
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How does FAX work in VOIP environments?
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FAX was designed for analog networks, and does not interoperate well at all with VOIP networks. The reason for this is that FAX communication uses the signal in a different way to regular voice communication.
When VOIP technologies digitize and compress analog voice communication it is optimized for VOICE and not for FAX. Subsequently, if you connect a Fax machine via an ATA adapter to the VOIP network it will work, but you are likely to encounter problems during fax transmissions. If you must do it this way, you should ensure that you are using the G 711 codec, which has a minimum of compression.
To deal with fax, you have the following options:
The easiest way to deal with this is to connect the fax machine directly to the existing analog phone line and bypass your VOIP environment altogether.
Replace the fax machine with a fax service provider. There are many available at a very low cost per month (cheaper then the phone line subscription)
Implement T38, which requires a T38 compatible gateway and a T38 compatible fax machine, fax card or fax software.
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What different types of CODECS are there?
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A Codec converts an analog signal to a digital one for transmission over a data network. The following Codecs are in use today
GSM - 13 Kbps (full rate), 20ms frame size
iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
ITU G.711 - 64 Kbps, sample-based. Also known as alaw/ulaw
ITU G.722 - 48/56/64 Kbps
ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
ITU G.726 - 16/24/32/40 Kbps
ITU G.728 - 16 Kbps
ITU G.729 - 8 Kbps, 10ms frame size
Speex - 2.15 to 44.2 Kbps
LPC10 - 2.5 Kbps
DoD CELP - 4.8 Kbps
Links:
Links to codecs and source code
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What is T38?
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T38 is a protocol that describes how to send a fax over a computer data network. T38 is needed because fax data can not be sent over a computer data network in the same way as voice communication. See How does FAX work in VOIP environments? for more information about this.
T 38 is described in RFC 3362, and defines how a device should communicate the fax data. In the picture above both the gateway and the fax machine behind the gateway would have to be capable of T38. For the G3 fax machine on an analog line, this process will be transparent. The analog fax machine does not need to know T38.
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What is FOIP - Fax over IP?
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FOIP stands for Fax over IP and refers to the process of sending and receiving faxes via a VOIP network. Fax over IP works via T38 and requires a T38 capable VOIP gateway as well as a T38 capable fax machine, fax card or fax software.
Modern multifunction fax machines support T38.
Fax server software that can talk ‘T38’ can send and receive faxes directly via the VOIP gateway and thus does not need any additional fax hardware. Currently, most fax servers require the use of separately licensed SoftIP or FOIP drivers to send and receive faxes without fax hardware.
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What is DID - Direct Inward Dialing?
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DID - Direct Inward Dialing (also called DDI in Europe) is a feature offered by telephone companies for use with their customers' PABX system, whereby the telephone company (telco) allocates a range of numbers associated with one or more phone lines.
Its purpose is to allow a company to assign a personal number to each employee, without requiring a separate phone line for each. That way, telephony traffic can be split up and managed more easily.
DID requires that you purchase an ISDN or Digital line and ask the telephone company to assign a range of numbers. You then need DID capable equipment at your premises which consists of BRI, E1 or T1 cards or gateways.
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What are the benefits of an IP PBX?
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Much easier to install & configure than a proprietary phone system
Easier to manage because of web based configuration interface
No need for separate phone wiring
Allows users to hot plug their phone anywhere in the office - users simply take their phone, plug it into the nearest Ethernet port and keep their existing number!
Allows easy roaming - calls can be diverted anywhere in the world because of the SIP protocol characteristics
Significant cost reduction by leveraging Internet
SIP standard eliminates proprietary, expensive phones
Scalable
Better reporting
Better overview of system status and calls
Much easier to install & configure than a proprietary phone system:
A software program running on a computer can take advantage of the advanced processing power of the computer and user interface & features of Windows. Anyone with an understanding of computers and Windows can install and configure the PBX. A proprietary phone system often requires an installer trained on that particular proprietary system!
Easier to manage because of web based configuration interface:
A VOIP phone system has a web based configuration interface, allowing you to easily maintain and fine tune your phone system. Proprietary phone systems often have difficult to use interfaces which are often designed so that only the phone system installers can effectively use it.
Call cost reduction:
You can save substantially by using a VOIP service provider for long distance or international calls. Easily connect phone systems between offices/branches and make free phone calls.
No need for separate phone wiring – use computer network:
A VOIP phone system allows you to connect hardware phones directly to a standard computer network port (which it can share with the adjacent computer). Software phones can be installed directly onto the PC. This means that you do not need to install & maintain a separate wiring network for the phone system, giving you much greater flexibility to add users/extensions. If you are moving into an office and have not yet installed phone wiring, you can save significantly by just installing a computer network.
No vendor lock in:
Use standard phones: VOIP phone systems are open standard – all modern IP phone systems use SIP as a protocol. This means that you can use almost any SIP VOIP phone or VOIP gateway hardware. In contrast, a proprietary phone system often requires proprietary phones to use advanced features, and proprietary extension modules to add features.
Scalable:
Proprietary systems are easy to outgrow: Adding more phone lines or extensions often requires expensive hardware upgrades. In some cases you need an entirely new phone system. Not so with a VOIP phone system: a standard computer can easily handle a large number of phone lines and extensions – just add more phones to your network to expand!
Better customer service & productivity:
Since calls are computer based, it's much easier for developers to integrate with business applications. For example: an incoming call can automatically bring up the customer record of the caller, dramatically improving customer service and cutting cost by reducing time spent on each caller. Outbound calls can be placed directly from Outlook, removing the need for the user to type in the phone number.
Software based Phones are easier to use:
It's often difficult to use advanced phone system features such as conferencing on proprietary phones. Not so with software based SIP phones – all features are easily performed from a user friendly windows GUI.
More features included as standard:
Since a VOIP phone system is software based, it's more easy for the developers to develop, add and improve feature sets. Therefore most VOIP phone systems come with a rich feature set, including auto attendant, voice mail, call queueing and more. These options are often very expensive in proprietary systems.
Better control via better reporting:
VOIP settings store inbound and outbound call information in a database on your server, allowing for much more powerful reporting of call costs and call traffic.
Better overview of current system status and calls:
Proprietary systems often require expensive ‘system’ phones to get an idea what is going on on your phone system. Even then, status information is cryptic at best. With VOIP systems you can define which users can see phone system status graphically via a web browser.
Allow users to hot plug their phone anywhere in the office:
Users simply take their phone, plug it into the nearest ethernet port and they keep their existing number!
Allows easy roaming of users:
Calls can be diverted anywhere in the world because of the SIP protocol characteristics
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IP PBX: How an IP PBX / VOIP phone system works
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A VOIP Phone System / IP PBX system consists of one or more SIP phones / VOIP phones, an IP PBX server and optionally includes a VOIP Gateway. The IP PBX server is similar to a proxy server: SIP clients, being either soft phones or hardware based phones, register with the IP PBX server, and when they wish to make a call they ask the IP PBX to establish the connection. The IP PBX has a directory of all phones/users and their corresponding SIP address and thus is able to connect an internal call or route an external call via either the EX4U local VOIP gateway or the EX4U international VOIP switches. Optional with the hosted IP PBX.
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What SIP-based IP PBX is available?
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EX4U Telecom advanced hosted IP PBX Centrex with unlimited DIDs, extensions, voicemail unified messaging, unlimited callt to USA & Canada, and more in our VoIP section.
IP PBX answers, screens and routes calls, provides unified virtual extension numbers, processes large call volumes simultaneously.
Easy manageability
EX4U TELECOM IP PBX / Centrex is easy to set up via an intuitive user-friendly web interface, which enables controlling basic settings of the system, configuring call forward and routing rules, setting an e-mail address for voice mail and adding new users to the system
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Sip phones / VOIP Phones types
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A VOIP phone system requires the use of SIP phones / VOIP phones.
SIP phones come in several versions/types:
SIP / VOIP Soft phones - software based SIP phone
A software based SIP phone is a program which makes use of your computer's microphone and speakers, or an attached headset to allow you to make or receive calls.
USB VOIP phones
A USB phone plugs into the USB port of a computer and with the use of a SIP/ VOIP soft phone software behaves just like a phone. Essentially its not more then a microphone with a speaker, however because they appear like a normal phone they are more intuitive to use for a user.
Hardware SIP Phone
A hardware based SIP phone looks like and behaves just like a normal ‘phone’. However it is connected directly to the data network. These phones have an integrated mini hub, so that they can share the network connection with the computer. That way you do not need an additional network point for the phone. Examples of hardware SIP phones are Linksys, Cisco, Zyxel, Grandstream, Nokia, Palm.
Use analog phone via an ATA adapter
If you want to use your current phone with the VOIP phone system, you can use an ATA adapter. An ATA adapter (Examples: Linksys,Zyxel, Zoom) allows you to plug in the Ethernet network jack into the adapter and then plug the phone into the adapter. That way your old phone will appear to the VOIP phone system software as a regular SIP phone.
Please contact us for your specific VoIP hardphone or our free softphones.
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VOIP gateway information - Learn about VOIP gateways
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A VOIP gateway is a device which converts telephony traffic into IP for transmission over a data network. They are used in 2 ways:
1. To convert incoming PSTN/telephone lines to VOIP/SIP:
In this manner the VOIP gateway allows calls to be received & placed on the regular telephony network. In many business cases, it is preferable to continue to use traditional phone lines because one can guarantee a higher call quality and availability.
2. To connect a traditional PBX/Phone system to the IP network:
In this manner the VOIP gateway allows calls to be made via VOIP. Calls can then be placed via a VOIP service provider, or in the case of a company with multiple offices, inter office calls costs can be reduced by routing the calls via the Internet. VOIP gateways are available as external units or as PCI cards. The vast majority of devices are external units. A VOIP gateway will have a connector for the IP network and one or more ports to connect the phone lines to it.
Types of VOIP gateways:
1. Analog units: Analog units are used to connect regular analog phone lines to it. Analog units are available for between 2-24 lines.
2. Digital units: Digital units allow you to connect digital lines either one or more BRI ISDN lines (Europe), one or more PRI/E1 lines (europe) or one or more T1 lines (USA).
VOIP gateway manufacturers:
There are lots of VOIP gateways available today, and as demand has increased drastically, prices have decreased considerably.
Please contact us for detailed information.
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What do the terms FXS and FXO mean?
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FXS and FXO are the name of ports used by Analog phone lines (also known as POTS - Plain Old Telephone Service).
FXS - Foreign eXchange Subscriber interface is the port that actually delivers the analog line to the subscriber. In other words it is the ‘plug on the wall’ that delivers a dialtone, battery current and ring voltage.
FXO - Foreign eXchange Office interface is the port that receives the analog line. It is the plug on the phone or fax machine, or the plug(s) on your analog phone system. It delivers an on-hook/off-hook indication (loop closure). Since the FXO port is attached to a device, such as a fax or phone, the device is often called the ‘FXO device’.
FXO and FXS are always paired, i.e similar to a male / female plug.
Without a PBX, a phone is connected directly to the FXS port provided by a telephone company.
If you have a PBX, then you connect the lines provided by the telephone company to the PBX and then the phones to the PBX. Therefore, the PBX must have both FXO ports (to connect to the FXS ports provided by the telephone company) and FXS ports (to connect the phone or fax devices to).
FXS & FXO & VOIP
You will come across the terms FXS and FXO when deciding to buy equipment that allows you to connect analog phones to a VOIP Phone System or traditional PBXs to a VOIP service provider or to each other via the Internet.
An FXO gateway
To connect analog phone lines to an IP phone system you need an FXO gateway. This allows you to connect the FXS port to the FXO port of the gateway, which then translates the analog phone line to a VOIP call.
An FXS gateway
An FXS gateway is used to connect one or more lines of a traditional PBX to a VOIP phone system or provider. You need an FXS gateway because you want to connect the FXO ports (which normally are connected to the telephone company) to the Internet or a VOIP system.
An FXS adapter a.k.a. ATA adapter
An FXS adapter is used to connect an analog phone or fax machine to a VOIP phone system or to a VOIP provider. You need this because you need to connect the FXO port of the phone/fax machine to the adapter.
Connecting - FXS/ FXO procedures – how it technically works
If you are interested to know in more technical detail how an FXS/ FXO port interoperate, here is the exact sequence:
When you wish to place a call:
You pick up the phone (the FXO device). The FXS port detects that you have gone off hook.
You dial the phone number, which is passed as Dual-Tone Multi-Frequency (DTMF) digits to the FXS port.
Inbound call:
The FXS port receives a call, and then sends a ring voltage to the attached FXO device.
The phone rings
As soon as you pick up the phone you can answer the call.
Ending the call – normally the FXS port relies on either of the connected FXO devices to end the call.
Note: The analog phone line passes approximately 50 volts DC power to the FXS port. That’s why you get a faint ‘shock’ when you touch a connected phone line. This allows a call to be made in the event of a power cut.
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What is a STUN Server?
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A STUN (Simple Traversal of User Datagram Protocol [UDP] Through Network Address Translators [NATs]) server allows NAT clients (i.e. computers behind a firewall) to setup phone calls to a VOIP provider hosted outside of the local network.
The STUN server allows clients to find out their public address, the type of NAT they are behind and the internet side port associated by the NAT with a particular local port. This information is used to set up UDP communication between the client and the VOIP provider and so establish a call. The STUN protocol is defined in RFC 3489.
The STUN server is contacted on UDP port 3478, however the server will hint clients to perform tests on alternate IP and port number too (STUN servers have two IP addresses). The RFC states that this port and IP are arbitrary.
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What is a SIP server?
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A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy or a Registrar.
An example of a SIP server is 3CX Phone System - a Windows-based SIP server
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VOIP defined
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Voice over IP (also called VoIP, IP Telephony, and Internet telephony) refers to technology that enables routing of voice conversations over the Internet or a computer network. To place calls via VOIP, a user will need a software based sip phone program OR a hardware based VOIP phone. Phone calls can be made to anywhere / anyone: Both to VOIP numbers as well as people with normal phone numbers.
Read about what types of VOIP phones / SIP phones exist and what VOIP gateways are and how they can be used.
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VOIP Definitions
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VoIP – Voice over Internet Protocol (also called IP Telephony, Internet telephony, and Digital Phone) – is the routing of voice conversations over the Internet or any other IP-based network.
SIP – Session Initiation Protocol – is a protocol developed by the IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality.
PSTN – the public switched telephone network – is the concentration of the world's public circuit-switched telephone networks, in much the same way that the Internet is the concentration of the world's public IP-based packet-switched networks.
ISDN – Integrated Services Digital Network – is a type of circuit switched telephone network system, designed to allow digital (as opposed to analog) transmission of voice and data over ordinary telephone copper wires, resulting in better quality and higher speeds, than available with analog systems.
PBX – Private Branch eXchange (also called Private Business eXchange) – is a telephone exchange that is owned by a private business, as opposed to one owned by a common carrier or by a telephone company.
IVR – In telephony, Interactive Voice Response – is a computerised system that allows a person, typically a telephone caller, to select an option from a voice menu and otherwise interface with a computer system.
DID – Direct Inward Dialing (also called DDI in Europe) is a feature offered by telephone companies for use with their customers' PBX system, whereby the telephone company (telco) allocates a range of numbers all connected to their customer's PBX.
RFC – Request for Comments (plurals Requests for Comments but RFCs) is one of a series of numbered Internet informational documents and standards very widely followed by both commercial software and freeware in the Internet and Unix communities.
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What does ENUM mean?
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ENUM stands for Telephone Number Mapping.
Behind this ‘abbreviation’ hides a great idea: To be reachable anywhere in the world with the same number – and via the best and cheapest route. ENUM takes a phone number and links it to an internet address which is published in the DNS system. The owner of an ENUM number can thus publish where a call should be routed to via a DNS entry. Whats more, different routes can be defined for different types of calls - for example you can define a different route if the caller is a fax machine. ENUM does require the phone of the caller to support it.
You register an ENUM number rather like you register a domain.
ENUM is a new standard, and is not that widespread yet. Though it looks set to become another revolution in communications and personal mobility.
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