SIP is the internet standard for real time voice and video communication. SIP (Session Initiation Protocol) was developed by the IETF and published as RFC 3261.
SIP is an internet protocol for live communications used in setting up and tearing down voice or video calls. It is a signaling protocol used to create, modify, and terminate sessions with one or more participants in an IP network. A session can be a straightforward two-way phone call or it can be a multi-media conference session with many persons participating. SIP has made possible an array of services that seemed unthinkable just a few years ago: internet conferencing, IP telephony, instant messaging, presence, voice and video communication, data collaboration, online gaming, application sharing, and much more.
SIP is doing for real-time communications what HTTP did for the web and SMTP for email. It is the main driver in the acceleration of the IP Telephony revolution. With SIP Telephony, a viable alternative to traditional PBX has emerged. SIP telephone systems deliver features that enhance users’ mobility and productivity, while securing substantial cost-saving advantages. This is making proprietary hardware based PBXs obsolete.
Some of the most common questions about SIP are answered in this SIP FAQ.
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. In essence, with T38 a fax is converted to an image, sent to the other T38 fax device and then converted back to an analog fax signal. Most VoIP Gateways and ATA’s now support T38 reliably. An example fax setup using a Linksys 3102 can be found here
T38 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.
3CX Phone System for Windows includes a full featured T38 fax server that allows faxes to be sent and received from anywhere in the network. Faxes can be received as PDF and forwarded via email. Faxes can be sent from anywhere in the network via Microsoft Fax Services (which is included with Windows 2003 and 2008)
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.
More information about STUN and VoIP in general can be found in our SIP / VoIP Video tutorials, ’Voip Nuggets’. VoIP Nuggets are short youtube technical training tutorials about VoIP. Click here for the latest list of VoIP Nuggets
Stun functionality is seamlessly handled by 3CX Phone System for Windows – an easy to install windows based PBX.
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.
- IPtelephony.org - VOIP Telephony and news resources
- VOIP resources - voiceoverinternet.com
- FrameIP - French website: VOIP Telephony and news resources
- Pernau.at - SIP software listing
- Monitor your IP PBX server with PA Server Monitor
- Smith on VOIP - up to date information and news on the latest VoIP products and services
- Nick on IT - insights on the latest developments in IP Telephony
- Asterisk on Windows - 3CX Phone System for Windows
- 3CX VOIP FAQ - VOIP FAQ & Info
PBX stands for Private Branch Exchange, which is a private telephone network used within a company. The users of the PBX phone system share a number of outside lines for making external phone calls.
A PBX connects the internal telephones within a business and also connects them to the public switched telephone network (PSTN).
One of the latest tendencies in PBX phone system development is the VoIP PBX, also known as IP PBX, which uses the Internet Protocol to transmit calls.
Nowadays, there are four different PBX phone system options:
- Hosted/Virtual PBX
- IP PBX
- Hosted/Virtual IP PBX
IP PBX is a software-based PBX phone system solution which helps accomplish certain tasks and delivers services that can be difficult and costly to implement when using a traditional proprietary PABX.
3CX Phone System for Windows is a good example of an IP PBX phone system.
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
There are two types of SIP Phones. The first type is the hardware SIP phone, which resembles the common telephone but can receive and make calls using the internet instead of the traditional PSTN system.
SIP Phones can also be software-based. These allow any computer to be used as a telephone by means of a headset with a microphone and/or a sound card. A broadband connection and connection to a VOIP provider or a SIP server are also required.
Read more about 3CX SIP / VoIP phone.
Read more about the different types of VOIP / SIP phones.
Voice over IP is the same as Voice over Internet Protocol, and is better known as VoIP.
Voice over IP refers to the diffusion of voice traffic over internet-based networks. Internet Protocol (IP) was originally designed for data networking and following its success, the protocol has been adapted to voice networking.
Voice over IP (VoIP) can facilitate tasks and deliver services that might be cumbersome or costly to implement when using traditional PSTN:
- More than one phone call can be transmitted on the same broadband phone line. This way, voice over IP can facilitate the addition of telephone lines to businesses.
- Features that are usually charged extra by telecommunication companies, such as call forwarding, caller ID or automatic redialing, are simple with voice over IP technology.
- Unified communications are secured with voice over IP technology, as it allows integration with other services available on the internet such as video conversation, messaging, etc.
These, and many other advantages of voice over IP, are making businesses adopt VoIP Phone Systems at a staggering pace.
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.
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. The IP PBX FAQ helps answer common questions about VOIP, SIP, IP PBX / VOIP Phone System hardware & Software, implementation and more.
Auto-attendant (or automated attendant) is a term commonly used in telephony to describe a voice menu system that allows callers to be transferred to an extension without going through a telephone operator or receptionist. The auto-attendant is also known as a digital receptionist.
For a caller to find a user on a phone system, a dial-by-name directory is usually available. This feature lists users by name, allowing the caller to press a key to automatically ring the extension of a user once his/her extension is announced by the auto attendant.
If a user is not available, the auto-attendant directs callers to the appropriate voice mailbox of the user to leave a voicemail message.
Having an auto-attendant in a phone system is a very useful and cost-effective feature for a business, as it replaces/helps the human operator by automating and simplifying the incoming phone calls procedure.
3CX includes a FREE auto-attendant feature in 3CX IP PBX for Windows. Read more about 3CX Phone System for Windows auto-attendant feature.
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 PABX. 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.
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 withsoftware 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
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. Fax server software that can talk ‘T38′ allows sending and receiving faxes directly via a VOIP gateway and, consequently, does not need any additional fax hardware.
3CX includes a T38 compatible network fax server in its 3CX Phone System for Windows. Faxes are converted to PDF files and forwarded via email. Outbound faxes are sent via Microsoft Fax from anywhere in the network. Other fax servers currently in the market require the use of separately licensed and expensive Dialogic SoftIP drivers.
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.
3CX has a lot of useful information about VoIP on its website. Check out these links:
- VoIP Nuggets: Youtube Video tutorials about VoIP & SIP
- VoIP FAQ: Frequently asked questions about VoIP
- VoIP Articles: Articles about VoIP, SIP and PBX
- 3CX Blog: Check out 3CX’ VoIP & SIP blog
- Forums: Ask questions about VoIP & SIP
- Free VoIP PBX: 3CX Phone System for Windows
Another good starting point is at Wikipedia: http://en.wikipedia.org/wiki/Voice_over_IP
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. At present many registrars and VOIP providers are providing this as a free service.
ENUM is a new standard, and is not that widespread yet. Though it looks set to become another revolution in communications and personal mobility.
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 a VOIP gateway or a VOIP service provider.
How an IP PBX integrates on the network and how it uses the PSTN or Internet to connect calls
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.
A VOIP gateway (or PSTN 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.
An analog VOIP gateway
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. Analog VOIP gateways start at as little as $200. 3CX Phone System supports numerous VoIP gateways. A list of supported VoIP Gateways with configuration guides can be found here.
They can all be bought online via one of the many VOIP product online shops. More information about VoIP Gateways and VoIP in general can be found in our SIP / VoIP Video tutorials, ’Voip Nuggets’. VoIP Nuggets are short youtube technical training tutorials about VoIP. 3CX Phone System for Windows automatically configures VoIP Gateways to allow you to easily continue using your existing PSTN lines. You can download the Free edition here.
- 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.
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
FXS and FXO are the name of ports used by Analog phone lines (also known as POTS – Plain Old Telephone Service) or phones.
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.
FXS / FXO without a PBX
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 with a PBX
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. There are a number of different FXO gateways available. You can view different types that 3CX Phone System supports here.
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. Alternatively, you can use it to connect analog phones to it and re-use your analog phones with a VoIP phone system. 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.
FXS/ FXO gateways are widely available. 3CX Phone System for Windows automatically configures FXS/FXO Gateways to allow you to easily continue using your existing PSTN lines and/or analog phones. You can download the Free edition here
More information about FXS / FXO and VoIP in general can be found in our SIP / VoIP Video tutorials, ‘Voip Nuggets’. VoIP Nuggets are short youtube technical training tutorials about VoIP & SIP. Click here for the latest list of VoIP Nuggets.
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.
- 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.
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)
The SIP URI standard has been defined in the RFC 3261 standard. 3CX Phone System for Windows uses SIP URIs
Interactive Voice Response or IVR is a telephone technology that communicates with a caller through configurable voice menus and data in real time. In an IVR system, callers are given the choice to select options by pressing digits.
IVR systems can normally handle and service high volumes of phone calls. With an Interactive Voice Response system, businesses can reduce costs and improve customers’ experience as Interactive Voice Response systems allow callers to get information they need 24 hours a day without the need of costly human agents.
Some IVR applications include telephone banking, flight-scheduling information and tele-voting.
3CX Phone System for Windows has a built-in IVR that is designed to boost the competence of any business by increasing flexibility, simplifying processes and reducing costs, at the same time as improving customer satisfaction.
Learn more about 3CX IP PBX’s Interactive Voice Response system.
Read more about 3CX Phone System for Windows.
Although the SIP server is the most important part of the SIP based phone system, it only handles call setup and call tear down. It does not actually transmit or receive any audio. This is done by the media server in RTP.
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. Examples of SIP phones are SJPhone from SJlabs (http://www.sjlabs.com), Xten (http://www.xten.net) or 3CX VOIP Phone for Windows.
3CX VoIP phone
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.
A USB phone
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 Cisco, Linksys, Aastra, Snom and Grandstream.
A hardware SIP phone
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 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.
An ATA adapter which allows an analog phone to connected to a VOIP system
VoIP phones are very inexpensive to buy and can bought online via one of the many VOIP product online shops. 3CX VoIP Phone System for Windows supports all popular VoIP phones and most models can be automatically provisioned too. You can download the Free edition here.
More information about VoIP phones and VoIP in general can be found in our SIP / VoIP Video tutorials, ‘Voip Nuggets’. VoIP Nuggets are short youtube technical training tutorials about VoIP. Click here for the latest list of VoIP Nuggets.
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 theSDP 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.
The implementation of an IP telephone system in a business requires the use of a very specific type of phone: the IP Telephone.
IP Phones are sometimes called VoIP telephones, SIP phones or softphones. These are all the exact same thing and are based on the principle of transmission of voice over the internet, or what is better known as VoIP (or voice over internet protocol) technology.
IP Telephones come in several types. Learn more about the different kinds of IP phones.
Microsoft Response Point and 3CX Phone System for Windows are both IP PBXs that replace the traditional PBX and can improve business communications and help cut costs.
But the two phone systems differ in the delivery of some important advantages. The main differences between Microsoft Response Point and 3CX Phone System for Windows are:
- Microsoft Response Point ties your business to specific hardware. On the contrary, 3CX Phone System for Windows works with most popular VOIP gateways and SIP phones, allowing you to choose the hardware that best suits your business’ needs and budget.
- Microsoft Response Point is focused on a small number of core capabilities. In the case of 3CX Phone system for Windows businesses can enjoy a full set of enterprise level features at a very low cost.
- Microsoft Response Point is limited to 50 users. The FREE edition of 3CX Phone System for Windows and the three commercial editions can be used with an unlimited number of extensions.
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 superceded 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.
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.
A voice mail system is a centralized system used in businesses for sending, storing and retrieving audio messages, just like an answering machine would do at home.
Each extension in a phone system is normally linked to a voice mailbox, so when the number is called and the line is not answered or is busy, the caller listens to a message previously recorded by the user. This message can give instructions to the caller to leave a voice message or gives other available options, such as paging the user or being transferred to another extension or a receptionist.
A voicemail system in a business is essential to keep external and internal communications flowing seamlessly and efficiently.
3CX has integrated a free voice mail system in its IP PBX for Windows. 3CX Phone System for Windows delivers a complete voice mail solution that incorporates Unified Communications by allowing voice mail to be forwarded to the user’s email inbox.
Learn more about 3CX’s voicemail solution.
Read more about 3CX Phone System for Windows.
A VoIP telephone, also known as a SIP phone or a softphone, allows the user to make phone calls to any softphone, mobile or landline by using voice over IP (VoIP). This way the voice is carried through the internet instead of the traditional PSTN system.
A VoIP telephone can be a simple software-based softphone or a hardware device that looks a lot like a common telephone.
Some of the common features of a VoIP telephone are: caller ID, call park, call transfer and call hold.
3CX has developed a completely free VOIP telephone that can secure significant savings on telephone bills in a very simple way: all that the user requires is a broadband connection (DSL or cable), a connection to a VOIP provider or aSIP server, and a headset with a microphone and/or a sound card.
Read more about 3CX VoIP telephone.
Read more about the different types of VoIP telephones.
FAX was designed for analog networks, and does not travel well over a VOIP network. 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, there are a number of things you need to take note of when you move to a VoIP Phone System.
If you want to continue using your old fax machine, and you want to connect to your VoIP phone system, its best to use a VoIP Gateway and an ATA that supports T38. T38 is a protocol designed to allow fax to ‘travel’ over a VoIP network. An example configuration of such a setup can be found here.
It is also possible to convert to computer based fax and choose a VoIP phone system that supports fax. 3CX Phone System for Windows includes a full featured fax server that is able to receive faxes and forward them in PDF format to e-mail. Faxes can be sent from anywhere in the network using theMicrosoft Fax client & fax server (which comes free with Windows Server 2003 and 2008)
Another way to deal with fax when you switch to voip are to connect the fax machine directly to the existing analog phone line and bypass your VOIP system.
SIP forking refers to the process of “forking” a single SIP call to multiple SIP endpoints. This is a very powerful feature of SIP. A single call can ring many endpoints at the same time.
With SIP forking you can have your desk phone ring at the same time as your softphone or a SIP phone on your mobile. For example, you would use SIP forking to ring your deskphone and your Android SIP Phone at the same time, allowing you to take the call from either device easily. No forwarding rules would be necessary as both devices would ring. In the same manner SIP forking can be used in an office and allow the secretary to answer calls to the extension of his/her boss when he is away or unable to take the call.
3CX Phone System full supports SIP forking. You can download the free edition of 3CX IP PBX for Windows here.
SIP uses Methods / Requests and corresponding Responses to establish a call session.
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 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
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.
The benefits of replacing your old PBX with an IP PBX
What is an IP PBX?
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
Figure 1 – How an IP PBX integrates into the network
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 a VOIP gateway or a VOIP service provider. More information and commonly asked questioned about IP PBXs can be found on IP PBX, SIP & VOIP FAQ
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 adjacent 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.
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 IP PBX, should be the obvious choice for any company.
- 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
- Better reporting
- Better overview of system status and calls
- More IP PBX benefits
Unified Communications is defined as the process in which all means of communication, communication devices and media are integrated, allowing users to be in touch with anyone, wherever they are, and in real time.
The objective of Unified Communications is to optimize business procedures and boost human communications by simplifying processes.
Read more about Unified Communications.