What is VoIP?
Last 50 years people have been using conventional PBX systems which require separate networks for voice and data communications. But with the latest VOIP telephony revolution, businesses are quickly migrating to VOIP PBX systems. VOIP, which stands for Voice over Internet Protocol, is basically the transmission of voice traffic over IP-based networks. Initially designed for data networking, the Internet Protocol (IP) was adapted to voice networking following its successful positioning as the global standard for data networking.In Voice over Internet Protocol, a conversation is converted to packets of data that flit all over the Internet or private networks, just like e-mails or Web pages, though voice packets get priority status. The packets get reassembled and converted to sound on the other end of the call. With VOIP phone systems users are not limited to making and receiving calls through the IP network, traditional phone lines can be used to guarantee a higher call quality and availability. With the use of a VOIP gateway incoming PSTN/telephone lines can be converted to VOIP/SIP. This way the VOIP gateway allows the user to receive and place calls on the regular telephony network.
VOIP PBX systems provide mobility to employees, flexibility when a business expands as they are much easier to manage than the traditional PBX, and can also considerably reduce telephony administration costs.
What is a PBX Phone System?
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:
PBX
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 PBX.
What is a VoIP Telephone?
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.
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 dont 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.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.
Whar are the advantages of IP PBX?
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.
What is Unified Communications?
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.
What is SIP - Session Initiation Protocol?
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.
What is SDP - Session Description Protocol?
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.
What is SIP - Session Initiation Protocol?
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.
What is H323?
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.
What is ECHO cancellation?
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.
What is RTP - Real Time Transport Protocol?
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.
What is RTCP - Real Time Transport Protocol?
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.
What is a SIP-URI?
A SIP URI is the SIP addressing schema to call another person via SIP. In other words, a SIP URI is a users 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:
This e-mail address is being protected from spambots. You need JavaScript enabled to view it
sip:
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sip:
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The SIP URI standard has been defined in the RFC 3261 standard.
What are SIP Methods / Requests and Responses?
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.
Can you list all known SIP responses?
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
Example of SIP Call session between 2 phones
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.
How does FAX work in VOIP environments?
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.
What different types of CODECS are there?
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
What is T38?
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.
What is FOIP - Fax over IP?
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 multi function 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 EICON SoftIP or Cantata FOIP drivers to send and receive faxes without fax hardware.
What is DID - Direct Inward Dialing?
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.
What are the benefits of an IP PBX?
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
How an IP PBX / VOIP phone system works?
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.
What does ENUM mean?
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.
