Category Archives: Knowledge Base

The Importance of a Firewall

Firewalls perform three very important, crytical functions in relation to SIP.

We discussed what Network Address Translation (NAT) is, its purpose on the network and why this is a problem when adopting SIP in the enterprise. In this issue, we will look at how Ingate SIP-capable security products solve the issues with NAT and do so while maintaining security.

Firewalls perform three very important functions. These functions are critical for any SIP implementation – including SIP trunking — and may provide other benefits as well depending on the construction of the network and the service that the business wishes to employ. The following are the most important of these functions:

  1. The firewall resolves NAT traversal issues and enables the adoption of SIP and SIP trunking by securely permitting SIP signaling and related media to traverse the firewall. Without this function, most companies will have one-way audio only.
  2. The Ingate products also normalize the SIP Traffic so that the IP-PBX at the customer site, and the service provider’s network are fully compatible. While SIP is a standard, each implementation can be slightly different, and the service providers may each require a different level of authentication from the business. With thefirewall in place, these requirements can be met.
  3. Security becomes important with the introduction of SIP trunking and other SIP services because as a server, the IP-PBX is subject to the same types of threats as any other server on the enterprise network. So the properly architected network will include a firewall device to assure that only authorized users are allowed to send and/or receive phone calls using the service, and that no malicious attacks are launched which could render the IP-PBX inoperable.

What is NAT?

Understanding Network Address Translation (NAT) for use with SIP.

We often hear of problems with NAT Traversal and SIP. This week we provide a short synopsis of Network Address Translation and its purpose on the network and why this is a problem when bringing SIP into a network.

Since the addressing and routing of SIP is done at the application layer, the biggest problem the SIP protocol now has is the disconnect between the IPv4 addressing and routing at the application layer versus the IPv4 addressing and routing at the transport and network layers. Network Address Translation (NAT) occurs at the transport and network layers, and thus the challenge.

The purpose of a Network Address Translation (NAT) firewall for businesses is to provide the translation between a single public IP address on the WAN and multiple private IP addresses for all of the workstations, servers and other IP equipment within the LAN. The router running NAT should never advertise the LAN network addresses to the WAN network backbone. Only the networks with global addresses may be known outside the router. However, global information that NAT receives from the border router can be advertised in the LAN network the usual way. Typical or traditional firewalls apply NAT to the TCP/IP protocol at the transport and network layers.

NAT’s basic operation is as follows. The network addresses inside a private domain can be reused by any other private domain. For instance, a single Class A address could be used by many private domains. At each exit point between a private domain and the public WAN backbone, NAT is installed. If there is more than one exit point it is of great importance that each NAT has the same translation table.

In order for SIP to work effectively, the NAT issue must be resolved, and that is where the Session Border Element such as the firewalls are very important for enabling SIP services to an enterprise network.

SIP Routing Rules and Policies

Routing to accomodate SIP applications.

The Deep Packet Inspection capable firewalls offer the ability to apply Routing and Dial Plan rules to all incoming SIP traffic. As the Ingate product has the ability to look at Layer 2 through Layer 7 of the OSI model, Routing and Dial Plan rules can combine the use of several layers at once. Combining such things as the TCP/IP (Transport Layer) with the SIP protocol (Application Layer) ensures that only predefined SIP traffic is processed.

Best Routing Rules:

  1. Match From Header, where the router can match on the From Header SIP URI, (the person making the call). In addition the router can separate the Transport whether UDP, TCP or TLS, and further we can specify which IP address or range of IP addresses at the Network layer from which we can accept calls.
  2. Matching Request URI. The Request URI Header is a routable header of any SIP Request. The router can Match & Remove a Prefix, Match any specific Alpha/ Numeric characters or even range of characters. This also includes Domain matching.
  3. Forward To. The Forward To section defines where to ‘actually’ send the call – perhaps to a predefined account, with Registration and/or Header Replacement requirements/behavior; or to an IP address or Domain. It can also change the call request to a different Transport and port if required, and even dynamically assign the use of our B2BUA if needed.

The actual Dial Plan, then, combines these three attributes to provide the ultimate in flexibility and security in defining:

  • accepting where the call is coming from and
  • where the call is going. If the SIP traffic is not predefined it will be denied.

This also gives the ability to have multiple different IP-PBX vendors and multiple different ITSP accounts. N+1 ITSPs to N+1 IP-PBXs. There is no limit to the customization of call routing in some routers.

Best Policies:

Policies related to SIP have to do with allowing or disallowing SIP traffic based on SIP Methods, SIP Mime Content, SIP Domains and other higher-level rules. A SIP Method policy can be implemented to ensure incoming SIP packets are matched on the particular SIP Method and Traffic to specified domains. If required, Authentication can be applied for processing the packet. Further policies can be applied to filter MIME Content types, to ensure the type of SIP Traffic is allowed. Filtering based on specific Header information is also possible.

Other Routing Rules and Policies can also be applied to allow for SIP Domain forwarding, Static SIP URI forwarding, SIP Registrar Authentication, and more.

The Basics of Sip Trunking

A basic understanding of how Session Initiation Protocol (SIP) works.

SIP Trunking is a term applied to the services offered by LECs (Local Exchange Carriers), ILECs (Independent Local Exchange Carriers), CLECs (Competitive Local Exchange Carriers) and ITSPs (Internet Telephony Service Providers) to terminate Voice over IP (VoIP) calls to the Public Switched Telephone Network (PSTN).

SIP Trunking allows enterprises and small businesses to eliminate a PSTN gateway at their site and outsource that function to a carrier. It is typically a lower-cost alternative to Primary Rate Interfaces (PRIs) because SIP trunks can be purchased in single-trunk increments (as compared to 23 channel increments for a PRI).

Other ways in which SIP trunks decrease costs:

With SIP trunks, a single network can be maintained within the organization, rather than having both a voice and data network.
Internet bandwidth can be used more efficiently.
Moves, Adds and Changes can be completed without major wiring upgrades.

SIP Trunks are delivered in several ways:

Over the Public Internet – SIP Trunking Anywhere: Allows any enterprise, anywhere, to adopt SIP Trunking and assign some, possibly unused, bandwidth to voice at no extra charge for the connection, and providing the highest ROI.

Managed Services: Carriers supply a dedicated, fully managed connection from their Point of Presence to the enterprise site. This service offers quality of service guarantees, but is somewhat more expensive.

MPLS Delivery: The carrier, usually an LEC, ILEC or CLEC, will delivery a managed service using Multi-Protocol Label Switching to insure the highest voice quality and reliability.

The voice quality, even over an un-managed public Internet connection, is excellent. Typical savings over PRIs range from 40-60% with the payback period for the equipment required, which may include an upgrade to the IP-PBX and the installation of an Ingate SIParator or Firewall, has been shown to range from 4 to 12 months.

With these facts in mind, there is no question that SIP Trunking offers compelling advantages for businesses large and small.

What is the SIP Protocol?

Learn how Session Initiation Protocol (SIP) works.

SIP (Session Initiation Protocol) is an Application Layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include telephone calls, multimedia distribution, multimedia conferences and presence. The SIP Protocol is defined
as part of IETF RFC 3261, located at

SIP invitations are used to create sessions that carry session descriptions, which allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user’s current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features for users. SIP also offers a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols, such as UDP, TCP and TLS.

The SIP requests and responses are written in plain text within the datagram of the IP Header. Contained in the SIP requests and responses are the addresses of the source and the destination of the participants. These addresses are SIP URI’s, which have a UserInfo and Host Address, and this host address can either be an IP address or a domain name. Therefore, the routing of SIP is done using IPv4 addresses at the Application layer and does not route at the Transport or Network layer.

As the addressing and routing of SIP are done at the Application layer, the biggest problem the SIP protocol now has is the disconnect between the IPv4 addressing and routing at the Application layer versus the IPv4 addressing and routing at the Transport and Network layers. Network Address Translation (NAT) occurs at the Transport and Network layers, and thus the challenge.

911 Service Over InfoStructure VoIP

How does 911 emergency service work with VoIP telephone service?

InfoStructure provides a safe and reliable means of communication in times of emergency. InfoStructure 911 Dialing service operates differently than traditional 911, and because your safety is important to us, it is important that you provide us the street address where you will be using your InfoStructure VoIP service.

Most of our customers have access to either basic 911 or Enhanced 911 (E911) service. With E911 service, when you dial 911, your telephone number and registered address is simultaneously sent to the local emergency center assigned to your location, and emergency operators have access to the information they need to send help and call you back if necessary.

Customers in locations where the emergency center is not equipped to receive your telephone number and address have basic 911. With basic 911, the local emergency operator answering the call will not have your call back number or your exact location, so you must be prepared to give them this information. Until you give the operator your phone number, he/she may not be able to call you back or dispatch help if the call is not completed or is not forwarded, is dropped or disconnected, or if you are unable to speak. As additional local emergency centers become capable of receiving our customers’ information, InfoStructure will automatically upgrade customers with basic 911 to E911 service. InfoStructure will not give you notice of the upgrade.

You must register the physical location where you will utilize InfoStructure VoIP for each phone line. Also note that if you move your device to another location, you must register your new location. If you do not register your new location, any 911 call you make may be sent to an emergency center near your old location. You will register your initial location of use when you subscribe to the service.

IMPORTANT NOTE: 911 Dialing service over InfoStructure VoIP will not function in the event of a high-speed Internet or power outage or if your high-speed Internet or InfoStructure VoIP service is terminated.

WiFi phone: A WiFI phone is one that enables users to make phone calls from public WiFi hotspots or residential WiFI network environments. Besides voice calls, these phones can be used to send e-mails wirelessly.

Glossary of VoIP Terms

Terms used when discussion Voice over Internet Protocol (VoIP).

Analog audio signals: Analog audio signals are used to transmit voice data over telephone lines. This is done by varying or modulating the frequency of sound waves to accurately reflect the pitch of the sound. The same technology is used for radio wave transmissions.

ATA: ATA or the analog telephone adaptor is the hardware device that connects the conventional telephone to the Internet through a high speed bandwidth line, provides the interface to convert the analog voice signals into IP packets, delivers dial tone and manages the call setup.

Bandwidth: Bandwidth is the volume of data that can be transmitted over a communication line in a fixed amount of time. It is expressed in bits per second (bps) or bytes per second for digital devices and in cycles per second, or Hertz (Hz) for analog devices. Bandwidth can also be defined as the difference between a band of frequencies or wavelengths.

Broadband: It is a term used to define high speed Internet connection, generally provided by cable TV, DSL or dedicated telecom lines. The high speeds are achieved by the carrying capacity of the cable that can carry multiple messages simultaneously.

Cable modem: The cable modem is a device that is used to connect a computer to the high speed coaxial cable run by cable TV companies to provide access to the Internet. The connection is made through an Ethernet port, which is a shared medium and can affect download speeds if too many users log on simultaneously to the Internet on that particular cable segment. However, despite this cable modems provide extremely fast access to the net.

Circuit switched networks: These networks have been used for making phone calls since 1878. They use a dedicated point-to-point connection for each call. This reduces their utility because no network traffic can move across the switches that are being used to transmit a call.

Client (Softphone client): The software installed in the user’s computer to make calls over the Internet.

Codec: Codec is a term that arises from the Compressor-Decompressor or enCOder/DECoder process. It is used for software or hardware devices that can convert or transform a data stream. For instance, at the transmitting end codecs can encode a data stream or data signal for easy transmission, storage or encryption. At the receiving end, they can decode the signal in the appropriate form for viewing. They are most suitable for videoconferencing and streaming media solutions.

Compression: This is a term that is used to indicate the squeezing of data in a format that takes less space to store or less bandwidth to transmit. It is very useful in handling large graphics, audio and video files.

Data compression: This is the process that is used to compress large data files into mall files so that they use less bandwidth during transmission and less disk space when stored. The compression depends upon the repeatable patterns of binary 0s and 1s. The higher the number of repeatable patters, the higher is the compression. The right compression codes can compress data files to 40% of their original size. The graphics files can be compressed even more – from 20% to 90%.

DSL modem: A DSL modem is a device that is used to connect one or more computers to the high speed DSL line provided by a DSL operator to gain access to the Internet. The customers use these modems to log on the net to download or transmit data. Since the DSL lines have high bandwidth capacity the data transfer speeds are very high.

E911: E911 is the short form of the term Enhanced 911, and is used for providing emergency service on cellular and Internet voice calls.

Emergency 911 calls: This is an emergency telephone number that handles all calls related to police, fire or medical emergencies. The number, which is allotted under the North American Numbering Plan (NANP), is answered by either a telephone operator or an emergency service dispatcher, who, in turn, alerts the appropriate emergency service.

H.323: An ITU standard that lays down guidelines for real time voice and videoconferencing utilities on the Internet. The H.323 standard supports voice, video, data, application sharing and whiteboarding and defines media gateways for conversion to packets.

Internet congestion: Internet congestion occurs when a large volume of data is being routed on low bandwidth lines or across networks that have high latency and cannot handle large volumes. The result is slowing down of packet movement, packet loss and drop in service quality.

IP address: An IP address, also known as Internet Protocol address, is the machine number used to identify all devices that are connected to the net. Each device has its own unique number which it uses to communicate. This number is fixed in the case of those computing devices that have a fixed IP address. The rest are allotted a dynamic IP address, which is valid for the period they are connected to the net. The numbers range from to

IP mapping: IP mapping is the process of identifying IP addresses on the basis of their geographical locations. The mapping enables web administrators to pinpoint the location of any computing device connected to the Internet.

IP Phone: An IP Phone is one that converts voice into digital packets and vice versa to make phone calls over Internet possible. It has built-in IP signaling protocols such as H.323 that ensure that the voice is routed to the right destination over the net. The IP phones come with several value added services like voicemail, e-mail, call number blocking etc.

IP telephony: IP telephony refers to the two-way transmission of voice over Internet. The voice is transmitted in real time by using the packet-switched technology over the IP network. Some of the applications that use IP telephony are IP-based phone services, voice over instant messaging and videoconferencing.

IP: IP, which is the acronym for Internet Protocol, defines the way data packets, also called datagrams, should be moved between the destination and the source. More technically, it can be defined as the network layer protocol in the TCP/IP communications protocol suite.

ITU: ITU, which is the acronym of International Telecommunication Union, is a telecommunications standards body based in Geneva. It works under the aegis of the United Nations and makes recommendations on standards in telecommunications, information technology, consumer electronics, broadcasting and multimedia communications.

Jitter: It is a term used to indicate a momentary fluctuation in the transmission signal. This happens in computing when a data packet arrives either ahead or behind a standard clock cycle. In telecommunication, it may result from an abrupt variation in signal characteristics, such as the interval between successive pulses.

Kbps: Kbps is the acronym for kilobits per second and is used to indicate the data transfer speed. If the modem speed, for instance, is 1 Kbps then it means that the modem can route data at the speed of one thousand bits per second.

Lag: Lag is the term used to indicate the extra time taken by a packet of data to travel from the source computer to the destination computer and back again. The lag may be caused by poor networking or by inefficient or excessive processing.

Latency: Latency is the time that elapses between the initiation of a request for data and the start of the actual data transfer. This delay may be in nanoseconds but it is still used to judge the efficiency of networks.

Mapping: The process of identifying all related data fields or data streams and putting them in an easily identifiable context. For example, IP mapping enables users to pinpoint the geographical location of any computing device on the Internet.

MGCP: Acronym of Media Gateway Control Protocol. Used for a Voice over IP system. It consists of a Call Agent and a set of gateways, of which at least one works as the “media gateway” and performs the conversions.

NANP: Stands for North American Numbering Plan. A telephone numbering system that has evolved the way area codes and numbers are allotted. The system was established in 1947 and covers the United States, Canada and a few neighboring areas. It uses a three-digit area code and seven-digit telephone numbers. Its fiat is, however, limited to the public switched telephone networks only.

Net Phone: A net phone uses the Voice over IP technology to make voice calls. These calls are made by converting analog sound signals into digital data packets, and then moving the packets to their destination over the net.

Packet loss: Packet loss is the term used to indicate the loss of data packets during transmission over a computer network. This may happen on account of high network latency or on account of overloading of switches or routers that are unable to process or route all the incoming data.

Packet switched networks: These are networks that break messages into small digital packets, stamp each packet with the destination IP address, and route them across different channels to their destination where they are reassembled in their proper sequence. This is done to avoid network congestion and speed up data movement from multiple sources.

Packet: A packet is a unit of data transmitted over the network in a packet-switched system. It consists of a header that stores the destination address, a data area which carries the information that is being transmitted, and a trailer which contains information to prevent errors during transmission.

Peer-to-Peer (P2P): The term peer-to-peer is used to indicate a form of computing where two or more than two users can share files or CPU power. They can even transmit real time data such as telephony traffic on their highly ad hoc networks. Interestingly, the peer-to-peer network does not work on the traditional client-server model but on equal peer nodes that work both as “clients” and “servers” to other nodes on the network.

POTS: POTS is the short form of plain old telephone service. It transmits voice as analog data on communication lines that are much slower when compared to today’s ISDN or FDDI lines. However, not long ago POTS, which is also known as the public switched telephone network, was the standard telephone system across the world.

Processor drain: This is a term used to indicate a drop in the quality of VoIP phone service when a user opens several applications on his computer simultaneously.

Protocol: It is a convention or standard that defines the procedures to be adopted regarding the transmission of data between two computing end points. These procedures include the way the sending device should sign off a message or how the receiving device should indicate the receipt of a message. Similarly, the protocols also lay down guidelines for error checking, data compression, and other relevant operational details.

PSTN: PSTN, which stands for Public Switched Telephone Network, refers to the telephone system that transmits analog voice data. Till recently, PSTN was the heart of all phone systems worldwide. However, most of the developed world is now switching to or has switched to telephone networks that are based on digital technologies, such as ISDN and FDDI. RJ45: RJ45, which is the acronym of Registered Jack-45, is a telephone connector that is used in Ethernet and Token Ring Type 3 devices. It has eight “pins” or electrical connections.

Router: A router is a network device that that handles message transfer between computers that form part of the Internet. The messages, which are in the form of data packets, are forwarded to their respective IP destinations by the router. A router can also be called the junction box that routes data packets between computer networks.

Sampling: This is a methodology used to measure the value of an analog signal at regular intervals, and encoding it into a digital format for VoIP phone services.

Service provider: A service provider is a business entity that provides a communication, storage or processing service for a fee. Some of the service providers in the digital world are the Internet service provider (ISP), application service provider (ASP), storage service provider, mobile phone service provider, web hosting provider, and of course, VoIP Service Provider.

SIP phone: A SIP phone is a telephone that uses the SIP (Session Initiation Protocol) standard to make a voice call over the Internet. The SIP phones come with several value added services like voicemail, e-mail, call number blocking etc. There are no charges for making calls from one SIP phone to another, and negligible charges for routing the call from a SIP phone to a PSTN phone.

SIP: SIP, which is the acronym of Session Initiation Protocol, is an IP telephony signaling protocol. It is primarily used for voice over IP (VoIP) calls, though with some extensions it can also be used for instant messaging. It is less complex than H.323, the other IP telephony protocol.

Soft switch: It is a software application that is used to keep track of, monitor or regulate connections at the junction point between circuit and packet networks. This software is loaded in computers and is now replacing hardware switches on most telecom networks.

Softphone: This is a software application that is installed in the user’s PC. It uses the Voice over IP technology to route voice calls over the net and provides several value added features, such as call forwarding, conference calling, and integration with applications such as Outlook for automatic dialing The audio is provided through a microphone and speakers plugged into the sound card. The only limitation of a Softphone is that the phone call has to made through a PC.

Voice chat: This is an application that enables two or more than two individuals to carry on a verbal conversation over the Internet. Voice chat is also known as audio-conferencing or telephone conferencing on the net.

Voice over IP (VoIP): VoIP or Voice Over Internet Protocol is the technology that is used to transmit voice over the Internet. The voice is first converted into digital data which is then organized into small packets. These packets are stamped with the destination IP address and routed over the Internet. At the receiving end the digital data is reconverted into voice and fed into the user’s phone.

Voicemail: It is a telephone messaging system that digitizes the analog voice signals and stores them on disk or flash memory in a central computer. These messages can then be retrieved by users by logging on to the server or forwarded to another voice mailbox. Most voice mail systems have auto attendant capabilities, that is they can use prerecorded messages to route callers to the appropriate person or mailbox. Voicemail is usually a free feature in VoIP service plans

IM: IM, which stands for Instant Messenging, is a software that allows users to exchange messages in real time. However, to do so both the users must be logged on to the instant messaging service at the same time. Some of the popular IM services are: MSN Messenger, AOL Instant Messenger, Yahoo! Messenger, Google Talk and ICQ.

VoIP Gateway: This device provides the conversion interface between the public switched telephone network (PSTN) and an IP network for voice and fax calls. Its primary functions include: voice and fax compression/decompression, packetization, call routing and control signaling. It also provides an interface to Gatekeepers or Softswitches, billing systems, and network management systems.

VoIP PBX: VoIP PBX, which stands for Voice over Internet Protocol Private Branch eXchange, is a telephone switch that converts IP phone calls into traditional circuit-switched TDM connections. It also supports traditional analog and digital telephones.

VoIP Phone: A VoIP Phone is one that uses the Internet to route voice calls by converting the voice data into IP packets and vice versa. The phones come with built-in IP signaling protocols such as H.323 or SIP that help in the routing of data to the right destination. A VoIP phone can also be a software application that is installed in the user’s PC. In this case it is known as the Softphone. Also, the calls in this case have to be made from the PC, and not through a telephone instrument.

VoIP services: The VoIP Services are packet-based services that use the Internet to move voice data. These services are much cheaper than the traditional PSTN services because the investment in infrastructure is low. They also come with several value added features which make them more lucrative than the conventional landline phone services.

Web phone: A web phone is a device that allows users to make voice calls over the Internet.

WiFi Hotspot: An area where a wireless access point enables users carrying wireless-enabled laptops to log on to the Internet. The limiting condition is that the access point is configured to broadcast its presence and does not require authorization for access. Generally, WiFI hotspots are located in public places like airports, train stations, libraries, marinas, convention centers, coffee shops and hotels.

WiFi phone: A WiFI phone is one that enables users to make phone calls from public WiFi hotspots or residential WiFI network environments. Besides voice calls, these phones can be used to send e-mails wirelessly.

VoIP Frequently Asked Questions

Home & Small Office VoIP FAQ

Q. What do I need to get InfoStructure VoIP Service?

A. The requirements to have InfoStructure VoIP are fairly simple. You need the following:

A high-speed Internet connection
A billing and shipping address
An InfoStructure approved phone adapter
A touchtone phone

Q. Is InfoStructure VoIP hard to install?

A. No. It is very simple. You plug the phone adapter into your high-speed Internet Modem then plug your telephone into the phone adapter. You then plug the phone adapter into power and that is it!

Q. Does my computer need to be on for InfoStructure VoIP to work?

A. No. You can use the service without your computer being on. Your high-speed Internet connection needs to be working (the modem) and your phone adapter needs to be plugged into it.

Q. Can I make calls while I am using the Internet on my computer?

A. Yes. You can use your computer, the Internet and your phone at the same time.

Q. Can I keep my existing phone number?

A. Yes! InfoStructure has the ability to port your existing phone number in most parts of the Continental United States. To find out if your area is supported, simply email or call us at 800.419.4804.

Q. Will it work with the phone wiring & phone jacks in my home with more than one phone?

A. InfoStructure supports one phone connected to the back of the VoIP phone adapter, however the system does work with home wiring. It requires that you or a phone technician do a bit of simple wiring work and InfoStructure VoIP will work on the wiring thoughout your home. InfoStructure does not do this wiring.

Q. Does InfoStructure provide the high-speed Internet connection?

A. InfoStructure can provide DSL and other high-speed Internet services in most areas. You can also use any high-speed Internet connection from any provider with InfoStructure VoIP. InfoStructure is a well established Phone Company and Internet Service Provider. If you would like to find out more about getting your Internet Connection from InfoStructure, visit our High Speed Internet page, email or call us at 800.419.4804.

Q. If I move, will I have to cancel service and start it again?

A. No. Because you can use InfoStructure VoIP in any location with a high-speed Internet connection, you’ll never have to cancel your phone service. Just take your phone adapter & telephone with you and set it up on the Internet connection at your new location.

Q. Does InfoStructure VoIP work with 911 service in an emergency?

A. Yes. There are details you should know about 911 service over VoIP before you place your order.

Q. Will my FAX work with InfoStructure VoIP?

A. Yes. In most cases, FAX machines work with InfoStructure VoIP.

An Overview of IP Voice

Learn the basic networking concepts of IP voice for business and home

The ability to place and receive phone calls over the Internet Protocol is commonly called IP voice. It’s also identified by the underlying protocol known as VOIP (voice over the Internet Protocol) or its signaling protocol – SIP (session initiation protocol). Often the terms SIP and VOIP are combined as SIP VOIP to represent IP voice. IP voice technology leverages the versatility and global presence of the Internet. Though IP voice has been around for over a decade, it has undergone several evolutions since its inception. Despite the excellence of early-day core and edge technology by the likes of Cisco, Nortel and Avaya, the Internet of ten years ago was not quite ready to transmit voice because it was engineered mainly for computers, not people, to talk to each other. Since then, the Internet has been reconfigured for voice, and technology has been refined to the extent that the user experience of VOIP is as good as traditional business phone lines. Moreover, the inherent flexibility, mobility, scalability, and economies of VOIP technology have made it the technology of choice for millions of businesses and consumers worldwide.

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