PBX SIP VOIP UC WIFI what is this stuff

Links

These guys are in Ottawa – http://www.versature.com/about/contact-us

http://www.3cx.com/PBX/pbx-phone-system.html

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

PBX, IP-PBX

PBX is an abbreviation for Private Branch Exchange that is installed in a business location to facilitate communication between people inside the organization while allowing access to adequate external telephone lines. IP-PBX a digital (software) PBX system that use digital technology and Internet protocol (IP) to route phone conversations to the proper telephone handset
With a software-based telephone, a user plugs a headset into a computer and uses a virtual telephone to dial and receive telephone calls. Another advantage of an IP-based PBX is the ability to transmit calls internationally over the Internet, thereby avoiding long-distance charge.

Mobile PBX

A mobile PBX is a hosted PBX service that extends fixed-line PBX functionality to mobile devices such as cellular handsets, smartphones and PDA phones by provisioning them as extensions. Mobile PBX services also can include fixed-line phones. Mobile PBX systems are different from other hosted PBX systems that simply forward data or calls to mobile phones by allowing the mobile phone itself, through the use of buttons, keys and other input devices, to control PBX phone functions and to manage communications without having to call into the system first.

A mobile PBX may exploit the functionality available in smartphones to run custom applications to implement the PBX specific functionality. In addition, a mobile PBX may create extension identifiers for each handset that allow to dial other cell phones in the PBX via their extension shortcut, instead of a PSTN number.

SIP

The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). SIP was accepted as a 3GPP signalling protocol and permanent element of the IP Multimedia Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular systems. SIP is similar to the HTTP request/response transaction model. Each transaction consists of a client request that invokes a method or function on the server and at least one response. SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format. SIP is to provide a signalling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN).

SIP is a peer-to-peer protocol, thus it requires only a simple (and thus scalable) core network with intelligence distributed to the network edge, embedded in endpoints (terminating devices built in either hardware or software). SIP features are implemented in the communicating endpoints (i.e. at the edge of the network) contrary to traditional SS7 features, which are implemented in the network. Although several other VoIP signaling protocols exist (such as BICC, H.323, MGCP, MEGACO), SIP is distinguished by its proponents for having roots in the IP community rather than the telecommunications industry. SIP has been standardized and governed primarily by the IETF, while other protocols, such as H.323, have traditionally been associated with the International Telecommunication Union (ITU).

Each resource of a SIP network, such as a User Agent or a voicemail box, is identified by a Uniform Resource Identifier (URI), based on the general standard syntax also used in Web services and e-mail. A typical SIP URI is of the form: sip:username:password@host:port. The URI scheme used for SIP is sip:. If secure transmission is required, the scheme sips: is used and SIP messages must be transported over Transport Layer Security (TLS). A proxy server "is an intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is sent to another entity "closer" to the targeted user. Proxies are also useful for enforcing policy (for example, making sure a user is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it."

Many VoIP phone companies allow customers to use their own SIP devices, as SIP-capable telephone sets, or softphones. The market for consumer SIP devices continues to expand, there are many devices such as SIP Terminal Adapters, SIP Gateways, and SIP Trunking services providing replacements for ISDN telephone lines.

The free software community started to provide more and more of the SIP technology required to build both end points as well as proxy and registrar servers. As an example, the open source community at SIPfoundry actively develops a variety of SIP stacks, client applications and SDKs, in addition to entire private branch exchange (IP PBX) solutions that compete in the market against mostly proprietary IP PBX implementations from established vendors.

Unified Communications

Until very recently, voice and data communications in the business environment were not unified, telephones were simple black boxes sitting on the users’ desks, connected via their own network and completely isolated from the company’s computer data network. UC 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.

With the evolution of telephony toward IP (internet protocol), the integration of both communication worlds has become possible, allowing for companies to transform their business processes. This development of breaking down communication barriers and allowing people using different forms of communication, different devices, and different media to communicate to anyone, anywhere and at any time is known as Unified Communications.

What is FXs and FXO

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.

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
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 with a PBX

FXS & FXO & VOIP

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.

FXO gateway

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.

FXS gateway

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 (ATA) adaptor

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