IP Office VoIP

Office Phone System

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Telefonix credentials include Avaya SME Expert, Avaya Partner in Customer Excellence and Microsoft Small Business Specialist. Besides our experience of the IP Office system with over 2500 installs under our belt, Telefonix Voice & Data also has a unique set of IP Office phone system packages unique to the UK.

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IP Office VoIP

Technological innovation is changing the way we communicate. This time it is coming in the form of changing the way telephone calls are transmitted. It brings with it several new capabilities that change the meaning of the phrase telephone call through the use of Voice over Internet Protocol (VoIP). Basically, VoIP means "voice transmitted over a packet data network." VoIP is often referred to as IP Telephony because it uses the IP protocols to make possible enhanced voice communications throughout the world, wherever IP connections have been delivered. IP Telephony unites a company's many locations- including mobile workers- into a single converged communications network. Telephony calls using VoIP go above and beyond what's been possible in the past. When it comes to placing telephone calls, VoIP provides a range of support services and features unequalled in the world of telephony, but above all deliver them at low cost.

How Does VoIP Work?

Voice over Internet Protocol means basically what the acronym states: Voice travels over an Internet Protocol. Internet Protocol refers to the type of rules that the network uses to send and receive signals. IP Telephony works by converting voice communications into data packets. Conveniently, it runs on the popular Ethernet LAN (local area network) technology, which currently supports over 96 percent of the worlds companies' LANs.

Circuit-switched or Time-Division Multiplexed Telephony

Before digital networking with the Internet took off, everyone had to use the "Plain Old Telephone Services" (POTS). These run over a network called the Public Switched Telephone Network (PSTN). The PSTN has been around since the telephone was invented in either analog or digital form using circuit switched technology where the telephone call gets exclusive bi-directional use of a wire or circuit while the call is in progress. Because the circuit is exclusive to each conversation, PSTN and private branch exchanges (PBXs) must be sized to cope with peak demand and have enough circuits available for all expected conversations. This is not a flexible approach and results in a lot of infrastructure investment that the telephone companies need to recoup, via the cost of access charges and calls. The Internet has changed this where data services have driven down access charges and allowed voice to "travel for free" over a multipurpose data network.

Packet-Switched Telephony

Unlike circuit-switched connections, which always require use of dedicated bi-directional circuit for the duration of a call, VoIP technology has enabled telephony and other new and novel features and services to run over fixed and wireless networks including private local area networks. These newer network types use packet-switched protocols. Packet-switched VoIP puts voice signals into packets. Along with the voice signals, VoIP packets include both the sender's and receiver's network addresses. VoIP packets can traverse any VoIP-compatible network. Along the way, they can choose alternate, shared paths because the destination address is included in the packet. The routing of the packets is not dependent on any particular network route which means the network provides can provide a reliable service at a fraction of the cost of circuit switched providers.

What Advantage Does IP Office Have?

IP Office can provide support of PSTN, SIP, POTs, digital time division multiplexed telephones AND digital IP telephones all on the same system. This means you don't have to abandon the past to embrace the future, IP Office allows all the technologies to co-exist. IP Office connects to the PSTN and to IP trunks (the VoIP equivalent) so providing a "Hybrid" PBX function - where both legacy and future technologies can be used together to minimize operating costs and offer optimize business communications through both voice and data. IP Office has digital telephones built on both TDM and IP technology that provide the same user interface offering a flexible choice of solution that can mix, for example TDM telephones in the office and IP telephones at a remote site of at home. With the choice of IP telephones including real and virtual (software) telephones, IP Office can take communications to a new level.

Buying IP Office allows you choice - you can use the pure POTs or the pure VoIP capabilities of IP Office, or use both at the same time to allow seamless technology transition of your business without the disruption of having to choose between them now.

IP Office Turns VoIP into IP Telephony

In order to make use of VoIP, IP Office uses signaling protocols called H.323 , and Session Initiation Protocol (SIP) which allow IP Office to establish end-to-end connections for the voice path through the IP network. It ensures each end of the connection is able to transmit and receive voice and provides the network addressing for end to end packet transmission. IP Office also allows for connecting between the different technologies by translating the signals they use, for example an analog telephone may wish to connect to a VoIP destination. This requires both the signaling and voice transmission to be translated. IP Office does this easily as it contains technology elements called gateways and gatekeepers that enable translations to happen. With a conventional telephone system you plug your analog or digital TDM telephone into an extension socket connected to your PBX or Key System. With IP Telephony you connect your digital IP telephone to your IP PBX via the LAN. There are two basic types of IP telephones:

  • A physical telephone, which looks very similar to a standard telephone (IP Hard Phone)
  • A software application (IP Office Video Softphone) which runs on the user's PC, allowing them to use either a headset/microphone to make/receive calls anywhere they have IP connection

IP telephony has the advantage of allowing extensions to be deployed both locally and remotely through the use of IP routing and IP VPN services. When making use of IP telephony, there are a number of data centric considerations such as which data types have priority on the IP network when there is contention. This is set with IP/TCP "quality of service" and should not be ignored. In situations where LAN Bandwidth is limited, a quality of service capable LAN switch should be used to ensure voice packets are transmitted with the required priority on the network. If not, the conversation carried over IP appears as broken up (due to delays) or has unacceptable delays introduced in the conversation (latency and jitter).

IP Office Protocol Support

IP Office supports the following protocols and standards:

  • H.323 V2 (1998), Packet-based multimedia communications systems.
  • Q.931, ISDN user-network interface layer 3 specification for basic call control.
  • H.225.0 (1998), Call signaling protocols and media stream packetization for packet-based multimedia communication systems.
  • H.245 (1998), Control protocol for multimedia communication.
  • Session Initiation Protocol.
  • Audio CODECs:
  • G.711 A-law/U-law (64K).
  • G.723.1 MP-MLQ (6.3K).
  • G.729 Annex A, Annex B, Annex AB
  • Silence Suppression.
  • Fax Relay (IP Office to IP Office Fax Transport over IP).
  • T.38 Fax support (SIP trunks and SIP endpoints).
  • Local End Echo Cancellation 25ms.
  • Out of band DTMF.
  • Jitter buffer, 5 frames of jitter buffer.
  • Internet Standards/Specification (in addition to TCP/UDP/IP).
  • RFC 1889 - RTP/RTCP, Real Time and Real Time Control Protocol.
  • RFC 2507, 2508, 2509 - Header Compression.
  • RFC 2474 - DiffServ, Type of Service field configurable.
  • RFC 1990 - PPP Fragmentation.
  • RFC 1490 - Encapsulation for Frame Relay.
  • RFC 2686 - Multiclass Extensions to Multilink PPP.
  • RFC 3261 - Session Initiation Protocol (SIP).
  • RFC 3489 - STUN.

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