|
|
FAQFrom KeyfonicsGeneral QuestionsWhy should I use a VoIP Phone System?Most people know that calling via a VoIP system can save you a lot of money when compared to a traditional telephone system. However, it's not just about reducing call costs: - other benefits include:
What are the main advantages of VoIP over analogue (PSTN) lines?The major advantage of VoIP over PSTN lines is cost, but there are numerous other advantages too: VoIP lines include digital features not commonly available on PSTN lines such as voicemail, caller ID, conference, music-on-hold, etc. Keyfonics provides over 90 advanced features at no extra cost with each price plan. How do I know if my existing network and telephony layout can use VoIP?Nearly all modern business environments can use VoIP. The only absolute requirement is that you have access to a broadband internet connection. Check out some typical business scenarios here. To support multiple extensions your existing network must be able to connect to the Internet. The only network settings we require are your DHCP server settings (running either on a network server or on an ADSL router) to enable phones to get their settings from our provisioning server. Also VoIP traffic requires certain ports to be opened on your firewall to allow call setup and voice traffic to cross the WAN (unless uPNP is being used on your routers). SIP typically uses port 5060 while RTP uses a pre-defined range (typically 16384-32767). We may need to talk to your IT personnel to get these settings, or you can provide them on our survey form. Do I need a computer to make or receive VoIP calls?No. A digital VoIP phone can just plug straight into your network (many VoIP phones also include a 2-port switch so they can share a single LAN connection with a PC if required). Traditional analogue phone can also be used by connecting them to Analogue Telephone Adapters (ATAs). An ATA is a box about the size of a packet of cigarettes that connects to your LAN and typically provides 2 standard telephone sockets). You will need a computer if you're going to use a softphone with your VoIP integration. A softphone is a piece of software that runs on a PC and provides features similar to a hardware VoIP phone. We recommend that softphones be used with headsets to provide optimum sound quality. What do I need to start using your service?
Can I use the Internet as normal for Internet access and email during VoIP calls?Yes, VoIP allows web surfing while making and receiving VoIP calls simultaneously. It shares the bandwidth connection with other LAN computers and, depending on the network hardware, can prioritize voice traffic to ensure high quality when the connection is shared. What about call quality and reliability?We find that as long as customers have sufficient internet bandwidth to operate the service correctly, they experience excellent call quality. Check out our bandwidth requirements for general guidelines. If you want more specific recommendations then request a free site audit. We use our own service and are usually among the first to become aware of any problems so issues are resolved quickly. Most of the problems we encounter involve equipment and Internet connection issues during the initial setup, and many of these can be avoided by filling in our site survey form. Call quality is highly subjective and is very much in the ear of the beholder, but using the industry standard for measuring sound quality called MOS or Mean Opinion Score where 1 is unintelligible and 5 is superb sound quality, then a typical G711 (u-law) call scores 4.4 (similar to or better than a landline) while a G729a call scores 3.92 (similar to a mobile phone call). Do I have to have broadband to use VoIP?Yes. ISDN and dialup modem users can use VOIP but they won't like it and we don't recommend it. On a standard ADSL line with 384 kbps upstream, you can get 3 to 4 concurrent calls. ADSL Max lines can support around 5-6 concurrent calls. For a busy office of 10 - 12 people, customers typically would upgrade their ADSL line to ADSL 2+ which gives a usable upstream bandwidth of 1.25Mb or so, depending on the length of the line to your premises. For every further 10 - 12 users another ADSL2+ line would be needed. These figures are based on a busy office with nearly all users using their phones at the same time. If your requirements differ you may be able to support more users with fewer lines. Can I keep my existing numbers?Yes, if you have a BT number it can be ported to our system (subject to survey). The process requires that you sign up with a temporary new number then we put the port through, and replace the temporary new number. Number porting takes 7 - 14 working days for a simple BT line. It's important to remember that when you port a number away from BT, BT will cease that line, and all its associated services (such as broadband and FeatureLine). What will my new number look like?A single UK geographical number (e.g. 0114 123 45678) is associated with your network of extensions. You can choose any UK area code (or a more memorable Silver and Gold number), or keep your existing BT number if you prefer. How many numbers do I need?It depends on how many calls you expect to have at any one time e.g. our 10-line package gives 10 inbound lines and as many simultaneous outbound calls as your network will allow. If you require DDI (Direct Dial In - an external number associated with a particular extension) then you'll need one line per DDI number. Can I divert calls to another phone number or Voicemail?Yes. A user can choose to redirect all calls until further notice, or only when their extension is busy, or only on no-answer. If the redirection is to an external number (e.g. a mobile) then the outbound call will be charged at the prevailing rate. Are the call charges from a landline/mobile to a VoIP phone different?No. We provide a regular UK landline number (or you continue to use your ported number) so people pay the same to call you as before. Can I use my fax machine?VoIP lines are optimised for voice and are not well suited to the high-frequencies used for fax. We therefore do not recommend putting fax machines on VoIP lines and would suggest attaching fax machines to the PSTN line which hosts your ADSL. How much does it cost?Please take a look at our line package plans. Can I take my number with me when I travel?Yes, as long as you have a high-speed Internet connection available and your phone adapter is with you. In this case, calling would work the same as if you were dialling from your home or business and you would not incur any additional charges by being abroad. If you use a softphone on your PC you can use it anywhere in the world where there is a reliable broadband connection (including wireless), though remote areas may offer reduced call quality. Will my VoIP phone work if the power fails?No. Most ordinary analogue phones don’t need a separate power connection but, as your VoIP phone relies on your Internet connection, if your broadband router has no power, your phone won't work either. Are there any phone numbers that VoIP phones can't call?You can ring most phone numbers, apart from UK premium numbers that start with 09. The big exception is the emergency services. You can't call 999 on a VoIP phone at present, so in an emergency you would need to use a mobile or landline phone.
Technical QuestionsWhat's a softphone?A softphone is a computer application that allows users to make telephone calls directly from their computer. The software for a softphone usually mimics the appearance of a real handset and can take the form of either a standalone program with its own window, or an embedded program in a Web application or other PC program. Commands can be entered through either the keyboard or the onscreen interface. Conversations are conducted on a headset with a built-in microphone, with a microphone and the computer’s speakers, or on a USB phone -- a handset that plugs into computers’ USB ports. The computer’s sound card is used to provide audio input and output for the softphone. What’s the difference between a VoIP and SIP phone?None. It's just another name for a VoIP phone. SIP (Session Initiation Protocol) is the name of the protocol used to actually setup a call. What’s best: an ATA or IP Phone?It depends on your preference and budget. An ATA (Analogue Terminal Adaptor) will allow you to use existing analogue phones with VoIP, but any analogue feature keys (e.g. transfer, hold, BT Star services etc) will not work. IP phones are much more feature-rich and provide many more features. Call quality is generally similar for both types. What's the difference between G.729a and G.711?G.711 and G.729a are both standards-based voice codecs (encoder-decoders) that take the voice signal and compress it for transmission across the Internet. G.711 (or PCM) is the 'granddaddy' of codecs and has been used on the traditional telephone network for years. It provides excellent call quality with minimum compression at the expense of using more bandwidth (typically around 80 kbps per channel). G.729a is much more highly-compressed and offers slightly-reduced call quality in return for a significant reduction in bandwidth usage (typically 10 Kbps per channel). Because of the extra compression, G.729a will consume more CPU power per channel than G.711. The codec used is negotiated between the calling and called party’s hardware. If a called user’s hardware device or VoIP provider only supports one codec then this is the codec that will be used. Keyfonics supports the above mentioned codecs plus G.723, G.726 and GSM but virtually all modern IP phones will support at minimum G.711 and G.729a. Which VoIP signalling protocols are commonly used?VoIP signalling protocols are used to setup and tear down calls, carry the required information to locate end users, and negotiate device capabilities. Keyfonics supports both the SIP (Session Initiation Protocol) and IAX (Inter-Asterisk Exchange) protocols.
Jargon BusterWhat is VoIP?VoIP stands for Voice Over Internet Protocol and is a standards-based technology that enables you to make and receive telephone calls through the Internet instead of using traditional analogue PSTN (Public Switched Telephone Network) lines. Keyfonics allows you to use your VoIP phone to call a traditional landline via the PSTN at significantly reduced rates. Calls to other Keyfonics customers can be made without using the PSTN at all, and are free. What are IP PBXs?IP PBXs (Private Branch Exchanges) are complete phone systems that provide advanced telephony features and services between VoIP and PSTN networks. Common features and services include: call transfer, conference, voicemail, music on-hold, auto-attendant (IVR), and auto call routing. The Keyfonics service provides a hosted IP PBX server via the Internet. What's an IVR?An IVR (Interactive Voice Response) system is a "digital receptionist" that issues voice prompts and responds to caller key presses. Usually used to route callers to the appropriate department or provide pre-recorded information e.g. “Press 1 for Sales, 2 for Technical, 3 for Customer service or hold for any other queries”. What are VoIP Gateways?VoIP gateways are devices that take analogue voice signals and convert them to IP for transport over the LAN or WAN. What are FXO and FXS ports?Foreign Exchange Office (FXO) ports are interfaces (typically a phone or fax) used to connect with the central office or PSTN analogue lines. Foreign Exchange Station (FXS) ports are interfaces used to connect with end user devices (e.g. phone or fax) and are used to provide the bell current and current to drive the FXO device. What are PSTN failover lines?PSTN (Public Switched Telephone Network) failover lines are used as backup connections in the event your VoIP or Internet connection goes down. These are optional ports on ATA devices or IP phones that connect directly to the analogue PSTN lines coming from the telephone company. This setup requires having both regular analogue telephone lines and an account with us. What are latency, jitter, and packet loss and how do they affect a call?LatencyLatency is the time between the moment a voice packet is transmitted and the moment it reaches its destination, and is measured in milliseconds (ms) - thousandths of a second. Too much of it leads to noticeable delays and potentially an obvious echo on slow network links. A latency of 150ms or less is barely noticeable so is deemed acceptable. Higher than that, quality starts to suffer. When latency reaches 300 ms or more, voice quality becomes unacceptable. JitterJitter refers to how variable latency is in a network. High jitter, greater than approximately 50 ms, can result in both increased latency and packet loss. When talking to someone it's important that they hear what you say in the same order that you say it, otherwise they won't understand what you're telling them. Unfortunately, jitter causes packets to arrive at their destination with different timing and possibly in a different order than they were sent (spoken), with some arriving faster and some slower than they should. To correct the effects of jitter, VoIP phones collect packets in a buffer and put them back together in the proper timing and order before the receiver hears them. This works, but it's a balancing act. Processing that buffer adds delay to the call, so the bigger the buffer, the longer the delay. Remember the effects of latency? Keep in mind, no matter how big the buffer is, it is finite in size. If voice packets arrive when the buffer is full then packets are dropped and the receiver will never hear them. These are called discarded packets. The latest IP phones from Snom, Aastra and Polycom all have effective jitter buffers which can negate the effects of jitter over good network connections. Packet LossJust as it's important to hear what someone says in the order they say it, it's also important to hear all of what they're saying. If you miss one out of every 10 words or 10 words all at once, chances are you're not going to understand much of the conversation. This is packet loss — some of the voice packets are dropped by network routers or switches that become congested (lost packets), or discarded by the jitter buffer (discarded packets). Knowing the average packet loss for a call gives you an overall sense for the quality of the call. A call with less than 1 percent average packet loss will always sound better than a call with 10 percent loss. But average loss doesn't tell the whole story. You need to know what type of packet loss you encountered. There are two kinds of packet loss: "random" and "bursty". Think about two calls each with average 1 percent packet loss. Call A loses one in every 100 packets over the entire call (random loss) while Call B loses 100 packets in two clumps at the beginning and the end of the call (bursty loss). For a given percentage of loss, bursty loss is usually more noticeable than random loss. What is DDI?Direct Dial In - a unique external number that will be routed directly to a particular extension. What is RTP?Real-time Transport Protocol (RTP) provides end-to-end network transport functions suitable for applications transmitting real-time data such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of- service for real-time services. RTP is used to transmit voice packets in a VoIP system. What is Provisioning?Provisioning enables us to set up your phones remotely from our system and update or alter settings without any customer intervention. This enables us to define global settings across all phones, or alter settings on a specific phone e.g. setup the extension number or user details as required by your environment. With the appropriate information we can preconfigure the phones prior to delivery for you to install yourself, or we can do it for you via our on-site installation service. |