If it's about VoIP, SIP or Internet Telephony but it doesn't seem to fit anywhere else, post it here.
User avatar
By WelshPaul
#140
Some VoIP providers allow you to have as many simultaneous calls (these are equivalent to the old concept of telephone 'lines') on your service as you can handle locally. Unlike ordinary telephony if you want 1000 calls sent to you all at once, VoIP can do it!

But you need to make sure that your local set-up is capable of handling these calls.

There are three prime considerations which dictate both the quality and number of simultaneous calls you can make or receive over your VoIP service.
  1. Your internet connection

    Broadband ADSL has a lower bandwidth back up to your ISP than down from them i.e. it is asymmetrical. Typically, ADSL bandwidth is described as up to 8Mbps or 24Mbps which is the down speed. But because VoIP is symmetrical - a telephone call is, of course, two-way, most people listen as well as talk - the major bottleneck is your upstream bandwidth which is normally much lower, usually something between 256Kbps and 2Mbps. The more bandwidth you have the more simultaneous calls you can achieve and the less interference you will have from your use of other internet applications such as internet browsing, file downloading, FTP or email (see below).

    So, as a general principle, always get the fastest ADSL service you can find and afford. A business ADSL service will have a lower contention ratio than a residential one and will be less congested. Find a good ISP - they are not all equal!

    Voipfone provide their own voice prioritised ADSL service which is specifically designed for use with their network. If you are looking for a business class, broadband service look here first as we can then control the whole call from your phone into our network.

    http://www.voipfone.co.uk/broadband.php

    If you are really heavy users you could also consider buying symmetrical DSL - SDSL.
  2. Your use of your internet connection

    Unlike downloading a file or sending an email, a telephone call is instant and happens in real time & it cannot be delayed, slowed down or paused.

    So if you wish to use both voice and data very heavily you need to design your network correctly. If you are finding that your calls suffer from jitter & calls stuttering and breaking up & it is almost certainly being caused by lack of bandwidth as your voice traffic competes with other data traffic for scarce bandwidth.

    One solution is to use Quality of Service (QoS) equipment to manage the traffic over your network. This is now included in some everyday routers and can ensure that your voice services get priority over all other traffic on your network.
    If you have several people all sharing the same internet connection for voice and data and are using it heavily for both, you should consider separating your voice connection (VoIP) from your data connection (internet, email etc). This means using two ADSL circuits. This also has the benefit of giving you a back-up if one circuit fails. To obtain an even higher level of back-up, use a two different Service Providers.

    It is also worth ensuring that your local network wiring is up to scratch - bad internal cabling can lead to all sorts of problems. Avoid using WiFi for VoIP; it works, particularly with devices specifically designed for it, but for PCs and softphones the encryption/decryption process adds complexity and slows things down.

    There are diagrams here showing typical network setups:
    http://www.ukvoipforums.com/viewtopic.php?f=9&t=61

    Another option is to change to a codec that uses less bandwidth.
  3. Your choice of codec

    Since voice and sound are analogue, they need to be converted (or encoded) to a digital format suitable for transmission over the Internet. A codec is an algorithm used to do this job; it codes and decodes a voice conversation.

    All VoIP telephones, both softphones and telephones and adapters, use codec's and unless you specify which one, it will use the default codec.

    There are a variety of different ways this encoding and decoding can be done - many of which utilise compression in order to reduce the required bandwidth of the conversation. Reducing the bandwidth will reduce the quality of the call somewhat but will enable you to squeeze more simultaneous conversations over your connection. Getting this trade-off right is up to you!

    Your choice of codec will radically affect the number of simultaneous conversations over your network so it is worth experimenting with them. We recommend using two in particular:
    • G711a This is a very high quality codec which delivers CD quality sound & much better than an ordinary telephone call. But to do this it uses quite a lot of bandwidth & around 90Kbps which reduces the number of simultaneous calls to 2 for a connection with an upload of 256Kbps.
    • GSM This is the codec used by mobile phones and delivers the same sort of quality. It uses less bandwidth than G711a, so is good if you need to have lots of calls in progress on a limited bandwidth connection. It uses around 35Kbps; this will allow 5 calls on a connection with an upload of 256Kbps.
    For guidance, the chart below shows the maximum simultaneous calls theoretically possible over a perfect connection. Bear in mind, that in the old telecoms world, each call requires one line so using a GSM codec can give you the equivalent of up to 17 lines on a standard ADSL connection!
    image.jpg
    Of course, the number of possible simultaneous conversations does not define the maximum number of extensions you can have because not all extensions will be in use simultaneously. The relationship of extensions to calls is called the contention ratio and it will vary for every business.

    A sophisticated call centre using call management software may require a contention ratio of almost 1:1 while a normal office may be more like 5:1 (ie only 20% of your extensions are expected to be making a call at any one time. The only real way to work out how many extensions your network can support is to trial it.

    (When calculating your bandwidth requirements you also need to bear in mind that an internal call, extension to extension, counts as two simultaneous calls.)
Other considerations
  1. We always recommend a physical cable connection (Ethernet, cat5e or Cat6) from your router to your phone or PC rather than a wireless connection. WiFi works well enough, but it adds complexity and lag, so if you can, it is best avoided for routine business use.
  2. Your router. Some routers are better than others and it doesn't seem to be a cost issue. Commercial routers are expensive and don't seem to help much with VoIP - they tend to try to be too clever. Our members are finding that the simple routers are the best. A basic Linksys or Netgear (non-wifi, non-VoIP 4 port router plus a good switch in front of it such as the HP Procurve seems to be as good as any.)
  3. We recommend using a dedicated, good quality, SIP telephone. Adapters (ATAs) work very well, but they are in the end, adapters, not telephones and they offer reduced functionality over a good phone. (However, ATAs attached to DECT phones make an excellent wireless VoIP solution.)
  4. We don't recommend using a softphone on your PC for serious business use as their quality is never the best and your PC must be switched on all the time. Many people do though - and seem happy with the result. If you do use a softphone find a good quality mono headset for it.
  5. Configuring your equipment correctly is vital to the quality of service you receive. Poorly configured and out of date firmware in both phones and routers is often the cause of poor call quality and the frequent un-registering of phones. Make sure that whatever equipment you are using is kept up to date because improvements in VoIP technologies are happening daily.
Who is online

Users browsing this forum: CommonCrawl [Bot] and 0 guests

Supported Products: OBi504vs OBi508vs Firm…

Supported Products: OBi200 OBi202 OBi300 OB…

Enter your email address here: https://haveibeenpw…

Well, with VoiceHost you can! I needed to report …