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VoIP designed for your business
VoIP offers significant benefits to your business so its critical that you work with a business VoIP Provider who can provide you with the technical skills to optimise your benefits and ensure a hassle free implementation. VoIP operates under a number of different protocols including H323, SIP and IAX. In addition some VoIP Providers use their own proprietary solution, limiting the ability to capitalise on the wide range of open source hardware available in the market place.
SIP was created by the Internet Engineering Task Force (IETF) and gained prominence around 2002 when Microsoft selected it as the standard for VoIP providers who were included in the launch of Microsoft Windows XP or as part of MSN Messenger. Since then SIP has become the accepted standard for most end users with a proliferation of PC software, IP Phones, VoIP Gateways and Analogue Telephone Adaptors (ATA). As an Internet designed protocol SIP is much less rigid than H323. Most SIP VoIP devices can readily transverse quite complex firewall and network designs and are almost plug and play. Additional protocols continue to develop as VoIP becomes established as the standard telecommunication technology.
Because VoIP converts data into small packages and uses the public Internet there are several technical issues that need to be addressed:
Delay/Network Latency is inherent in VoIP technology since the VoIP packets have to travel from the origination point to the destination through the end users own network and the public internet. Some of these delays are fixed by the laws of physics but delays within the end user network can be minimised by giving VoIP packets priority over other network traffic. Alternatively, for multiple users it may be advisable to have a dedicated ADSL or cable connection for the VoIP service. When VoIP traffic is not in conflict with other internet traffic the delay and network latency is not noticeable in voice conversation.
Packet Loss is simply where an individual packet has not arrived at the destination within the designed time frame and therefore creates a gap in the conversation. A single packet loss may have no discernible impact on the conversation but a larger element can make the conversation difficult to understand. Packet loss is often due to network congestion and within the confines of the end user network it is again possible to minimise this by proper network management.
Jitter is the result of variations in the delay during period of the phone call. This is countered by using buffering to store the VoIP packets at the destination prior to then being heard.
For more information about how your business can benefit from VoIP contact sales@ewcoms or call +44 (0)20 3135 0145. or +61 (0)280 147185. To understand more about your hardware requirement go to VoIP Hardware
Remember, VoIP is not a simple substitute for a traditional telephone line. Emergency calls cannot determine your location and if there is a power failure you may be unable to make a telephone call.
EWC VoIP Solutions for your business