Description of VOIP technology, the different VoIP protocols

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.

As with most technologies there are a number of different standards for VoIP which are not compatible with each other. The original VoIP standard, H323, was designed by the International Telecommunications Union (ITU) As such it is a very well documented technical standard and is used extensively in the wholesale VoIP market. Although H323 was also the defacto standard for end users in the early days of VoIP it had drawbacks in such an environment. The most problematically issue at that time was the ability to deal with firewalls. Often VoIP calls ended up with one way conversation since the return element of the call could not transverse the firewall. Although there are now ways of overcoming this problem with H323 it gave rise to the emergence of an additional standard Session Initiation Protocol (SIP).

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.

More recently, the term SIP Trunks has been used by VoIP Providers to identify VoIP services that allow multiple inbound or outbound calls. SIP trunks use the SIP protocols that have now become the defacto standard for most VoIP services and hardware. A SIP trunk is in effect no different from any other SIP based VoIP service, it just denotes that the service is capable of having multiple concurrent inbound and outbound calls. All V4B VoIP numbers are SIP trunks , allowing our customers all the benefits of a SIP trunk but without the extra charges that many VoIP providers charge for this service.

Regardless of the technological protocol used, it is important to opt for an open source solution. his ensures that the widest range of hardware and services are available and gives the opportunity to make free calls between different VoIP Providers. For most business uses the optimum protocol is currently SIP. There are a vast number of different SIP hardware devices on the market allowing you business to achieve the maximum benefits from using VoIP

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
Packet Loss
Jitter
Echo

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.