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VoIP Quality Of Service Information

VoIP works by transferring voice data over high speed Internet using packet switching technology. The voice data is converted to a digital format and transferred to the person you are calling. Voice data must be transferred on a Real Time Network, and this may not always be available. Erratic quality of service is often the result of using general purpose networks for the Internet to transfer calls. Transferring data over a fixed bandwidth can result in jitter or voice echo, delays, nonlinear distortion and packet losses. But let’s face it all you care about is the quality of the phone call. There are ways to ensure you choose the VoIP with the best quality of service, but first lets get a better understanding of how the quality of service is controlled.

Although, call quality is a relative matter, it may sound fine to me, but you may think it sounds like your talking in a tin can,  there are ways that VoIP providers are able to ensure the best quality result is possible. Electronic monitoring of the network will produce consistently clearer calls and better quality of service. Many VoIP’s are switching from the old RTP, real time transport protocol to an improved MPLS or multi-protocol label switching. This is improving the clarity of calls and the quality of service. Packet schedulers, similar to ATM, are also improving the quality of service.

Layer 2/3/4 switches and routers provide modern techniques for managing the VoIP’s traffic, although fairly effective at increasing quality of service,  these systems can be very difficult for network administrators to manage. At this point replacing this system is not the goal, since it provides the quality of service required. Rather, utilizing the system as it is, while improving the signaling protocols and the management of traffic is the course of action for now.

Still in the testing phase, a new technology developed by Cortec Systems in Australia is holding the promise of increase quality of service. The “freeway, as it is called” will allocate areas of a network at the beginning of a call to ensure quality of service, by reducing the threat of a clogged network. It accomplishes this by determining the requirements of the call before it takes place, and then interacting with the IP-PBX to ensure a quality call is initiated. This technology is not expected to be available for quite some time. However, VoIP’s quality of service can still be respectable, with today’s technology.

So now you know the basics and more, how do you determine your VoIP providers' quality of service? Ask the right questions:

1. Who are they peered with? VoIP is sensitive to latency, if they are not peered with the big 3 Savvis, UUNet, Level 3 then odds are the latency will suffer.
2. Are they running QOS on their routers and switches?
3. Do they have E-911?
4. Do they have a pre-install test phase?
5. Are they willing to run your company VoIP over DSL?
If they answer yes, you can stop asking questions and move to the next provider.

Answers;
1. They should peer or have a direct connection with two of the big three.
2. QOS is a must, if they don't have it move on.
3. E-911 is very important see E-911.
4. If they do not have a pre-install test they are obviously not concerned about the quality of service.
5. DSL latency is to unpredictable for business voice

Don't have faith in the usual suspects. Some of the bigger name companies in the business, you’ve seen their names all over the internet and possibly on TV, will not always be a good bet. Traditional TDM voice services are provided at Layer 2, VoIP is Layer 3. The two worlds couldn't be farther apart.
 


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