Wednesday, December 19, 2012

High Reliability VoIP Phone Service Providers

It is pretty clear now that the days of analog wireline and even ISDN PRI connections are numbered. Clouds and computer networks offer more flexibility, a richer feature set and lower cost of ownership for voice communications. If you want to start enjoying these benefits, you need to embrace packet based phone service. That’s another way of saying VoIP.

Check out the offerings from high reliability VoIP phone service providers...The range of emotions on VoIP as a telephone service replacement varies from “wouldn’t have anything else” down to “never will I put that in my business.” Why such a diversity of opinions? It all comes down to your experience with the technology and when you got into it.

Today’s top VoIP providers for business are a far cry from the early days when VoIP was looked at strictly as a way to avoid long distance charges. Sure, putting a CODEC (coder/decoder) on your broadband service converts an analog phone into a IP phone. The conversion process turns a telephone set into a computer peripheral so that it can share the network with PCs, printers, routers and the like. Unfortunately, just getting a telephone to talk over the Internet leaves a lot to be desired when it comes to call quality.

What’s wrecking the voice quality? Most computer equipment on the network was designed with the TCP/IP protocol in mind. This protocol is perfect for the Internet in that it expects errors to occur and patiently resends lost data until there is a perfect set at the far end. You can’t predict exactly how long it will take to transmit a file and you have no idea how it got from point A to point B, but who cares? Your document, photo or database is exactly duplicated at the far end.

VoIP is sensitive to anomalies in network performance that aren’t even noticeable when uploading or download data files. That’s because VoIP is a real-time protocol. When you are talking, there is a steady stream of audio that must be converted into packets, sent through the network, and reconverted back into audio at exact time it is generated.

Consider what happens if there is even a half-second delay through the network. That small amount may not even be noticed on file transfers. With a VoIP phone call, that half-second is an eternity. The delay in itself doesn’t affect voice quality, but it does stymie conversations. You find that you and the other party can’t talk at the same time. You have to let the other person talk, then pause a half-second, and then start talking yourself. Essentially, the phone has been turned into a two-way radio.

That’s not the worst of it. If packets are lost in file transfers, TCP/IP will detect that the data wasn’t received properly and resend it. You can’t do that on a VoIP call. If you did, your words would arrive out of order and make no sense at all. So, the real time protocol simply ignores missing packets and keep on going. The effect of lost packets is missing information in the voice stream that shows up as voice distortion. It may be slight or the whole conversation may sound garbled.

Sometimes the packets are preserved just fine, but they take different paths from sender to receiver. This causes a variation in transport time called jitter. You can compensate for this somewhat by buffering the incoming packet stream to smooth it out. More than a little of this will cause the latency issue of needing long pauses before the other party can talk. Some jitter can cause packets to arrive out of order which distorts the voice as bad or worse than dropping the late arrivers completely.

The fact is that voice calls are fragile compared to one-way communications like file transfers or even streaming video. What does it take to ensure high reliability and high quality VoIP telephony?

High performance VoIP service providers avoid many potential issues by keeping the calls on their own networks and avoiding the Internet completely. These networks are often called SIP trunks after the Session Initiation Protocol that is at the heart of VoIP signaling. Essentially, the network operator becomes your phone service provider and uses a dedicated network connection to connect from the phones in your business to the switching equipment at their end. Since they own the network, they can carefully control it for latency, jitter and packet loss.

The type of CODEC you use also affects voice quality. A wideband CODEC can sound as good as any other phone call. There are even HD or HIgh Definition CODECs that expand the voice range so that the audio is more natural sounding and easier to understand. If you elect to go with a narrow band CODEC to cram more calls on the same SIP trunk, you may find that voice quality is marginal to unacceptable. It all depends on what your and your customer’s expectations are when it comes to call quality.

Are you interested in the benefits of VoIP solutions to replace your aging analog phones or in-house PBX system, but worried about degrading the quality of your calls? If so, learn more about the features and benefits offered by high reliability VoIP phone service providers.

Click to check pricing and features or get support from a Telarus product specialist.

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