Packet Voice Moves to the Cloud

Many businesses and individuals have already switched to VoIP for their telephone service. The original impetus is a cost savings over legacy phone service. Beyond that, enterprise VoIP services offer the opportunity for reduced staff and maintenance coupled with a richer feature set. Just when you thought that VoIP was a settled technology, a new wrinkle comes along. That’s VoIP up in the cloud.

Why move your phone service to a cloud? Let’s take a look at why this might be advantageous to your company and how we got from the old-timey candlestick and desk sets to something as amorphous as a cloud.

A generic technical term for VoIP technology is packet voice. Packets are a computer network protocol most often associated with Ethernet. A packet is a self-contained piece of data that has all the information needed to get it from source to destination. Packets are packets. It doesn’t matter what’s inside. Packets can carry voice, images, video, data and whatever else you can reduce to binary digits.

The major technical leap in packet voice is changing from a switched circuit telephone network to a packet switched network. Switched circuit means that every telephone call has an electrical circuit set up to connect the two (or more) phone sets involved in the conversation. This is unique to the public telephone system and has been that way since its inception. All telephone activity was originally voice calls made between electrically passive telephone sets. Look at one of those old phones. Do you see an AC cord? No way. The electrical power for the call was supplied by batteries at the telephone switching center.

When computer networks became more prevalent than telephone networks, the idea of what a network is changed dramatically. Both telephone and computer networks can run over twisted pair copper wire equally well. What’s different is that computer networks don’t set up and tear down electrical circuits every time that one piece of equipment wants to exchange data with another. The circuits that connect each computer device are continuous. What is switched is the data packets themselves. Ethernet switches or routers ensure that packets get to their intended destinations without having to create new circuit connections on the fly just for a single or group of packets.

Here’s why that’s important. VoIP or packet voice is transported over computer networks rather than switched circuit telephone networks that are only used for telephones. By connecting the phone to the LAN, you can eliminate the separate telephone network. All that wiring and the personnel needed to maintain it and change connections as phones are moved, added or deleted goes away. The cost of running the telephone network also goes away.

Something much larger doesn’t go away. That’s the switching system needed to interconnect telephones, even those using packets, and connect to the Public Switched Telephone Network (PSTN) for outside lines. That specialized switching system is most often called a Public Branch Exchange or PBX. Small businesses with few phones and outside lines may use a simpler system called KTS or Key Telephone System. Those are the phones with individual line lights that tell you if a particular phone line is being used or available for your call. Medium and larger companies have the more automated PBX equipment that manages both internal calls and multiple outside lines.

This is where the cloud comes in. An in-house PBX system can be set up to work with legacy analog phones (switched circuit) or the newer IP telephones used for VoIP. You can connect the phones and PBX to your computer network to eliminate the separate telephone wiring, but you still have to make connections to the public phone network if you want to take or make outside calls. PBX systems aren’t cheap, either. They require a major investment, ongoing maintenance efforts and the likelihood that you’ll have to replace the one you have when it goes obsolete or runs out of capacity.

Why not move that PBX to the cloud the same way that you move servers from in-house to the cloud? That’s the idea behind Hosted PBX, also called Hosted VoIP. The hosting is done by a very large packet voice switching system located at a cloud service provider. You don’t have to buy this system, maintain it or worry about upgrades and obsolescence. That’s the job of the cloud service provider. You simply pay a monthly fee for each phone or “seat” that you need.

Ironically, this concept takes us back to the early days of telephony when there was a large central switching system that connected to each telephone. The difference between the old “Ma Bell” approach and Hosted VoIP is that you are no longer tied to the single phone company that owns the phone wires. You can order network connections called SIP trunks to connect to any cloud provider you want. VoIP technology also allows new applications that integrate telephones and computers, important for call centers and contact centers.

Packet network technology has enabled computers to network within companies and all over the world. This technology is now being used to transmit video conferences and telephone calls on the Internet and private network connections. The rise of cloud computing companies takes packet switching equipment to the next level by moving the central hardware to the cloud, where it can be managed by specialized service providers. The advantage to you is lower costs and more features.

Are you wondering whether your business telephone system is optimum for your current requirements? Why not explore other options that include VoIP, SIP Trunking and Hosted PBX services? Get pricing and features for competing enterprise grade VoIP services to compare with your current setup.

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.

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.

Other Flavors of Packet Voice

When we think of telephone service these days it's in terms of two battling technologies. There's circuit switched telephony that has been the standard for over a hundred years versus VoIP, the emerging standard. But, actually, VoIP is just the best known example of technologies to transmit voice by packet over networks.

The real technology battle is the shift from switched circuit networks to packet networks. But that's been going on forever. Aren't we there yet? No, not really. But that's the direction we're headed until something else is invented.

Circuit switched networks have their advantages. Once you establish a path or circuit between two points, you have exclusive use of that connection until you are done with it. The advantage is that a continuous connection usually has little latency since there are no routing decisions to be made once the circuit is set up. The same is true for jitter, although signal quality is only as good as the connection quality. Analog has crosstalk issues and noisy lines can drop out portions of a digital bitstream.

The big disadvantage with circuit switched networks is that they tie up an expensive resource whether it's being used or not. That's the efficiency improvement that packet switched networks offer. By breaking up data into bite size (or is that byte size?) packages, routers can act as traffic controllers to keep the network busy but not overloaded.

Packet switched networks are a natural for data communications. The Internet protocol, TCP/IP handles breaking large data files into packets and reassembling them on the other end. It also ensures accurate delivery by forcing a retransmission of damaged packets. What VoIP does is mimic computer data by breaking the continuous voice signal into IP compatible packets and sending them down the same network with data file packets. But the TCP or Transmission Control Protocol that ensures accurate delivery doesn't work so well with VoIP. Since a conversation is continuous, retransmitting a broken packet and delivering it later would only cause confusing interference.

VoIP's popularity can be traced to the fact that IP is so popular and nearly ubiquitous in corporate LANs as well as the Internet. But beyond the internal corporate network, there are other types of packet switched networks for WAN or Wide Area Network transmission.

VoATM or Voice over ATM networks also breaks a continuous speech signal into packets. But ATM (Asynchronous Transfer Mode) breaks the signal into 53-byte packets known as cells. The small cells reduce the effect of jitter or variable time delay and one lost or damaged cell doesn't markedly degrade the quality of the voice stream.

VoFR or Voice over Frame Relay networks use frames that transmit large packets of data that vary in size but can be up to a 1,000 or more bytes in length. Frame Relay networks have been very popular for connecting multiple business locations together, but are giving way to the newer MPLS or Multi-Protocol Label Switching networks that provide the same service and more.

Both Frame Relay and ATM networks are private network technologies. They pre-date the massive popularity of the Internet and don't use Internet Protocol. But they do support voice, data and video using their own proprietary protocols.

What about MPLS networks? Being multi-protocol, as the name says, a private MPLS network can easily support VoIP traffic. Since this is also a private networking arrangement, the service quality variabilties of the public Internet are avoided. When VoIP is transmitted on an MPLS network it is known as VoIPoMPLS. But there's a way to directly implement voice on an MPLS network that is known as VoMPLS. This is a fairly new technology that is designed to work in on converged network carrying voice, video and data simultaneously over the same paths. MegaPath, a leading competitive carrier, has just announced a VoMPLS service for its network that combines manged voice, data and security for business-grade users.

Even wireless is not immune to the move from circuit to packet switching. Right now cellular services offer switched voice connections on one set of channels and packet switched data on another. But this may be changing with the emergence of WiMAX and LTE as wireless broadband services. A new forum has started up to develop VoLGA or Voice over LTE via General Access.

Is your company looking at options to reduce telephony costs or make better use of the network resources you already have? A reduced cost digital trunking service or some form of packet voice service may be the answer.