Data networks do not use circuit switching. Your Internet connection would be a lot slower if it maintained a constant connection to the Web page you were viewing at any given time. Instead, data networks simply send and retrieve data as you need it. And, instead of routing the data over a dedicated line, the data packets flow through a chaotic network along thousands of possible paths. This is called packet switching.
While circuit switching keeps the connection open and constant, packet switching opens a brief connection -- just long enough to send a small chunk of data, called a packet , from one system to another.
It works like this:. Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers, as well. VoIP technology uses the Internet's packet-switching capabilities to provide phone service. VoIP has several advantages over circuit switching. For example, packet switching allows several telephone calls to occupy the amount of space occupied by only one in a circuit-switched network.
Using PSTN, that minute phone call we talked about earlier consumed 10 full minutes of transmission time at a cost of Kbps. With VoIP, that same call may have occupied only 3. Based on this simple estimate, another three or four calls could easily fit into the space used by a single call under the conventional system. And this example doesn't even factor in the use of data compression , which further reduces the size of each call.
Let's say that you and your friend both have service through a VoIP provider. You both have your analog phones hooked up to the service-provided ATAs. Let's take another look at that typical telephone call, but this time using VoIP over a packet-switched network:. Probably one of the most compelling advantages of packet switching is that data networks already understand the technology. By migrating to this technology, telephone networks immediately gain the ability to communicate the way computers do.
It will still be at least a decade before communications companies can make the full switch over to VoIP. As with all emerging technologies, there are certain hurdles that have to be overcome. We'll look at those in the next section. The current Public Switched Telephone Network is a robust and fairly bulletproof system for delivering phone calls.
Phones just work, and we've all come to depend on that. On the other hand, computers, e-mail and other related devices are still kind of flaky. Let's face it -- few people really panic when their e-mail goes down for 30 minutes. It's expected from time to time. On the other hand, a half hour of no dial tone can easily send people into a panic. So what the PSTN may lack in efficiency it more than makes up for in reliability. But the network that makes up the Internet is far more complex and therefore functions within a far greater margin of error.
What this all adds up to is one of the major flaws in VoIP: reliability. One of the hurdles that was overcome some time ago was the conversion of the analog audio signal your phone receives into packets of data. How it is that analog audio is turned into packets for VoIP transmission? The answer is codecs. A codec, which stands for coder-decoder , converts an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay.
It's the essence of VoIP. Codecs accomplish the conversion by sampling the audio signal several thousand times per second. For instance, a G. It converts each tiny sample into digitized data and compresses it for transmission. When the 64, samples are reassembled, the pieces of audio missing between each sample are so small that to the human ear, it sounds like one continuous second of audio signal.
There are different sampling rates in VoIP depending on the codec being used:. Codecs use advanced algorithms to help sample, sort, compress and packetize audio data. The codec works with the algorithm to convert and sort everything out, but it's not any good without knowing where to send the data.
In VoIP, that task is handled by soft switches. This is the numbering system that phone networks use to know where to route a call based on the dialed numbers. A phone number is like an address:. The switches use "" to route the phone call to the area code's region. The "" prefix sends the call to a central office, and the network routes the call using the last four digits, which are associated with a specific location.
Based on that system, no matter where you're in the world, the number combination " " always puts you in the same central office, which has a switch that knows which phone is associated with " They look for IP addresses, which look like this:. IP addresses correspond to a particular device on the network like a computer, a router, a switch, a gateway or a telephone. However, IP addresses are not always static.
They're assigned by a DHCP server on the network and change with each new connection. This mapping process is handled by a central call processor running a soft switch. Think of the user and the phone or computer as one package -- man and machine. That package is called the endpoint. The soft switch connects endpoints.
The soft switch contains a database of users and phone numbers. If it doesn't have the information it needs, it hands off the request downstream to other soft switches until it finds one that can answer the request. Once it finds the user, it locates the current IP address of the device associated with that user in a similar series of requests. It sends back all the relevant information to the softphone or IP phone, allowing the exchange of data between the two endpoints.
Soft switches work in tandem with network devices to make VoIP possible. For all these devices to work together, they must communicate in the same way. This communication is one of the most important aspects that will have to be refined for VoIP to take off.
As we've seen, on each end of a VoIP call we can have any combination of an analog, soft or IP phone as acting as a user interface, ATAs or client software working with a codec to handle the digital-to-analog conversion, and soft switches mapping the calls. How do you get all of these completely different pieces of hardware and software to communicate efficiently to pull all of this off?
The answer is protocols. There are several protocols currently used for VoIP. These protocols define ways in which devices like codecs connect to each other and to the network using VoIP. They also include specifications for audio codecs. The most widely used protocol is H. It provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as VoIP. Actually a suite of protocols, H. As you can see, H. That's what allows it to be used for so many applications.
The problem with H. An alternative to H. Smaller and more efficient than H. MGCP is geared toward features like call waiting. You can learn more about the architecture of these protocols at Protocols. One of the challenges facing the worldwide use of VoIP is that these three protocols are not always compatible. VoIP calls going between several networks may run into a snag if they hit conflicting protocols.
Since VoIP is a relatively new technology, this compatibility issue will continue to be a problem until a governing body creates a standard universal protocol for VoIP. VoIP is a vast improvement over the current phone system in efficiency, cost and flexibility. Like any emerging technology, VoIP has some challenges to overcome, but it's clear that developers will keep refining this technology until it eventually replaces the current phone system.
The central call processor is a piece of hardware that runs the soft switch. VoIP has its distinct advantages and disadvantages. The greatest advantage of VoIP is price and the greatest disadvantage is call quality.
For businesses who deploy VoIP phone networks -- particularly those who operate busy call centers customer service, tech support, telemarketing, et cetera -- call quality issues are both inevitable and unacceptable. To analyze and fix call quality issues, most of these businesses use a technique called VoIP call monitoring. VoIP call monitoring, also known as quality monitoring QM , uses hardware and software solutions to test, analyze and rate the overall quality of calls made over a VoIP phone network [source: ManageEngine ].
Call monitoring is a key component of a business's overall quality of service QoS plan. Call monitoring hardware and software uses various mathematical algorithms to measure the quality of a VoIP call and generate a score. The most common score is called the mean opinion score MOS.
The MOS is measured on a scale of one to five, although 4. An MOS of 3. To come up with the MOS, call monitoring hardware and software analyzes several different call quality parameters, the most common being:.
There are two different types of call monitoring: active and passive. Active or subjective call monitoring happens before a company deploys its VoIP network. Active monitoring is often done by equipment manufacturers and network specialists who use a company's VoIP network exclusively for testing purposes [source: VoIP Troubleshooter.
Active testing can't occur once a VoIP network is deployed and employees are already using the system. Passive call monitoring can detect network traffic problems, buffer overloads and other glitches that network administrators can fix in network down time. Another method for call monitoring is recording VoIP phone calls for later analysis. This type of analysis is limited, however, to what can be heard during the call, not what's happening on the actual network.
Looking for a solution from a Cisco partner? Connect with our partner ecosystem. Skip to content Skip to search Skip to footer. Watch video Contact Cisco. Get a call from Sales. Refresh your IP Phones. Watch the video Coming soon - Webex Desk Hub Hotdesking for a new way of working. Watch video.
IP phones with the latest technology to meet a range of needs. Zoiper works on all popular platforms! Improve the efficiency of your communications Clickdial CRM integration, number recognition, Outlook and Thunderbird plugins.
Small footprint Zoiper does not rely on Java, Flash or. Finding all this VoIP stuff a tad complicated? Desktop versions: Pick any of our Skins that suits your needs the most, or change an existing one or make your own.
Use remote provisioning and installer service to automate the deployment of preconfigured phones. Lock down some of the functionality to reduce the internal support costs. Mobile versions: Use our built-in QR code scanner to provide your employees, collegues or customers with a fool proof way to configure our softphone on iOS or android.
All they need to do is scan a QR Code that you generated on our website and they will be ready to make their first calls within seconds. Free or cheap calls with any VoIP provider! Combine multiple providers for the cheapest route to every destination. Want to use Zoiper in your company or call center? Follow us Twitter Facebook.
0コメント