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VoIP Services 101

Want to know more about how VoIP works? You've come to the right place. Here we will explain the difference between a traditional phone system and a VoIP phone system, in addition to providing you with common technical VoIP terms, features, and processes.

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Circuit Switch Technology (PSTN)

Here's how a traditional phone system operates

Historically, telecommunication companies have relied on what is commonly referred to as ‘circuit-switched technology’ to transport telephone calls. This technology establishes a 'permanent' connection between the calling and the receiving parties for the entire duration of the call, therefore keeping anyone else from using the physical lines while the call is in process.

The problem with circuit-switched technology is that it requires a significant amount of bandwidth dedicated to each call. Furthermore, circuit-switched technology can only support certain types of calls (i.e. telephone-to-telephone). Moreover, the hardware needed to run circuit-switched networks is very expensive because voice and data services must be carried by different wires, and therefore, need separate hardware to accommodate the two types of traffic. The high cost of this hardware has caused many traditional telephone companies to start using parts of the Internet infrastructure to connect PSTN calls. You may have already placed or received a call using VoIP technology without even knowing it!

Naturally, the traditional telephone companies pass along the costs of building and maintaining a circuit-switched network to the consumer in the form of higher rates for their telephone services. Telecom companies may save some money by borrowing from Internet bandwidth, but if your call is placed on a regular telephone using PSTN hardware, you won't see these savings.

typical traditional PSTN phone
voice over internet protocol

Voice over Internet Protocol Technology

The more affordable and easier way to make calls

As the name implies, VoIP refers to calls that traverse networks using Internet Protocol (IP). This can mean that calls are traveling over the Internet, or it may mean that calls are traveling over privately managed data networks that use IP to transport calls from one location to another.

The voice stream is broken down into packets, compressed, and sent toward its final destination by various routes (as opposed to establishing a 'permanent' connection for the duration of the call), depending on the most efficient paths given network congestion.

At the receiving end, the packets are reassembled, decompressed, and converted back into a voice stream by various hardware and software elements. Whether the call originated on a PC, telephone, or an Integrated Access Device (IAD), and whether it is going to be terminated on a PC, telephone, or IAD, will determine the type of software and hardware needed to initiate and complete the call. Over the years, broadband phone providers have been working on improving and re-engineering the hardware and software used in VoIP calls. Today you can compare a VoIP provider’s voice-quality to the traditional circuit-switched technology and find that the VoIP provider’s quality is often superior. VoIP services and features have also evolved and allow customers many new and exciting options, such as online account management, conference calling, call forwarding, and virtual numbers.

infographic of a standard VoIP phone system

A Technical Overview of VoIP

Uncovering the technical side of VoIP

Many years ago it was discovered that sending a signal to a remote destination could be done in a digital fashion: before sending it we have to digitalize it with an ADC (analog to digital converter), transmit it, and at the end transform it again in analog format with DAC (digital to analog converter) to use it.

VoIP services work in just that manner, converting your voice into data packets, sending them, and then reassembling them into sound at their destination.

Digital format can be better controlled. We can compress, route, and convert it to a better format, and so on. Also, we saw that a digital signal is more noise tolerant than an analog signal.

TCP and IP networks are made of IP packets containing a header (to control communication) and a payload to transport data. VoIP uses the header to navigate the network to its destination. The payload carries bits of the conversation.

What are the advantages to using VoIP rather than the Public Switched Telephone Network (or more commonly, the phone company)?

When you are using a PSTN line, you typically pay a line manager company for the time used. The more time you talk, the more you'll pay. In addition, you will probably not have the option of speaking with more than one person at a time.

In contrast, VoIP Services allow you to talk as long as you would like with multiple people (other people may also need to be connected to the Internet) as far away as you want for free or for a fraction of the PSTN cost. You can also browse the Internet at the same time; sending images, graphs, and videos to the people you are talking with.

popular VoIP-related terms

The Basic Steps in VoIP Communication

What it takes to complete your call

On the back end of VoIP, here is what happens when you place a call over a VoIP connection:

  1. ADC converts analog voice to digital signals (also known as bits)
  2. The bits are compressed into a format for transmission. There are a number of protocols, SIP being the most common for VoIP.
  3. The voice packets are compressed even further into data packets using a real-time protocol (typically RTP over UDP over IP).
  4. A signaling protocol calls the users: ITU-T H323 is the standard signaling protocol.
  5. Upon arrival at the destination, the packets are disassembled, data is extracted, and converted analog voice signals are sent to the sound card (or phone).

All of the steps must occur in real-time to avoid waiting too long for a vocal answer! (See QoS section)

VoIP settings

The Tech Behind Your VoIP Call

Additional technical information on your VoIP connection

Analog to Digital Conversion: This process occurs inside computer hardware, typically an integrated card in your PC or an external telephone adaptor.

Today every sound card allows 16-bit conversion from a band of 22050 Hz (for sampling you need a freq of 44100 Hz according to the Nyquist Principle) obtaining a throughput of 2 bytes * 44100 (samples per second) = 88200 Bytes/s, 176.4 Kbytes/s for a stereo stream.

For VoIP, we needn't such a throughput (176kBytes/s) to send voice packets.

Compression Algorithms: Now that we have digital data it can be converted to a standard format that can be quickly transmitted.

PCM, Pulse Code Modulation, Standard ITU-T G.711

  • Voice bandwidth is 4 kHz, so sampling bandwidth has to be 8 kHz (for Nyquist).
  • We represent each sample with 8 bit (having 256 possible values).
  • Throughput is 8000 Hz *8 bit = 64 kbit/s, as a typical digital phone line.

In real application mu-law (North America) and a-law (Europe) variants are used which code analog signal in a logarithmic scale using 12 or 13 bits instead of 8 bits.

Quality of Service (QoS): VoIP applications require real-time data streaming to support an interactive data voice exchange.

Unfortunately, TCP/IP cannot guarantee this kind of purpose; it just makes a "best effort" to do so. So we need to introduce tricks and policies that can manage the packet flow in EVERY router we cross. If you subscribe to one of the broadband phone company providers their technical support can help you setup your router to optimize voice transmission. Technical support can be a distinguishing factor in determining which of the VoIP providers you choose.

VoipReview tools

Improving VoIP Transmission

Things to consider in order to get the most out of your VoIP service

  1. TOS field in IP protocol to describe the type of service: high values indicate low urgency while lower values denote more real-time urgency.
  2. Queuing packets methods:
    1. FIFO (First in First Out), the less intelligent method that allows passing packets in arrival order.
    2. WFQ (Weighted Fair Queuing), means the fair passing of packets (for example, FTP cannot consume all available bandwidth), depending on the type of data flow, typically one packet for UDP and one for TCP in a fair fashion.
    3. CQ (Custom Queuing), a user can define priority.
    4. PQ (Priority Queuing), a number (typically 4) of queues with priority levels for each one: packets in the first queue are sent first, then (when the first queue is empty) starts sending from the second one and so on.
    5. CB-WFQ (Class Based Weighted Fair Queuing), like WFQ but, in addition, we have class concepts (up to 64) and the bandwidth value associated for each one.
  3. Shaping capability allowing limitation of the source to a fixed bandwidth for:
    1. download
    2. upload
  4. Congestion avoidance, such as RED (Random Early Detection).
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