CIRCUIT SWITCH TECHNOLOGY (PSTN)
Historically, telecommunications 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.
The problem with circuit-switched technology is that it requires a significant amount of bandwidth dedicated to each call, and it can only support certain types of calls (i.e. telephone to telephone). Moreover, the hardware needed to run circuit-switched networks is very expensive due, in large part, to the fact that voice and data services must be carried by different wires and thus need separate hardware to accommodate the two types of traffic. The higher cost of the hardware has caused many traditional telephone companies to resort to 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 those savings.
As the name implies, VoIP refers to calls that traverse networks using Internet Protocol (IP). This may mean that the calls are going over the Internet, or it may simply mean that calls are traveling over privately managed data networks that are using IP to transport the calls from one location to the other.
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 comparable and often superior. VoIP services and features have also evolved and allow customers many new and exciting options, such as online account management, three-way calling, call forwarding, and extra/virtual numbers.
VoIP 101 (DETAILED TECHNICAL OVERVIEW):
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, digitalizing voice in data packets, sending them, and reconverting them into voice 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/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 PSTN-Public Switched Telephone Network (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 that 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.
The Steps in VoIP communication:
- The ADC converts analog voice to digital signals (also known as bits)
- The bits are compressed into a format for transmission. There are a number of protocols, SIP being the most common for VoIP.
- The voice packets are compressed even further into data packets using a real-time protocol (typically RTP over UDP over IP).
- A signaling protocol calls the users: ITU-T H323 is the standard signaling protocol.
- 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)
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.
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.
Here are some methods to improve VoIP transmission:
- TOS field in IP protocol to describe type of service: high values indicate low urgency while more and more lower values denote more and more real-time urgency
- Queuing packets methods:
FIFO (First in First Out), the less intelligent method that allows passing packets in arrival order.
WFQ (Weighted Fair Queuing), means 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.
CQ (Custom Queuing), user can define priority.
PQ (Priority Queuing), a number (typically 4) of queues with a priority levels for each one: packets in the first queue are sent first, then (when first queue is empty) starts sending from the second one and so on.
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.
Shaping capability allowing limitation of the source to a fixed bandwidth for:
- Congestion avoidance, such as RED (Random Early Detection).