Telecom Terms and Conceptsby Sean Walberg
Voice over IP (VoIP) is all the rage these days, largely due to the proliferation of Internet Telephony Service Providers (ITSPs) such as Vonage, and the rising popularity of so-called soft PBXes such as Cisco Call Manager and the open source Asterisk. On the surface it seems easy to treat VoIP as another application on the network and just encapsulate voice signals into IP packets. However, coercing voice into data packets requires knowledge of how traditional voice services are provided and what overhead tasks are required to direct the call from one party to another.
Familiarity with these terms and concepts will also help consumers as they start investigating VoIP services and products. Not all ITSPs are the same, and understanding how your current service is provided will allow you to properly compare service offerings and avoid surprises down the road. Similarly, those looking to upgrade their current PBX can start to separate the marketing talk from the things that matter, simply by understanding how the phone system works.
The Public Switched Telephone Network (PSTN) is what most of us are familiar with. Subscribers are given a pair of copper wires (called a loop) connected from the demarcation point at their site to a phone switch at the Central Office (CO). On the user side, telephones are connected in parallel to the demarcation point. At the CO, this loop terminates in a card on a Class 5 phone switch. The Class 5 switches generally aggregate into one or more Class 4 switches which are responsible for marshaling traffic between all the Class 5 switches and the rest of the PSTN.
So far, the data guys in the crowd are thinking this looks quite similar to a data network, and they'd be right. The differences occur when people need to talk. In a data network, a subscriber wanting to talk to another subscriber would drop a packet on the wire with a destination address of the other party. The packet would flow, hop by hop, over the network until received at the other end. We call this packet switched because each packet is switched from the ingress interface of a device to an egress interface closer to the destination. Voice networks, however, are circuit switched, which means that a bidirectional connection has to be nailed up from the source to the destination before people can talk.
As I said before, the interface from you to the phone company is a copper loop. The port on the carrier side is termed Foreign Exchange Subscriber (FXS), and it provides dial tone and voltage to the remote device (your phone). Normally your phone is said to be on hook, which means the circuit between you and the FXS port is open from an electrical standpoint. When you pick up the phone and go off hook, the phone completes the circuit and allows current to flow. The CO provides you with the familiar dial tone and waits for you to enter digits. Each digit is a combination of two frequencies, which is called Dual Tone Multi Frequency (DTMF) signaling. It is further defined as in band, since the signals go over the same loop you use for talking. Contrast this to out of band signaling where the control connection would be over a different connection (we'll see examples of this later).
When the remote CO gets enough digits from you to determine who you're calling, it must build a connection to the other party. To do so, it sends a series of messages across the Signaling System 7 (SS7) network to build a dedicated voice circuit over the PSTN to the destination network. The remote CO rings the other party's phone, and when they go off hook, sends the voice traffic along. At this point both people can talk.
While the connection between the telephone and CO is an analog copper loop, the rest of the network is digital. Analog voice is sampled into digital signals for transit across the PSTN. Voice frequencies are generally under 4KHz (thousands of cycles per second), and the Nyquist theorem says we have to sample at twice this rate (8KHz) to get a good representation of the original signal. Choosing eight bits per sample, this means a voice stream consumes 64,000bits/S which is the fundamental building block of digital telephony. One such channel is called a DS0. The method used to convert the analog signal to a digital sample is called Pulse Code Modulation (PCM).
In order to maintain our sampling rate we have 125 microseconds (uS) between consecutive samples (1/8,000Hz), which is more than enough time to transmit the eight bits of information. Multiple samples can then be sent consecutively, each belonging to a different stream. This is called Time Division Multiplexing (TDM). Twenty-four such samples can be smashed together to give the next building block of digital telephony, called the DS1, otherwise known as a T1. Each DS1 frame consists of a single framing bit followed by 24 DS0 samples, for a total of 193 bits per frame. At 8,000 frames per second, we get the familiar 1.544Mbit/S for a T1. In Europe they use 32 DS0s and call it an E1.
From here we can cram more channels onto the wire as long as we respect the 125uS requirement. The important thing is that each conversation always appears at the same spot in each frame (also called a time slot). If there happens to be no conversation in a particular time slot, data must still be transferred! Likewise, if nobody is talking into their phone, the time slot is still used.
With all 192 data bits of the frame in use there's no spot for signaling between the two sides of the DS1. Robbed Bit Signaling (RBS) steals a bit from some of the DS0s, effectively reducing the sample size to seven bits or 56,000kbit/S. This is another example of in-band signaling. Alternatively, Integrated Digital Services Network (ISDN) protocols can be used to dedicate one of the voice channels to signaling and the other 23 to voice in a form known as Primary Rate Interface (PRI). The out-of-band signaling channel is called the D (data) channel, the voice channels are B (bearer) channels.
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