Development #1: All New Phone Systems Are VoIP

A closer look at the first item in our list of eight major recent developments and trends in telecom

Telecommunications technology is constantly changing and improving – seemingly faster and faster every year – and at Teracom, we keep our training courses up to date to reflect these changes. In the last post, we identified eight major developments and trends in telecommunications incorporated in our training.

In this post, we take a closer look at the first one: “All New Phone Systems Are VoIP”:

Basic Principle of Operation
The voice entering the microphone is digitized in the near-end phone. Typically 20 ms of digitized voice is packaged in an IP packet, which is carried in an Ethernet MAC frame on copper and fiber LAN cables to the far-end phone. There, the digitized voice is extracted from the packet and used to re-create the voice coming out of the speaker at the far end.

There are, of course, many details not mentioned, including the digitization method, called a codec, the Real-Time Transport Protocol (RTP) that adds timing information, the User Datagram Protocol (UDP) that adds error control and indicates the port number on the far-end phone, and how the bits are represented on copper and fiber LAN cables, to mention a few.

SIP and Softswitches
In a traditional phone system, voice travels on a dedicated circuit to a telephone switch, which physically transfers it to a different circuit to get it to the far end. Not so with VoIP! The near-end VoIP telephone creates a packet addressed to the far-end telephone, then the packet travels over LAN cables and through routers, interspersed with many other packets, to the far-end telephone. The VoIP packet does not pass through a telephone switch. The two VoIP phones exchange packets directly.

So a question is: how does the near-end telephone know what the far-end telephone’s IP address is? This is accomplished with the Session Initiation Protocol (SIP), which implements servers allowing the calling party to find out the IP address of the called party – if the called party wants to accept the call… a privacy shield to prevent Spam over Internet Telephony (SPIT). The servers implementing SIP are called softswitches. They are call setup assistants, and drop out of the picture once the call is established. The phones communicate packets directly.

SIP Trunking
What happens if the two telephones are in different cities? How does the packet move from the near-end VoIP phone to the far-end VoIP phone? One method is to use a gateway to convert the VoIP to an old-fashioned phone call and carry it over PBX trunks and/or telephone company trunks to the far end, where a gateway converts it back to VoIP… but this loses out on the voice-data-video integration synergy of IP communications. Another method is to carry the VoIP packet over the Internet… but there are no quality guarantees on the Internet. A third choice is to pay a carrier to move the VoIP packet from one building to another, as an IP packet, with guaranteed quality. This is called SIP trunking. It should be called VoIP trunking.

That is a thumbnail sketch of VoIP. If you would like to learn more, this is covered in the following Teracom training:

Course 101 Telecom, Datacom and Networking for Non-Engineers
(about an hour out of three days in-class)

Course 130 Understanding Voice over IP
(two days in-class)

The VoIP DVD-Video courses
(3 DVDs, six hours)

Cheers!

“No longer Greek to me! After taking your course, I sat in on a round table at a conference yesterday where VoIP was discussed by Time Warner Cable and Vonage – and I understood most of their diagrams and explanations – something that would have been Greek to me two weeks ago. Thank you!” — Bob Sabin, Tel Control, Inc.