This article is a brief guide about the tools and devices that are needed for
softphone usage and communication. After reading through this page you will be fully
familiar with all the essential terms concerning the devices and tools used with softphones.
A VoIP softphone application is basically used for telephoning functions over the Internet
or on a LAN (Local Area Network). This functionality essentially covers voice calls.
The following sections in this article will introduce you the basic tools (both physical and
logical) that are used in a softphone.
The Session Initiation Protocol (SIP) is one of the basic standards for establishing connection
between communicating tools (software and/or hardware) in the VoIP world. The SIP protocol
uses the so-called SIP accounts that are used for identifying the end points in the
VoIP communication. This account is used for registering the softphones on the VoIP
communication network and for initializing the phone line between the
SIP, VoIP protocol support
In order to have a softphone that can communicate on a VoIP network you need to have
VoIP and SIP support implemented in you software phone
application. Initially these protocols are defined in RFC documents that
give a detailed guide for all the tools and processes need to be used.
The Private Branch Exchange (PBX) is the hardware or software tool that is used for
handling the SIP account, registering the VoIP communication tools and
devices, and establishing the phone line between them.
The PBX works in a VoIP communication network as a phone center in the ordinary
telephoning world. It is also capable for connection the VoIP and the PSTN (public switched telephone
network) together through a simple adapter that works as a physical phone system from the
PSTN network and as a VoIP end point from the VoIP network. The adapter also has
one or more SIP accounts registered to the PBX.
The voice communication needs some basic devices installed in your computer. In case of traditional
telephones, these devices are built in the telephone set, while in case of IT
communication they can be separate devices.
The speaker is used for playing the voice that comes from the remote communication party.
It can be a traditional speaker but even a headset (either with or without a microphone).
The microphone is the physical device that captures the voice from its environment and
transmits it to a program that creates digital audio data from it. The microphone can be
built in the computer (especially in laptops or mobile devices). It can be a separate
microphone or the part of a headset, as well. The actual microphone solution does not affect
the communication process but affects the quality of the voice communication.
Reliable Internet Connection
Audio and rather video data transmission requires fast and reliable Internet connection.
The compression and coding rate of the codecs are good for decreasing the number and the
size of the packets that need to be sent over the Internet. It is recommended
to have an Internet connection of high performance and bandwidth for the smooth communication.
Audio (and video) codec support
The VoIP communication works as follows. You talk into the microphone, your
analog voice is digitized by the VoIP background software, it can also be suppressed for
the easier transmission, then the digitized audio data is transferred through the
network (LAN, WAN, Internet) and on the other side, the digital audio data is
transformed back to analog voice that is played through the speaker.
The voice digitization, suppression and the transformation of the digital audio data
into analog voice again are done with the use of the so-called audio codecs. These codecs are
defined in certain RFC documents. They define the processes that need to be implemented
in the VoIP solutions for the proper communication. This process is done in case
of video communication, too.
When two VoIP solutions try to communicate, they need to use the same codecs for
audio data handling. This is essential because decoding on the remote side needs to be
done with the same algorithm as coding on the other side. This process is essentially done
in the softphone, but in some special cases it can be done in the PBX, however this is not
the default solution, as the softphones are built basically for peer to peer communication.
This is essential as the PBX is not overloaded by the communication itself.
The terms and devices that were introduced above are the basics for softphone communication.
If you have these basic tools, you will be able to communicate in the VoIP network and
with a PSTN adapter installed; you can also communicate with the wired and the
mobile telephone network too.
This article introduced you the basic knowledge about the tools and devices that are used with
software telephones. If you have read through this page carefully, you already have
all the information you need to start on your own solution.
As you are now familiar with all the terms concerning this topic, now it is time to take a step further
and explore what other extraordinary solution Ozeki VoIP SIP SDK can provide for you.