Ozeki VoIP SIP SDK dramatically accelerates softphone development saving time
and costs in this way. However, before jumping into the middle of softphone
development, let's elaborate some basic terms related to this topic.
Softphone is the short term for software phone that is originally the software
model of a physical telephone device. However, in a lot of aspects it
extends the traditional telephoning functionality with some other useful features like
video phoning or conference calling.
The softphone is a program that is capable for Voice over IP communication with other softphone
solutions or even with physical VoIP devices or traditional telephone sets. The communication type is limited to the
used communication equipments.
Ozeki VoIP SDK provides magnificient technology for softphone application development.
You can create
a simple solution for voice calls, an extended softphone for
video calls or even an advanced VoIP application for
In the next sessions, you will get to know all the terms related to softphone technology
and the proper definition of them. First of all, you need to know the two basic protocols.
Voice over IP protocol
Voice over Internet Protocol is a standard protocol that was originally defined
for voice communications over the Internet. Today this protocol is extended for
video communication and data transmission purposes, too. The protocol definition is set
in some RFC (Request for Comments) documents that specify all the methods to use in the
The steps involved in originating a VoIP telephone call are signaling and media
channel setup, digitization of the analog voice signal, encoding, making packets,
and transmission as Internet Protocol (IP) packets over a packet-switched network.
On the receiver side, similar steps (usually in the reverse order) such as reception
of the IP packets, decoding of the packets and digital-to-analog conversion reproduce the original voice stream.
Ozeki VoIP SDK makes your work more efficient by providing full VoIP support for your applictions.
If you want to know more about VoIP technology, you can read the related manual pages on our website.
Session Initiation Protocol
Voice over IP communication often uses the SIP protocol for establishing communication lines between the
communicating parties. The main goal of this protocol is to create, modify and terminate
communication sessions in between the end points.
Ozeki VoIP SDK saves you a great deal of effort by providig a full SIP protocol implementation.
If you want to know more about the SIP protocol, please check the related manual site.
If you want to know how you can use the SIP backgroun for a VoIP softphone application, you
can start with an easy to learn softphone solution.
The next sections contain the basic definitions for softphone-related terms.
The phone line or channel provides the infrastructure for the data that is transferred over the
Internet. This term is the model for the physical telephone cables in the
traditional telephoning systems.
The voice over IP communication is made of voice digitization, data transmission and
decoding the audio data into analog voice on the other side. In this model all the
data travels through the phone line as simple Internet packets.
Once the connection between two end-points has made, the communication is
created for the two contacts. The softphones grant the possibility of more
than one separate communication lines at the same time. You can communicate
using multiple media types, such as Voice communication or Video messages.
Ozeki VoIP SDK provides revolutionary technology for even multiple phone line handling in
your VoIP applications. If you want to know how easy you can use this extraordinary feature,
check the developers guide article on this topic.
The communication between two end points, regardless of technology background, is
made through phone calls. A call is made through an
established phone line or channel and basically means the actual data transmission and
voice (and video) encoding and decoding processes.
The phone call starts with the dialing process as in case of traditional telephony, then
the phone line is established through the SIP protocol and when both parties accepted
the line establishment, they can start communicating. The call ends when one of the
end points ends it.
Making an audio call needs some background support from the software and hardware
solutions in use. The speech of the communicating parties is analog voice that needs to be encoded into
digital audio data that can be transmitted through the phone line. When the audio data arrives to
the remote communication party, it needs to be decoded back to analog voice information that can be played
through a speaker or a telephone set. This coding and decoding process is done
with the use of the so-called codecs.
The codecs are used for data coding and decoding, data encryption, conversion or
suppression, too. These processes are essential for VoIP communication and they also
affect the quality of the communication. VoIP communication originally requires the use of the same
codecs on both communication end points, however, in some special cases the PBX can translate
between codecs during the communication.
Ozeki VoIP SDK provides an extraordinary support for a great number of audio and video codecs.
You can check the full list and features on the supported codecs page.
If you want to know how you can use the codec support and how you can change codecs in your
VoIP application, check the developers guide manual page on this topic.
The Private Branch Exchange (PBX) is a software or hardware equipment that is responsible for
establishing the phone line between the communicating parties. The communication software and
hardware devices register to the PBX with a SIP account and the SIP protocol is also used for
After the communication end points are registered, they can dial each other's
number (SIP account). When the line is established between them, the PBX exits the communication
and they switch to peer to peer connection. The PBX will have a role again at the end of the call
when the session between the two end points needs to be closed.
The Voice over IP communication programs are useful when you are not available for actual
communication. The softphone solutions can be easily extended with audio and video message
sending and receiving functionalities that are not the part of traditional telephone sets.
The audio and video messages are handled similarly to simple communication.
The phone line is established between the end points and the data transmission works in
the same way. The only difference is that the communication is not done in real time.