How Does Voice Over Ip Work; Why Use Voip; What Is The Relationship Between Codec And Voip; What Advantage Does Voice Over Ip Provide - ZyXEL Communications P-2608HWL-D1 Support Notes

P-2608HWL Series Support Notes

How does Voice over IP work?

In VoIP, voice data is sent digitally in discrete packets through the Internet, not through the traditional circuit
switch of PSTN. To do so, an analog-to-digital converter is required at sender side to translate voice (analog
signal) to digital signal before transmission. At the receiver end, an analog-to-digital converter converts the
digital signal back to analog so the voice can be heard on the phone.

Why use VoIP?

Traditionally voice data is transmitted using circuit switching. Since circuit switching is designed to carry
voice, it does it very well. However, as broadband networks become a mainstream for network access and
technologies have evolved, we don＇t want to confine ourselves to just using text-based applications (such as
e-mail, instant messaging, etc.) for communication over the Internet. Thus, the convenience of voice
communication through the Internet has quickly become popular.
In addition, it would take a much longer time, more effort and money to implement new features using circuit
switching. Since the IP technology is a standard and various applications are available, it is easier and more
cost-effective to integrate new services and applications using IP.

What is the relationship between codec and VoIP?

In order to send voice (analog signal) over IP, it first needs to be digitized. Codec is a technique used to
digitize analog signals into digital signals and vice versa. There are various speech codecs available VoIP.
Each codec has its advantages and disadvantages.

What advantage does Voice over IP provide?

VoIP provides advanced integration of text, video and voice in emails. This cannot be done using traditional
circuit switching (PSTN).

What is the difference between H.323 and SIP?

H.323 and SIP are proposed by different groups. Session Initiation Protocol (SIP) is a standard introduced by
the Internet Engineering Task Force in 1999 to carry voice over IP. Since it was created by the IETF, it
approaches voice and multimedia from the Internet, or IP. Whereas H.323 emerged around 1996, and as an
International Telecommunication Union standard, it was designed from a telecommunications perspective. Both
standards have the same objective - to enable voice and multimedia convergence with IP protocols.
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