Telephony VoIP IP Interview

Questions And Answers Guide.

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 Telephony VoIP IP Interview Questions And Answers

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Telephony VoIP IP Job Interview Preparation Guide.
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Question # 1
Explain the IPV4 and IPv6 address bit?
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Answer:-
yes IPV6 is advanced version of IPV4.
the address of IPV^ is 128 bits with extend able memory and
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that of IPV$ is 32 bits.
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Question # 2
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What is IP addresses range A,B,C,D,E?
Answer:-
0.0.0.0 126.255.255.255--class-A
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127.0.0.0 127.255.255.255--LoopBack IP address.
128.0.0.0 191.255.255.255--class-B
192.0.0.0 223.255.255.255--class-C
224.0.0.0 239.255.255.255--class-D
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240.0.0.0 255.255.255.255--class-E
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Question # 3
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Tell me voip test setup that you have for testing?
Answer:-
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Press and release the mute button , than press the
following number on dialpad 8378# ( TEST ) , This start
self test.
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Question # 4
Explain url?
Answer:-
URL stands for : Uniform Resource Locator.
To access information over internet using web browser URL
is used as a address to access particular web site.
URL is the convertion of IP to human understandable form.
Users give the name of website in the browser which is
later on converted to IP address by DNS server. And the
request page is send to users browser.
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Question # 5
Do you know in voip telephone witch part will convert data anolog to digital, digital to anolog?
Answer:-
A codec (Coder/Decoder) converts analog signals to a
digital bitstream, and another identical codec at the far
end of the communication converts the digital bitstream
back into an analog signal.
In the VoIP world, codec's are used to encode voice for
transmission across IP networks.
Codec's for VoIP use are also referred to as vocoders,

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 Telephony VoIP IP Interview Questions And Answers

for "voice encoders".

Codecs generally provide a compression capability to save
network bandwidth. Some codecs also support silence
suppression, where silence is not encoded or transmitted.
Regards
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Question # 6
Explain How does VOIP work?
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Answer:-
The basic principle of Voip is very simple. It's the same
technology you have probably used already to listen to
music over the Internet. Voice sounds are picked up by a
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microphone and digitized by the sound card. The sounds are
then converted to a compressed form, compact enough to be
sent in real time over the Internet, using a software
driver called a codec. The term codec is short
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for "encoder/decoder". The sounds are encoded at the
sending end, sent over the Internet and then decoded at the
receiving end, where they are played back over the
speakers. The only requirements are a connection between
the two computers of an adequate speed, and matching codecs
at each end.
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To be usable, a Voip system also needs a method for
establishing and managing a connection, for example,
calling the other computer, finding out if they accept the
call, and closing the connection when a user hangs up.
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Because Voip allows two way communication, and even
conference calls, it's a lot more complicated than simple
audio streaming. How calls are managed is the area in which
Voip systems fundamentally differ, and two Voip users must
be using the same system (or compatible ones) in order to
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be able to call each other.
Because most Internet users don't have a permanent Internet
address (IP address, a number like 212.44.88.17 that
uniquely identifies that computer, at that moment), Voip
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systems don't generally work by calling another computer
direct although that may be an option for those who do
have a permanent address. Instead, each user of the service
registers with an intermediate server, which maintains a
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record of their IP address all the time they are connected.
An example of a Voip application that works this way is
Picophone. The small size of the PicoPhone application file
(about 64Kb, barely larger than Windows Notepad)
demonstrates clearly that the basic principles of Voip are
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not complicated to implement.
Another reason for using an intermediate server is that it
eases the problem of getting Voip to work through the
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firewalls that everyone uses these days. Many firewalls
block any data from the Internet that is not sent in
response to a specific request. This makes it impossible to
call another computer direct. Because the called computer
did not request any data from the caller, the call request
would be blocked. By establishing a connection with a
server, the Voip software opens a channel of communication
through which other computers can call it. Communication
may continue using the server, or information may be passed
via the server that allows the two computers to open a
direct connection between them and continue using that.
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Question # 7
What do internet telephony, packet telephony, IP telephony and converged network means?
Answer:-
The first thing all mean the same thing. Which is using IP
(Internet protocol) for voice services. Some voice networks
are only packet-switched and have no access outside of
their own VoIP network. Most VoIP networks have a Gateway
that connects to a circuit-switched external network which
gives them acces to external calling. One of the gateways
responsibilites is to convert G.711 Circuit-switched media
(typically a T1 provided by a telco company) to the 7.723
Packet-switched media that will traverse the companies VoIP
network. A device called a gatekeeper will then convert the
IP address (used by H.323 protocol) to a standard telephone
number (E.164 address) that can be used for external
calling.
A converged network is a network that passes both Voice and

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 Telephony VoIP IP Interview Questions And Answers

Data over the same set of devices. Converged networks

generally implement QoS (Quality of service) on all actived
network devices to ensure the VoIP has priority over
standard data because of it's more rigid demands.
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Question # 8
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Explain VOIP?
Answer:-
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VOIP means voice over internet protocol. In which the voice is send through internet (or any IP network). For that the analog voice is converted into digital
data and use appropriate CODEC (for bandwidth saving) and send through IP
network. At the receiving end the digital data is converted back into analog
voice.
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Question # 9
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Explain What are the advantages to VoIP?
Answer:-
The big advantage is VoIP may save you money depending on
how much you are currently spending for local and long-
distance calls. What you will need to do is get the total
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cost the phone company is charging and compare it against a
VoIP plan that interests you. With most plans, you get free
calls within the U.S. and Canada for a low flat rate.
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International calls usually have very low rates with no
connection fees. For both residential customers and
businesses that make a lot of long distance and
international calls, the savings can be several hundred
dollars a year.
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Another advantage is with the features available with VoIP.
Features such as caller ID, call waiting, call forwarding,
3 way conferencing and voice mail are usually included at
no extra cost. With the phone company, these services are
usually extra.
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In addition, you can make free phone calls anywhere there
is a high speed Internet connection available. That means
you can be in another state or even in another country and
make calls as if you were back at your home or business.
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You will just need to bring your phone adapter along with
you and possibly a phone in case one is not available
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Question # 10
Tell me What equipment is needed for VoIP?
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Answer:-
The main requirement is a broadband Internet connection
such as DSL or cable. Any other equipment such as a
telephone adapter or microphone usually comes with the VoIP
service provider.
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Question # 11
Tell me How does VoIP work?
Answer:-
To understand how VoIP works, you will be taken through the
process of voice transmission from one end to the other.
The process starts with a person talking into the
mouthpiece on one end of a VoIP call.
This analog voice signal must first be sampled and
digitized. Voice sampling is usually done 8,000 times per
second (8KHz). In order to reduce bandwidth, a voice CODEC
is used. A voice CODEC is a compression/decompression
algorithm that is optimized for the voice frequency range.
The bit stream uncompressed is 64Kbps. By using an
available CODEC, the bit stream can be reduced to 8Kbps or
less.
In order for the compressed voice data to be sent over the
Internet, it must go through a process called
packetization. This is a packet consisting of a small
sample of the voice data (usually 10-30 milliseconds).
While being routed through the Internet, these packets can
get delayed or even lost. This can cause degradation in
voice quality. Simply put, there are various mechanisms in

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 Telephony VoIP IP Interview Questions And Answers

place to compensate for these problems and help smooth out

the audio.
Once all the packets arrive on the listening end of the
call, they must be reassembled to their original state. The
packets are decompressed and converted from a digital to
analog voice signal.
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