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13 Cards in this Set

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What is the first step in VoIP?

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.

How is the sound sent over the internet?

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).

What causes degradation?

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 place to compensate for these problems and help smooth out the audio.

What happens when the packets reach their destination?

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.

What is required for VoIP?

Generally following things are required for voip:


1. Broadband connection


2. voip phone


3. nexton softswitches


4. router


5. audiocodec


6. astric server



What are the advantages of VoIP?

Cost is cut nearly in half. 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.

What do internet telephony,packet telephony,IP telephony and converged network means?

The first thing all mean the same thing. Which is using IP (Internet protocol) for voice services.

What 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.

What is a converged network?

A converged network is a network that passes both Voice and 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.

What can you tell me about the OSI Reference Model?

The OSI Reference Model provides a framework for discussing network design and operations. It groups communication functions into 7 logical layers, each one building on the next.

The OSI Reference Model provides a framework for discussing network design and operations. It groups communication functions into 7 logical layers, each one building on the next.



What are the benefits of subnetting?

Subnetting helps reduce network traffic and the size of the routing tables. It’s also a way to add security to network traffic by isolating it from the rest of the network.

How will you check if router goes down?As a Engg, in client place router goes down & client do not know anything about it, how will you check it & how you will make it up (by siting in your office)? What are all precheck you will do in client place & what cmd you will use to check it?

PING that router


check routing table


sh up route


shw ip route


shw ip int brief

How would your run a voip test setup?

press and release the mute button, press 8378# (TEST) this will start self test