ONT – VoIP Implementation Pt.1

What are the main benefits of packet telephony networks?

– More efficient use of bandwidth and equipment, and lower transmission costs.

 

– Consolidated network expenses

 

– Improved employee productivity

 

– Access to new communications devices

Describe the general use of Gateways in a packet telephony network.
Gateways interconnect and allow communication among devices that are not all necessarily accessible from within the IP network.
Describe the use of Mulipoint Control Units (MCU) in a packet telephony network.
An MCU is a conference hardware component. It is comprised of a Multipoint Controller and an optional Multipoint Processor that combines the received streams from conference participants and returns the result to all the conference participants.
Describe the general use of Gatekeepers in a packet telephony network.
Gatekeepers can provide both call routing, which is essentially resolving a name or phone number to an IP address, and Call Admission Control, which grants permission for a call setup attempt. They are also used as a central database of call extensions when many CallManager clusters exist.
Describe the main concept differences between digital signal and VoIP.

In almost all cases, digitizing analog voice needs to be performed, regardless of whether calls stay within a PBX system, or traverse through an IP network.

 

With VoIP, in addition to digitizing voice, requires IP-based signaling, and conversion of analog voice into IP packets and transport using IP-based protocols suck as Realtime Transport Protocol (RTP). 

What are the three most popular VoIP signaling and control protocols?

– H.323, which is an ITU standard

 

– Media Gateway Control Protocol (MGCP), which is an IETF standard

 

– Session Initiation protocol (SIP), another IETF standard 

Describe the process of call routing during call setup.
Call routing involves the destination telephone number being resolved to an IP address.
Describe the basic function of Call Admission Control (CAC) during a call setup.
Call Adminssion Control (CAC) is an optional step that determines whether the network has sufficient bandwidth for the call. If bandwidth is inadequate, CAC sends a messages to the initiator indicating that the call cannot get through because of insufficient resources.
What are some of the important parameters that must be negotiated during call setup?

– The IP addresses to be used as the destination and source of the VoIP packets between the call end points.

 

– The destination and source UDP port numbers that the RTP uses at each call end point.

 

– The compression algorithm (codec) to be used for the call. 

Describe how a distributed call control model works in a packet telephony network.
In a distributed call control model, multiple devices are involved in setup, maintenance, teardown, and other aspects of call control. The voice-capable devices that perform these tasks have the intelligence and proper configuration to do so. Protocols classified as distributed include H.323 and Session Initiation Protocol (SIP)
Describe how a centralized call control model works in a packet telephony network.
In a centralized call control model, centralized call control relieves the gateways and end points from being responsible for tasks such as call routing, call setup, Call Adminssion Control (CAC), and call teardown. Media Gateway Control Protocol (MCGP) is an example of a centralized call control protocol.
What are the four major steps in converting analog voice signals to digital ones?

1. Sampling

 

2. Quantization

 

3. Encoding

 

4. Compression (Optional) 

Describe the process of sampling is analog-to-digital conversion.
Sampling is the process of periodic capturing and recording of voice. The result of sampling is called a pulse amplitude modulation (PAM) signal.
Describe the process of quantization in analog-to-digital conversion.
Quantization is the process of assigning numeric values to the amplitude (height or voltage) of each of the samples on the pulse amplitude modulation (PAM) signal using a scaling methodology.
Describe the process of encoding in analog-to-digital conversion.
Encoding is the process of representing the quantization result for each pulse amplitude modulation (PAM) sample in binary format.
Explain how pulse code modulation (PCM) is used in converting analog voice signal to digital voice signal.
Pulse code modulation (PCM) is based on taking 8000 samples per second and encoding each sample with an 8-bit binary number, generating 64,000 bits per second (64Kbps) without compression.
What are the steps taken in converting a digital signal back to analog?

1. Decompression (optional)

 

2. Decoding and filtering

 

3. Reconstructing the analog signal 

Explain the basics behind the Nyquist Theorem
Based on the Nyquist theorem, a signal that is sampled at a rate at least twice the highest frequency of that signal yields enough samples for accurate reconstruction of the signal at the receiving end.
What are the three major services provided by a Digital Signal Processor?

1. Voice termination

 

2. Transcoding

 

3. Conferencing

Describe how transcoding works on a Digital Signal Processor.
Transcoding is a DSP resource that preforms codec conversion when 2 parties in an audio call use different codec.
Which UDP port numbers are reserved for RTP flows?
Ports 16384 through 32767 are reserved for RTP.
How much payload does Real Time Protocol (RTP) add to the encapsulated voice payload?
A VoIP packet adds 20 bytes for IP, 8 bytes for UDP, and 12 bytes for RTP to the encapsulated voice payload in the form of headers.
Describe how Compressed Real Time Protocol (cRTP) reduces the overhead imposed on the voice payload.
After an initial packet with all the headers is submitted, the following packets that are part of the same packet flow do not carry the 40 bytes of headers. Instead, the packets carry a hash number that is associated with those 40 bytes. The premise is that most of the fields in the IP, UDP, and RTP headers for not change among the elements of a common packet flow.
By how much is the overhead reduced when using Compressed Real Time Protocol?
This would depend on the header checksum (UDP checksum). If cRTP does not use this checksum, the size of the overhead is reduced from 40 bytes to only 2 bytes. If the checksum is used, the overhead is reduced to 4 bytes.
On what type of links should Compressed Real Time Protocol be enabled on?
Cisco recommends that cRTP only be enabled on slow link (links with less than 2Mbps). Its also recommended that cRTP only be performed by hardware, because of the processing overhead.
How many Compressed Real Time Protocol (cRTP) sessions are supported using Cisco’s IOS?
Cisco allows up to only 16 concurrent cRTP sessions by default. If enough resources are available on a device, you can increase this value.

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