VOIP CH 7

the time it takes for speech to exit the speakers mouth and reach the listeners ear

caused by the length a signal must travel via light in fiber or electrical impulse in copper based networks

defines many different causes of delay(actual packetization, compression, and packet switching), and is caused by devices that forward the frame through the network.
includes queuing delay for congested netoworks

propagation delay in conjuction with handeling delay can cause noticeable

cisco uses Digital signal process (DSP) to sample speech using what codec

what does cisco use to keep the overhead low

when packets are held in a queue bc of congestion on an outbound interface
occurs when more packts are sent out thean the interface can handle

the amount of time it takes to actually place a bit or byte onto an interface
its influence on delay is relatively minimal

ITU-T, a delay of no more then

what is the time it takes for a satellite transmission to reach the satellite then back to earth

how many seconds does queueing delay add

the variation of packet interval time

does jitter occure in packet based or switched based networks

the difference between when the packet is expected and when it actually recieved

whick conceals interarrival packet delay variable
-This is bc voice packets have high variable packet interarrival times
-count late packets
-adjust target to allow late-packet ratio

we have to be under 150ms to have this work

used within the cisco ios software to determine of jitter exists within the network

the jitter buffer found in the cisco ios queue can grow or shrink exponentially depending on the interarrival time of the RTP packets

when analog transmission is passed through amplifers to boost the signal not only was the voice boosed so was

Cisco IOS, by default, sends two 10-ms G.729 speech frames in every packet.

the more speech frames you put into a packet the

how do you reduce the IP/RTP/UDP overhead, multiple voice samples can be packed into a

what layer is udp

more voice samples per frame can what, only if bandwidthih is constrained

voice samples per frame and bandwidth utilization impack packet loss, the bigger the value the

so what kind of packet impack can occure

g.729 max voice samples and default

what is the codec of adaptive differential paulse code modulation (ADPCM)

how many bit samples do we use giving a transmission rate of 32 kbps, ADPCM

compression techniques that exploit redudent characteristics of the waveform itsself

techniques employ signal processing procedures that compress speech by sending only simplified parametric information about the original speech excitation and vocal tract shaping, requiring less bandwidth to transmit that information.

64 kbps PCM coding

40,32, 24, 16 kbps ADPCM coding
used between voice, public phone or PBX networks

16Kbps low delay CELP voice compression

when you have bandwidth constrate it can go to 8 kbps
ok quality

low bandwidth compression technique
5.3 and 6.3kbps
good quality but not as good as pbx
flexibilty when you have low bandwith situations
based on CELP

free speach codec
good for voip
slow quality degradation
high rebustness to packet loss
13.33 kbps encoding frame length
15.20 kbps encoding frame length

used for skype and pc to phone application

test a group of listners on voice quality

what is the high and low MOS

what is the best sounding codec at 64 kbps

second best codec at 32kbps

was developed to "hear" impairments caused by compression and decompression and not packet loss or jitter.

for PSQM A person can trick the human ear into perceiving a higher-quality voice, but a

what you hear your voice in the background
caused by 4 wire comversion to wire in traditional network

how does the PSTN stop Echo

what is echo draw drawbacks

where are the echo cancellers built

where can some maufactures stop echo at

where does cisco voip do all its cancellation at

25 ms is good for the user to hear or what will the user think

Echo cancellers are limited by the total amount of time they wait for the reflected speech to be received

classify and manage traffic through a data network to keep packet loss to a minimum

the accepted packer loss on a link is

what device responds if there is periodic packet loss

when you replay the last bit that was not dropped on voice transmission

if lost speech is only 20 sec the listener will kntice what the the speach

not wasting bandwidth when there is not sound to transmit
you can utilize this "wasted" bandwidth for other purposes

regardless if no one is speaking how mush data is moving

how much of a call is wasted bandwidth

when the VAD detects a drop-off of speech amplitude, it waits a fixed amount of time before it stops putting speech frames in packets. This fixed amount of time is known as
(sending silence)

vad is unable to to distinguish between speech and background noise known as

VAD is detecting when speech begins. Typically the beginning of a sentence is cut off or clipped

when a person stops talking not any useable sound

today's toll networks can handle how many a D/A (digital to analog)conversions before voice quality is affected

when using G.729 2 conversations from Digital to Analog will affect what

to tandem encoding. G.729 can handle how many compression/ decompression cycles

Simplify the routerconfiguration
Use a Cisco IOS Multimedia Conference Manager (for instance, H.323 Gatekeeper).
Use one of Cisco's management applications
Thes help you with what

audio and video packets

standard for transmitting delay-sensitive traffic across packet-based networks ( audio video)
Media on demand

RDP uses what to determine whether the packets are arriving in order

determine delay and jitter

to determine the interarrival packet time (jitter).

a thin protocol that provides support for applications with real-time properties, such as continuous media ( transport of the data)

for real-time conferencing of groups of any size within an Internet. ( does not send the payload )

provides a rich set of data for VoIP management

header ip 20 udp 12 rtp 8 how much bigger payload then g.729

the processing time increse compress header
decrease it takes to transmite
slow link throuput is going to increase
faster link the compression will decrease the throughput
*test*

enhances without using TCP
send multiples of the same packet and enable the receiving station to discard the unnecessary or redundant packets.

mechanism makes it more probable that one of the packets will make the journey from sender to receiver

a worthwhile mechanism to enhance reliability and voice quality, if you have unlimited bandwidth

Purchase leased lines
Purchase a telephony Virtual Private Network (VPN)
Take advantage of the existing data infrastructure and put voice on the data network

use seven diget ext when

what is compatible with her internet phone codec

H.225 is for

codetic infro is sent (in packet based system)

proxy is not needed in an

tjr31@zips.uakron.edu is an example of what

converts Bob.nextdoorneighbor.com to a DNS host name and goes to