ChecklistDesign

Network of VoIP
Tips
Good quality
Checklist of voice-over-IP
VoIP Network Design(VoIP)Tips
conversations depend on maintaining strict con
straintsandforjitter
delay, packet(contrast
loss, this with traditional data network traffic, where th
e focus is onWe¡¦ve
throughput). responsefoundtimethese
or design tips successful when deploying VoIP in a data
.Use
Codecs network.
theare
G.711
thecodec
hardware
end-to-end,
or software
unless
usedlack
to convert
of capacity
fromrequires
analog tocompression.
digital and b
ack. The
gives theG.711
best voice
codec quality, since it does no compression, introduces the least
than delay,
other
andcodecs
is lesstosensitive
packet loss. Other codecs, like G.729 and G.723, consume le
ss bandwidth but
compression, by doing
this introduces delay and makes the voice quality very sensitiv
e to packet
.Keep
Packet lost
losspackets.
loss well
occurs because
belowof1%congestion
and avoid orbursts
electromagnetic
of consecutivenoise.
lostItpackets.
can also o
ccurthe
and whenjitter
jitterbuffer
is highis too small to compensate. Increased bandwidth and good t
uninga small
network
.Use cancongestion,
often
speech
reduce
frame
which,sizein and
turn,reduce
reducesthejitter
numberandof speech
packet frames
loss. per packet.
When voice traffic is a stream of small packets, the effect of one being lost is
big lesspacket
severewith
thanmultiple
losing speech
a frames inside it. A good target is 20ms of speec
h perframe
one frame,
perwith
packet. Of course, using small packets increases the total bandwid
th requirement,
each
.Always
Packet-losspacket
use packet-loss
requires
concealment
becauseitsconcealment
masks
own fixed-size
the loss
(PLC).
ofheader.
a packet or two by using information f
rom the Packet
packet. last goodloss can occur randomly or in bursts. PLC helps with random packe
t loss.
for
.Actively
One-way doingdelay
The
minimize
PLCcost
=ispropagation
minimal,
one-way delay,
since
delayit¡¦s
keeping
+ transport
already
it below
usually
delay150ms.
+ packetization
part of the codec
delayprocessing.
+ jitt
er buffer
Voice quality
delaydegrades quickly when the total one-way delay is greater than 150m
.Propagation delay is the time to travel the physical distance from end to end. F
s.
or example,
take a signalitabout
may 100ms to go from Dallas to Singapore. When the traffic has t
o cover
like
.Transport
this,
long
delay
make
distances
surethethetotal
is network
timepath
spentisinside
as direct
eachasofpossible.
the devices in the networ
k, like switches,
routers, gateways, traffic shapers, and firewalls. Some devices add more latency
athan
software
others;firewall
for example,
running on a slow PC adds more delay than a dedicated hardwa
firewall. Look at the number of hops traveled by the voice traffic. Reduce the n
re-based
umberways
find
.Packetization
of hops
to reduce
and is
delay thethelatency
fixedintime
theneeded
devicesforthat
theare
codec
thetoworst
do its
offenders.
job. The G.
711 codecpacketization
smallest imposes the delay. In contrast, the codecs that do compression add de
lay ranging
25ms to 67ms.fromAlso, avoid converting from one codec to another along the network
.Jitter
path buffer delay is used to dampen variations in packet arrival rates. If the
and
network
the jitter
delay is high,
low you can afford to have a larger jitter buffer than in a
network high.
already
Copyright where
ý NetIQ
the Corporation,
delay is 2001. All Rights Reserved.
If you¡¦re
.Avoid usingconsidering
slow speed VoIP,
links.don¡¦t consider using it extensively on slow serial links.
Upgrade
on thosethe
paths
bandwidth
so the VoIP traffic and existing data traffic have plenty of room
.Use
VoIP to RTP
breathe.
traffic
headeruses
compression
the Real-time
for slow-speed
Transport Protocol
links. (RTP) to encapsulate the spee
ch frames.
header compression
RTP (called ¡§cRTP¡¨) can reduce the 40-byte RTP headers to a tenth of th
eir original
size, lowering the bandwidth consumed between routers. Enable it when there¡¦s a lin
k on throughput
with the path less than 500 kbps. So, why not always use cRTP? It adds latency
.Use datause
Routers packet
datafragmentation
packet fragmentation
for slow-speed
to cut large
links.packets into smaller ones, an
d thenatreassemble
them the other end. On slow links, this helps assure that small VoIP packets
don¡¦t
behind
.Use
Testingcalllarge
get
can
admission
delayed
help
datadetermine
packets.
to protect
Enable
the against
maximum
it for
too
number
link
manyof
speeds
concurrent
concurrent
belowcalls.
T-1.
VoIP conversations t
hat carry
can your network
with good quality. For example, your results may show that 50 calls ca
n be carried
would set thewell.
call admission
You threshold in the VoIP server or gateway to prevent
the priority
call
.Use
Voice 51st
from
traffic
simultaneous
being
scheduling
hasestablished.
stricter
forpacket-loss,
voice traffic.delay, and jitter requirements than trad
itionalsense
makes network
thattraffic,
it shouldsoreceive
it an appropriate quality of service (QoS). A pr
eferred
mark VoIPQoSpackets
methodwith
is tothe DiffServ setting for Expedited Flow (EF). Also, consi
der using
Fair QueuingWeighted
(WFQ), which raises the priority of low volume traffic. Giving VoIP
routers
higher decide
prioritywhich
helpstraffic to forward first when congestion occurs. Watch for
adverse
existing
.Get youraffects
data network
traffic,
on theready
though.
for VoIP, fully upgraded and tuned, before starting
a VoIPdata
Most deployment.
networks today aren¡¦t ready to carry good-quality voice conversations. It¡¦s
easy to aassess
whether network is ready or not, though, because VoIP traffic can be simulated
can
andbeitsmeasured
characteristics
and analyzed. This means you can make all the changes needed in
the network
their successandbefore
assurebeginning an expensive VoIP deployment.