IP Telephony Requirements

IP Telephony Requirements
Parameter Requirement
Bandwidth  With ADPCM and no RTP header

compression: 52 Kbps per call
 With G.729a and no RTP header
 With G.711 and no RTP header
Latency and jitter  <100 ms total

for toll-quality
 100 ms less 42 ms allocated for the IP Telephony system yields a 58 ms
budget for the network
 When G.729a encoding is used, 100 ms less 62 ms allocation for the IP
Telephony system yields a 38 ms budget for the network
Latency and jitter  <100 ms total
for acceptable
 100 ms less 42 ms allocated for the IP Telephony system yields a 108 ms
quality
budget for the network
 When G.729a encoding is used, 150 ms less 62 ms allocation for the IP
Telephony system yields a 88 ms budget for the network
Packet loss <1 percent for voice calls and no packet loss for fax and modem calls
Enterprises are rapidly adopting IP telephony for cost savings, productivity gains and
business innovation. But delivering a high-quality voice service takes more than just
buying the latest IP telephony equipment. Successfully deploying IP telephony to your
enterprise also means understanding the requirements for delivering toll-quality voice
over your company’s network infrastructure, and then appropriately planning for,
choosing and deploying the right IP telephony solution. Ensuring the network is ready
for IP telephony is a critical success factor.
To ensure an optimal IP telephony experience, Flexnet requires that customers undergo

thorough assessment of their networks. Flexnet engineers are specially trained to assist
with this assessment. As a result, Flexnet customers enjoy high standards of usability
and manageability, while reducing their total communications costs.
Here are the straight facts about planning for, and deploying, IP telephony in your
enterprise.
Network Requirements for Toll-Quality Voice

The fundamental requirement for achieving toll-quality voice is to deploy IP telephony
over a properly architected network infrastructure.
The LAN/WAN infrastructure must deliver sufficient throughput and meet latency, jitter
and packet loss requirements.
Deliver sufficient throughput:

The amount of bandwidth required for voice depends on the number of simultaneous
calls, the voice encoding scheme used in the IP handset or softphone, and the
signalling overhead.
 The International Telecommunications Union (ITU) G.711 codec is commonly

used in LAN deployments where LAN bandwidth is plentiful. With G.711 and RTP
header compression, each call requires 82 Kbps.
 ITU G.729 is commonly used in a WAN environment because it uses substantially
less bandwidth. With G.729 and no header compression, each call requires 26
Kbps.
The selection of G.729 across the WAN reflects the much reduced bandwidth
available between company sites. Where LAN’s nominally run at 100Mbps or
1Gbps WAN bandwidth may be as low as 512Kbps. The number of concurrent
calls allowed in and out across the WAN, the encoding and the data traffic
requirements need careful consideration and planning.
 With ADPCM and no RTP header compression, each call requires 52 Kbps.
Meet latency and jitter requirements:
Latency is the time from mouth to ear. It is the time it takes for a person’s voice to be
sampled, packetized, sent over the IP network, de-packetized and replayed to the other
person.
The following WAN factors that effect latency are:

 Physical distance on the WAN circuit. The laws of physics dictate the speed of
data travelling over copper and optic fibre.
 The IP ‘cloud’ presented by vendors all look the same but it is essential that there
is a fully meshed infrastructure connecting all states to minimise the transmission
paths and switch hops to arrive at its destination.
In addition the bandwidth’s in the core need to be sufficient to carry the traffic
without delay and congestion.
 The bandwidth at each site is the most obvious consideration but as noted above
the cloud is also critical. The lower the site bandwidth the higher the clocking
delays in transmitting the data across the link.
For example on a 512Kbps link a 100byte packet takes 100/512000 = 0.19ms
on a 4Mbps link the packet takes 100/4000000 = 0.025ms
 Finally a good quality router is essential to minimise delay. This is even more
important if advanced features such as processor intensive QoS are turned on.
If latency is too high, it interrupts the natural conversation flow and can cause the two
parties to confuse latency for pauses in speech. Latency must not exceed 100
milliseconds (ms) one way for toll-quality voice and must not exceed 150 ms one way
for acceptable quality voice. At 150 ms, delays are noticeable, but callers can still carry
on a conversation. Users hear jitter as degraded voice quality.
Jitter is variation in latency over the LAN and WAN, as the IP telephony packets arrive
in uneven patterns at their destination. Jitter has many sources, including network
congestion, queuing methods used in routers and switches, and routing options such as
MPLS or frame relay used by carriers.
The WAN vendors existing network performance, ability to scale as their combined
customer base grows its data volumes and the fundamental design factors discussed
above all affect the ability of the network to provide consistent performance minimising
jitter.
Packet loss requirements:

 The IP ‘cloud’ must be fully meshed connecting all states to minimise the
transmission paths and switch hops to arrive at its destination.
In addition the bandwidth’s in the core need to be sufficient to carry the traffic
without delay and congestion.
 Site access technology is critical. Vendors offer a wide variety of technologies to
meet a range of price and performance needs.
o ADSL is generally accepted as the lowest cost solution but is a contention
based technology. This means you are guaranteed to loose some voice
UDP packets as the data passes between the site and the core network.
ADSL is not recommended unless there is no other alternative.
o G.SHDSL or SHDSL can provide a non-contended access minimising the
potential for packet loss. Care must be taken to ensure a 1:1 ratio and that
the offer is not just symmetric ADSL.
o Frame Relay and Ethernet technologies also deliver excellent non-
contended performance.
o Wireless technologies are also an option for primary or back-up connections.
They fall into two bread categories:
o Microwave point to point services. These are affected by weather,
transmission distance, local terrain and topology and can be problematic.
o Mobile phone network based 3G data is emerging as a popular solution.
While an excellent solution for data back-up the underlying variability of
performance makes it unsuitable as a primary voice path.
Performing a Network Assessment
Flexnet’s IP Telephony Network Assessment is a complete service available to help you
plan, design and implement an IP telephony solution that meets your organization’s
MEASUREMENT GOOD ACCEPTABLE POOR
MOS Above 4.0 4.0 to 3.6 Below 3.6
MEASUREMENT GOOD POOR
Delay (msec) Below 150 Above 150
Jitter (msec) Below 30 Above 30
Loss (percent) Below 1 Above 1

specific needs and helps ensure that IP telephony will run smoothly. The assessment is
performed by qualified engineering team and is required prior to deployment.
Flexnet’s IP Telephony Network Assessment combines real-time and simulated testing,

which results in the pre-emptive discovery of network faults and potential performance
problems. Flexnet uses a network performance industry leading management tools that
use active application traffic to monitor and test actual applications and servers. It also
collects passive performance information from IP PBXs, gateways and other network
components. This provides repeatable, real-world tests for the most comprehensive
performance assessment.
The tests simulate IP telephony and the ability of the network to handle latency, jitter
and packet loss (see Table 2). Flexnet also reports on voice quality in the form of a
mean opinion score (MOS), which is a five-point scale established by the ITU in which 1
represents the poorest voice quality and 5 represents perfect voice quality.
By performing a pre-deployment network assessment, organizations gain an end-user

perspective of network behaviour. The management tools agents allow almost
immediate performance testing all the way to the desktop without the time, expense and
security concerns of deploying physical resources or installing client software.
The initial results of a network assessment are delivered in minutes, although a

thorough assessment test typically runs for several days. It is vital that assessments be
performed during peak operation hours to ensure an accurate picture of the network
traffic. If trouble spots arise before deployment or during ongoing operations, these tools
enable your solution provider or IT team to rapidly isolate the source of the problem.

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IP Telephony Requirements

Bandwidth  With ADPCM and no RTP header

Latency and jitter  <100 ms total

To ensure an optimal IP telephony experience, Flexnet requires that customers undergo

Network Requirements for Toll-Quality Voice

Deliver sufficient throughput:

 The International Telecommunications Union (ITU) G.711 codec is commonly

The following WAN factors that effect latency are:

on a 4Mbps link the packet takes 100/4000000 = 0.025ms

Packet loss requirements:

MOS Above 4.0 4.0 to 3.6 Below 3.6

MEASUREMENT GOOD POOR

Delay (msec) Below 150 Above 150

Jitter (msec) Below 30 Above 30

Loss (percent) Below 1 Above 1

Flexnet’s IP Telephony Network Assessment combines real-time and simulated testing,

By performing a pre-deployment network assessment, organizations gain an end-user

The initial results of a network assessment are delivered in minutes, although a

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