Watch the full OnDemand Webcast: http://bit.ly/VoIPVoFI It seems like just a few short years ago when the growth of VoIP on traditional networks was one of the hottest topics. But VoIP is now in use by just about every company with a network, large or small. And with the growth of 802.11 and Wi-Fi enabled smart phones, the use of voice over Wi-Fi (VoFi) is in hot pursuit. VoIP has unique, and typically more demanding, requirements for network performance than "traditional" network data. But one network carries all your traffic, both data and voice, so you need one network analysis solution to get an accurate picture of how VoIP is performing and what might be causing call quality degradation. VoIP traffic could be trumping your data traffic, while data traffic can create a log jam for VoIP. You need a single, simultaneous view of ALL network traffic to adequately monitor and troubleshoot overall network performance. Join us as we dive into this topic, working our way from overall requirements for voice transmission on IP networks down to detailed analysis of VoIP performance and even individual VoIP calls, all on highly utilized, high speed networks. In this web seminar, we will cover: - Overall VoIP requirements and metrics - How VoIP (RTP) and data (TCP) coexist - Unique challenges of VoIP on 10G networks - Unique challenges of monitoring and analyzing voice over Wi-Fi (VoFi) You will learn how to: - Determine if your network is up to the challenge of VoIP/VoFi - Troubleshoot voice quality issues, even down to individual calls - Identify conflicts between VoIP and traditional data traffic - Monitor VoIP on 10G networks in real time

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Jay Botelho
Director of Product Management
WildPackets
jbotelho@wildpackets.com
Follow me @jaybotelho
VoIP and VoFi
Quietly Competing For Your Network
Bandwidth
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Agenda
• Overall VoIP Requirements and Metrics
• How RTP (VoIP) and TCP (Data) Coexist
• Unique Challenges of VoIP at 10G
• Unique Challenges of VoFi
• Troubleshooting VoIP Issues
• Company Overview
• Product Line Overview

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Overall VoIP Requirements
and Metrics

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Maybe Not So Quietly …
• Global Internet video traffic surpassed global peer-
to-peer (P2P) traffic in 2010
• Percentage of overall consumer Internet traffic
‒ 2010: 40%
‒ 2012: 50% (e)
‒ 2015: 62% (e) (excluding P2P video file sharing)
• Percentage of all forms of video (TV, VoD, Internet,
and P2P) will be approximately 90 percent of global
consumer traffic by 2015
• Internet video to TVs tripled in 2010 and will increase
17-fold by 2015
http://www.cisco.com/en/US/solutions/collateral/ns341/n
s525/ns537/ns705/ns827/white_paper_c11-481360.pdf

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Key RTP (VoIP/VoFi) Issues
CAMP IT Pinpointing the Problem 5
Jitter
Latency
Packet
Loss

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Assessing the Monster's Impact
• While traditional network applications are very tolerant of jitter,
latency, and even some degree of packet loss, VoIP/Video/VoFi
is very sensitive to these troubles
• Levels of jitter, latency, and packet loss that would be easily
tolerated on a data network can be devastating on a converged
VoIP/Video network
• Pre- and post-deployment network assessments are critical
‒ You must understand your network’s ability to accommodate VoIP
• Current latency, jitter, and packet loss
• QoS capabilities
• Current bandwidth utilization (is there any room for VoIP)
‒ You must maintain a constant vigil after deployment to watch for
imminent troubles

7. © WildPackets, Inc. 7VoIP and VoFi – Quietly Competing For Your Network Bandwidth
Latency
Queue Latency &
Decision Latency
Network
Propagation
Delay
Encoding / Decoding
Compression / Decompression
Jitter Buffer Latency
0 ms
100 ms
200 ms
300 ms
400 ms
500 ms
600 ms
700 ms
800
ms
The ITU
recommends a
maximum one-
way delay of
150 ms for
VoIP

8. © WildPackets, Inc. 8VoIP and VoFi – Quietly Competing For Your Network Bandwidth
Latency's Effects
• Talkover
‒ Occurs when excessive latency delays audio
– Conversation cadence is not natural or comfortable
– Callers feel as if they must “push to talk” or say “over” to control the
conversation
• Echo
‒ The speaker’s voice feeds back into the listener’s microphone
‒ The speaker then hears his own voice returning from the listener’s end,
but delayed due to latency
‒ Most callers find it difficult to maintain normal speech when echo delay
is prolonged
‒ Some VoIP systems attempt to cancel echo, but are not always
successful
High latency may also cause additional troubles such as loss of
synchronization between audio and video for multimedia sessions.

9. © WildPackets, Inc. 9VoIP and VoFi – Quietly Competing For Your Network Bandwidth
Jitter
• Jitter is the variance in packet delivery intervals to the listener
• Jitter buffer adds additional delay to voice reaching the ear piece in
case other packets need to catch up
• Packets delayed too long in the network are not allowed to enter the
jitter buffer
Packets delayed more than the buffer delay
(100 ms as an example) are dropped
. . .. .. . . ........
......
Packets are buffered and
delayed at the Receiver
The “jitter” buffer releases
a G.711 packet every 20 ms
A G.711 packet sent
every 20 ms
Packet jitter and drops
31
2
4

10. © WildPackets, Inc. 10VoIP and VoFi – Quietly Competing For Your Network Bandwidth
Jitter's Effects
• Jitter causes weird “sound effects” that vary with
jitter severity and environmental factors
• Examples include:
‒ Static
‒ Stuttering or uneven audio – abnormal speech rhythm
‒ For multimedia systems, video may be “jerky” or irregular
• If jitter levels are high, packet loss can result
‒ In some cases, severe jitter may sound similar to packet loss,
even if no packets are actually dropped

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Packet Loss
• Most commonly caused by:
‒ Packet dropped due to physical layer corruption
‒ Congestion without adequate QoS provisions
‒ Jitter buffer discards due to excessive latency
• Causes missing sounds, syllables, words, or phrases
‒ DSP algorithms may compensate for up to 30 ms of missing
data
‒ More than 30 ms of missing audio (e.g. 2 RTP packets for
G.711) is noticeable by listeners

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Packet Loss Effects
• Causes missing sounds, syllables, words, or phrases
‒ DSP algorithms may compensate for up to 30 ms of missing data
‒ More than 30 ms of missing audio is noticeable by listeners
• An average person speaks at a rate of about 200 words per
minute
‒ That’s 3.33 words/sec = 300 ms per word
‒ For G.711, we would need to lose 15 consecutive RTP packets to lose
a whole word
‒ Dropping 15 packets/sec for G.711 would be a loss rate of 30%
• But losing only a few packets can still be very noticeable
‒ Loss of more than 2 consecutive packets will be heard
‒ Loss rates ≥ 2% will have a strong impact on quality
‒ Losses of 5 – 10% make calls all but intolerable
‒ Bursty periods of packet loss are worse than more dispersed loss

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Quality Scoring for VoIP
• One of the best initial troubleshooting tools for VoIP traffic
• Mean Opinion Score (MOS) – several flavors
‒ Algorithmic simulation of subjective audio assessment
‒ Most commonly used varieties are MOS-LQ (listening quality) and
MOS-CQ (conversational quality)
‒ Possible range of 1 (poor) to 5 (excellent)
‒ Maximum possible MOS = 4.4 with G.711
‒ Typical range in most networks is 3.5 – 4.2
• R-Factor – several flavors
‒ Based on latency, jitter, packet loss, bit rate, and signal-to-noise ratio,
codec effects (for low bit-rate codecs)
• The ITU algorithms consider about 20 quality inputs
‒ Possible range of 0 (poor) to 100 (excellent)
‒ Provides LQ, CQ, and other score variants

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How RTP and TCP Coexist

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Network Traffic: Qualitative Analysis
• The quality of your network traffic is potentially more
important than its quantity when it comes to VoIP
• Many traffic streams are “bursty” in nature
‒ Burstiness my occur over long period of time, or may consist of
rapid, recurring traffic spikes
‒ Prolonged rises in utilization may decrease the number of calls
that can occur simultaneously
‒ Sharp spikes may cause very noticeable quality issues with
ongoing calls
• Your baseline monitoring should consider not only
averages and long-term trends, but also the short-
term peaks and dips that characterize your traffic
flow

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Network Traffic: Quantitative Analysis
• Most network engineers are concerned about the
amount of traffic on their networks
‒ Utilization (percentage of bandwidth)
‒ Throughput (bits or bytes per second)
• You also need to be concerned about individual
utilization components
‒ How much bandwidth and throughput can be attributed to each
application or process?
• Clarifies which application traffic may need to be tuned or
controlled
‒ How well or poorly will the baseline (trended) behavior of each
application interact with VoIP
• Don’t forget to also consider the reverse case – VoIP’s impact on
existing applications

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The Impact of "Just One More Call"
• Although a network link may be able to support a
number of concurrent calls, one additional call is
often enough to cause quality problems…
x1113
x2111
x1112
x1111
1st Call
2nd Call
3rd Call
x2112
x2113
Example: The WAN can support 2 simultaneous calls.
What happens when a third call is attempted???
Call #3 Causes Poor Quality for ALL Calls

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Quality Score Trending
• Isolated scores are useful for validating single call
complaints, but overall VoIP health is best seen by
graphing long-term trends
0.00
0.50
1.00
1.50
2.00
2.50
3.00
3.50
4.00
4.50
5.00
8:45
9:15
9:45
10:15
10:45
11:15
11:45
12:15
12:45
13:15
13:45
14:15
14:45
15:15
15:45
16:15
16:45
17:15
MeanOpinionScore
Time
MOS
Overlaying VoIP trends
with network utilization,
errors, or other metrics
may reveal previously
unseen performance
relationships!

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Data Impacts on VoIP

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Got QoS?
• One of the most potent weapons for fighting VoIP
troubles is to provision Quality of Service (QoS)
parameters
• QoS enables network devices to prioritize and give
preference to packet streams that are sensitive to delay,
packet loss, jitter, and other performance inhibitors
• Standards-based QoS methods include:
‒ RSVP (nearly antiquated)
‒ IP Differentiated Services (DiffServ)
‒ MAC Layer QoS with IEEE 802.1p
‒ VLANs
• QoS may be obtained or supplemented via proprietary
means, such as traffic shaping via various flow
processing algorithms

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Ready for QoS?
• QoS provisions are based on the “weakest link”
concept
‒ If any device in a data path does not support QoS, then media
streams will not be afforded the preference they require for good
performance
• Pre-deployment assessment must ensure that ALL
devices can recognize and respond to QoS
parameters in packet headers
‒ Switches, routers, firewalls, proxies, and any other devices that
touch RTP packets must be “VoIP-friendly”

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Unique Challenges of VoIP at 10G

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Current Challenges
• Traditional NICs not up to the task
• Processing power is a limiting factor
• Storage capacity is a limiting factor
• I/O bus and disk write speeds are a limiting factor
• 10G forces clarity in analysis
• At 10G, it truly is looking for a needle in a haystack
• “Line rate” – be wary of that claim!
Importance: Packet-based PM tools remain only truly effective
approach to definitive monitoring and definitive troubleshooting –
Jim Frey, Enterprise Management Associates, Inc., July 2010

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Unique Challenges of VoFi

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A Real World Example
Source: Farpoint Group

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Quantitative Interference Impacts on VoFi
Source: Farpoint Group

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Quantitative Interference Impacts on VidFi
Source: Farpoint Group

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Packet-by-Packet
VoFi Call
Wired VoIP Call

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Troubleshooting RTP Issues

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VoIP Dashboard View

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Calls View

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Media View

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Down To The Details …

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Call Data Record (CDR)
Provides comprehensive, real-time
statistical and quality report for base-
lining, and 100% visibility into calls

35. © WildPackets, Inc. 35VoIP and VoFi – Quietly Competing For Your Network Bandwidth
Identifying Unauthorized RTP Traffic
• Look for bandwidth hogs
• Use filters and alarms

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Q&A
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