voip monitoring and analysis - still top of mind in network performance monitoring

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VoIP – Still Top of Mind in Network Performance Monitoring

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TRAC NPM Research Demographics

Sep 2013

406 participants

Company type: 70% - Enterprise 28% - Service

Providers

Company size: 41% - Large organizations 38% - Medium 21% - Small

Geography 56% - North America 24% - EMEA 14% - APAC

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Key IT Initiatives Impacting Network Performance

BYOD

Public Cloud services

Video conferencing

Virtual desktops

Enterprise Mobility

Big Data

VoIP

48%

54%

59%

65%

66%

69%

72%

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Ability to improve performance for home office users

Managing recreational traffic

Increase in number of IP flows to be managed

Managing bandwidth consumption per user/subscriber

Managing real-time applications (VoIP, video, etc.)

36%

40%

41%

59%

64%

Key Challenges for Managing Network Traffic

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Key Challenges for Managing UC Performance

Time spent on extracting session details

Inability to correlate multiple network layers

Visibility into bandwidth utilization

Visibility into the quality of user experience

Visibility into each session for UC technologies

Visibility into the impact of UC deployments on other applications on the network

24%

31%

38%

44%

44%

52%

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And It’s Not Just VoIP …

• 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 will increase 17-fold by 2015

http://www.cisco.com/en/US/solutions/collateral/ns341/ns525/ns537/ns705/ns827/white_paper_c11-481360.pdf

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VoIP/Video Is Just Different!

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VoIP/Video Is Just Different!

• Packet delivery not guaranteed

• Small and consistent packet sizes

• Highly regular packet spacing

• Reliable packet delivery

• Large and variable packet sizes

• Widely varying packet spacing

VoIP Data

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Key RTP (VoIP/VoFi/Video) Issues

CAMP IT Pinpointing the Problem 9

Packet Loss

JitterLatency

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Latency

Queue Latency & Decision LatencyQueue Latency & Decision Latency

Network Propagation

Delay

Network Propagation

Delay

Encoding / DecodingCompression / Decompression

Jitter Buffer Latency

Encoding / DecodingCompression / 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

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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.

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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 anddelayed 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

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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

• Packet dropped due to physical layer corruption

• Congestion without adequate QoS provisions

• Jitter buffer discards due to excessive latency

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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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Measuring Key RTP Metrics

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Making Sense of the Metrics

• A single value to reflect the user’s QoE (quality of experience)

• 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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Key Metrics So Far

• Latency

• Jitter

• Packet Loss

• MOS

• R-Factor

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Measuring MOS and R-Factor

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Can RTP and TCP Coexist?

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Coexistence Is Possible, But Challenging

• While traditional network applications are very tolerant of jitter, latency, and even to some degree packet loss, VoIP/Video/VoFi is not

• Tolerable levels of jitter, latency, and packet for TCP are devastating for RTP

• Pre- and post-deployment network assessments are critical‒ Pre-deployment: understanding your network’s ability to accommodate

VoIP• Current latency, jitter, and packet loss• QoS capabilities• Current bandwidth utilization (is there any room for VoIP)• Mix of RTP vs. TCP• Is all RTP traffic equal?

‒ Post-deployment: maintain a constant vigil after deployment to watch for imminent troubles

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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

• Understanding the character of network traffic is key‒ “Bursty” traffic - rapid, recurring traffic spikes that can occur over

long periods of time‒ Prolonged, slow rises in utilization may decrease the number of

calls that can occur simultaneously over the course of a day‒ Sharp spikes can 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 Callx2112

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

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)

• 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:‒ 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

• Watch for too much differentiated traffic!

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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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Monitoring QoS Configuration

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Unique Challenges of VoFi

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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

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Identifying Unauthorized RTP Traffic

• Look for bandwidth hogs

• Use filters and alarms

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Q&A

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