voip and vofi – quietly competing for your network bandwidth

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Jay Botelho Director of Product Management WildPackets [email protected] Follow me @jaybotelho

VoIP and VoFi Quietly Competing For Your Network

Bandwidth

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© WildPackets, Inc. 2 VoIP and VoFi – Quietly Competing For Your Network Bandwidth

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


Overall VoIP Requirements and Metrics


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/ns525/ns537/ns705/ns827/white_paper_c11-481360.pdf






Key RTP (VoIP/VoFi) Issues

CAMP IT Pinpointing the Problem 5

Jitter Latency

Packet Loss


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


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


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.


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

3 1

2

4


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


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


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


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


How RTP and TCP Coexist


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


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


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


Quality Score Trending • Isolated scores are useful for validating single call

complaints, but overall VoIP health is best seen by graphing long-term trends

0.000.501.001.502.002.503.003.504.004.505.00

8:45

9:15

9:45

10:1

5

10:4

5

11:1

5

11:4

5

12:1

5

12:4

5

13:1

5

13:4

5

14:1

5

14:4

5

15:1

5

15:4

5

16:1

5

16:4

5

17:1

5

Mea

n O

pini

on S

core

Time

MOS

Overlaying VoIP trends with network utilization, errors, or other metrics may reveal previously unseen performance

relationships!


Data Impacts on VoIP


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


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”


Unique Challenges of VoIP at 10G


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


Unique Challenges of VoFi


A Real World Example

Source: Farpoint Group


Quantitative Interference Impacts on VoFi



Quantitative Interference Impacts on VidFi



Packet-by-Packet

VoFi Call

Wired VoIP Call


Troubleshooting RTP Issues


VoIP Dashboard View


Calls View


Media View


Down To The Details …


Call Data Record (CDR)

Provides comprehensive, real-time statistical and quality report for base-

lining, and 100% visibility into calls


Identifying Unauthorized RTP Traffic • Look for bandwidth hogs

• Use filters and alarms


Q&A

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#wp_voip with any questions, comments, or feedback.

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Thank You!

WildPackets, Inc. 1340 Treat Boulevard, Suite 500 Walnut Creek, CA 94597 (925) 937-3200

voip and vofi – quietly competing for your network bandwidth

Technology

ms packet jitter

jitter latency packet

voip current latency

voip systems

jaybotelho voip

data network

ms of missing audio

jitter buffer packets