planning and troubleshooting voip performance
DESCRIPTION
Planning and Troubleshooting VoIP Performance shares insights on ThousandEyes helps visualize VoIP routing between branch offices and across the internet, optimize and plan new VoIP deployments and expansions, and troubleshoot VoIP performance to specific problem nodes, links and networks.TRANSCRIPT
Planning and Troubleshooting VoIP Performance
Nick Kephart, Director of Product Marketing Joao Antunes, Software Engineer
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We are building a performance management platform architected for the cloud era We make monitoring complex enterprise networks easy and enable you to find and solve problems regardless of where they occur
About ThousandEyes
Founded in 2010 by UCLA PhDs and backed by:
What We Do Our Background
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• VoIP – Voice over IP – Set of protocols designed to deliver voice communication over the
Internet – H.323, MGCP, SIP, H.248 RTP, RTCP, SRTP, SDP, IAX, etc.
• Two phases: – Signaling (e.g., SIP) – Audio transport (e.g., RTP) with audio frames (e.g., G.711)
What Is VoIP
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Corporate Network A
Corporate Network B
Making a VoIP Call
RTP
SIP SIP
SIP
Internet
SIP gateway
SIP gateway
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• Audio frames are encapsulated in RTP packets • RTP packets are encapsulated in UDP packets • UDP packets are encapsulated in IP packets
Inside a Voice Packet
Frame 1 RTP header Frame 2 IP
header UDP header
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• Audio codec – Generated traffic vs audio compression/quality
• QoS – IP Differentiated Services (DSCP) – Traffic shaping, firewall and LB configuration – 3 bits for class: Best effort, Assured Forwarding, Expedited
Forwarding, Voice Admit
• De-jitter buffer – Network jitter vs call latency
• Network – Latency, loss, etc.
Key VoIP Metrics and Concepts
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Time
Packet delay (from sender to receiver)
Latency
Packet 1 Packet 2 Packet 4 Sent at
Packet 1 Packet 2 Packet 4 Received at Packet 3
Packet 3
Latency Latency Latency Latency
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Time
Variation of the latency
Jitter
Packet 1 Packet 2 Packet 4 Sent at
Packet 1 Packet 2 Packet 4 Received at Packet 3
Packet 3
Min Latency Max Latency
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Time
99.9th percentile of the packet delay variation
Packet Delay Variation
Packet 1 Packet 2 Packet 4 Sent at
Packet 1 Packet 2 Packet 4 Received at Packet 3
Packet 3
Played at
Delayed playback
Min Latency Max Latency PDV = max latency – min latency De-jitter buffer should be able to accommodate PDV.
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E-Model (ITU-T Recommendation G.107, 1998-2014) Based on a mathematical model in which the individual transmission parameters are transformed into different individual "impairment factors” such as codec characteristics, delay, loss ratio, discard ratio, etc., to obtain a quality metric called R factor:
Mean Opinion Score (MOS)
Basic signal-to-noise ratio
Delay impairment
Equipment impairment
Advantage factor
(expectation)
• Network latency • De-jitter buffer size
• Ie (codec) • Packet loss robustness (codec) • Packet loss probability
• Network latency
Simultaneous impairment
4 Rec. ITU-T G.107 (12/2011)
There are three different parameters associated with transmission time. The absolute delay Ta represents the total one-way delay between the send side and receive side and is used to estimate the impairment due to excessive delay. The parameter mean one-way delay T represents the delay between the receive side (in talking state) and the point in a connection where a signal coupling occurs as a source of echo. The round-trip delay Tr only represents the delay in a 4-wire loop, where the "double reflected" signal will cause impairments due to listener echo.
7.1 Calculation of the transmission rating factor, R According to the equipment impairment factor method, the fundamental principle of the E-model is based on a concept given in the description of the OPINE model (see [b-ITU-T P-Sup.3]).
Psychological factors on the psychological scale are additive.
The result of any calculation with the E-model in a first step is a transmission rating factor R, which combines all transmission parameters relevant for the considered connection. This rating factor R is composed of:
AIe-effIdIsRoR +−−−= (7-1)
Ro represents in principle the basic signal-to-noise ratio, including noise sources such as circuit noise and room noise. Factor Is is a combination of all impairments which occur more or less simultaneously with the voice signal. Factor Id represents the impairments caused by delay and the effective equipment impairment factor Ie-eff represents impairments caused by low bit-rate codecs. It also includes impairment due to randomly distributed pack losses. The advantage factor A allows for compensation of impairment factors when the user benefits from other types of access to the user. The term Ro and the Is and Id values are subdivided into further specific impairment values. The following clauses give the equations used in the E-model.
7.2 Basic signal-to-noise ratio, Ro The basic signal-to-noise ratio Ro is defined by:
( )NoSLRRo +−= 5.115 (7-2)
The term No [in dBm0p] is the power addition of different noise sources:
»»¼
º
««¬
ª+++= 10101010 10101010log10
NfoNorNosNc
No (7-3)
Nc [in dBm0p] is the sum of all circuit noise powers, all referred to the 0 dBr point.
Nos [in dBm0p] is the equivalent circuit noise at the 0 dBr point, caused by the room noise Ps at the send side:
( )214004.0100 −−−+−−−= DsOLRPsDsSLRPsNos (7-4)
where OLR = SLR + RLR. In the same way, the room noise Pr at the receive side is transferred into an equivalent circuit noise Nor [in dBm0p] at the 0 dBr point.
2)35(008.0121 −++−= PrePreRLRNor (7-5)
The term Pre [in dBm0p] is the "effective room noise" caused by the enhancement of Pr by the listener's sidetone path:
»»¼
º
««¬
ª++= 10
)–10(
101log10LSTR
PrPre (7-6)
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VoIP Metrics
Average of packet delays
99.9th percentile of packet delay
variation
Packets dropped by the de-jitter buffer
Packets dropped by the network
MOS Score (1-5)
Audio codec used
Source
Destination
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• Regional expansion • New offices, call centers
and locations • Network topology and
routing • Capacity and utilization
Key Use Cases
• Latency, jitter and loss • Infrastructure faults • Routing issues • QoS and DSCP values
Pre-Deployment Post-Deployment
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Corporate Network A
Corporate Network B
VoIP Performance Management
RTP
SIP SIP
SIP
Internet
SIP gateway
SIP gateway
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Corporate Network A
Corporate Network B
VoIP Performance Management
Internet
Agent
RTP
Agent
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How ThousandEyes Voice Tests Work
Enterprise
Enterprise Agent Cloud Agent (at dozens of global POPs)
Active Tests (create a test in each direction)
ThousandEyes SaaS Platform
Branch office
Branch office
Enterprise Agent
External caller
Agent required on both ends
At least one Enterprise
Agent required
Demo
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Add a New Voice Test
Select voice test
Select target
Select source
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Add a New Voice Test: Advanced Settings
On Advanced Settings
Choose codec, DSCP and de-jitter
buffer
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Voice Metrics
Target Agent
Source Agents
Up to 30 day timeline
Metric selector
Metric averages
Jump to other views
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Voice Metrics: Three Agents Experiencing Loss
Source Agents
Packet loss
Lower MOS scores
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Path Visualization: Common Problem Node
Details for problem node
Target Agent
Three agents with issues
Issue summary
View the Live Demo http://vimeo.com/thousandeyes/
voip-monitoring