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.

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About ThousandEyes
Network Intelligence platform
that gives you a complete
picture from users to internal
and cloud-based applications
Surface insights from
a global data set
Lightweight, flexible
data collection
Unified view of diverse
performance data
Solve issues across
shared infrastructure
See any network like
it’s your own

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VoIP Monitoring
• VoIP Monitoring is critical because it is:
• A key component of the UC and UCaaS suite of services
• Very sensitive to network issues
• Notoriously difficult to troubleshoot
• Accelerated UCaaS adoption in the past two years
• “Early adopter phase” -> “Early mainstream phase”
• Increased reliance on the Internet and 3rd party infrastructure
• Traditional voice monitoring is reactive and expensive
• Based on call detail records (CDR) and packet capture (pcap)
• Limited for benchmarking and planning scenario’s

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So how does VoIP work?
• Session Initiation Protocol (SIP)
• Pre-requisite for the voice call
– RFC 3261: Standard protocol (however
propriety versions exist to force vendor
lock-down)
– Application level protocol residing above
TCP/IP stack
– TCP or UDP
– Text-based protocol like HTTP
– Encrypted with TLS
– Response Codes indicates the state of the
request message
Phase1: Signaling
VoIP Phone A SIP Server/Proxy VoIP Phone B
SIP RegisterSIP Register
SIP INVITE
100 Trying
SIP INVITE
180 Ringing
180 Ringing
200 OK
200 OK
AUDIO CALL
SIP BYE
200 OK
SIP ACK SIP ACK
SIP BYE
200 OK

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So how does VoIP work?
Phase1: Signaling
VoIP Phone A SIP Server/Proxy VoIP Phone B
SIP RegisterSIP Register
SIP INVITE
100 Trying
SIP INVITE
180 Ringing
180 Ringing
200 OK
200 OK
AUDIO CALL
SIP BYE
200 OK
SIP ACK SIP ACK
SIP BYE
200 OK
REGISTER
INVITE
CONNECT
DISCONNECT

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• Real Transport Protocol (RTP)
– Analog voice signals converted into data
packets and sent over UDP
– Audio frames are encapsulated in RTP
packets
– RTP packets are encapsulated in UDP
packets
– UDP packets are encapsulated in IP
packets
So how does VoIP work?
Phase 2: Audio Data
IP
header
UDP
header
Frame 1
RTP
header
Frame 2 Frame 3
RTP Audio Stream
SIP Network

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• How voice traffic is encoded and decoded
• Determines the quality of the VoIP conversation
• G.711, G.722, SILK
Key VoIP Concepts & Metrics
• MoS
• Latency
• Jitter (De-Jitter buffer)
• PDV
Codecs
QoS
• Prioritization of VoIP Traffic
• DSCP codes
– Traffic shaping, firewall and LB configuration
– 3 bits for class: Best effort, Assured Forwarding, Expedited
Forwarding, Voice Admit
VoIP Metrics

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Proactive VoIP Monitoring
Monitor an end-to-end VoIP call Keep track of VoIP quality metrics
Find QoS Changes Along the Path Measure SIP performance
Audio Codec Used
MoS Score
Average of packet
delays
Packets dropped by
the de-jitter buffer

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Key Use Cases
• Regional expansion (new
offices, call centers and
locations)
• Migration to UCaaS or
hybrid solution
• Network topology and
routing
• Capacity and utilization
• Latency, jitter and loss
• Infrastructure faults
• Routing issues
• QoS and DSCP values
Pre-Deployment Post-Deployment

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How ThousandEyes Voice Tests Work
1
Monitor your on-prem/hosted server from Enterprise Agents in branch offices.
RTP tests require agents on both ends of the call.
For inter-office monitoring across the enterprise WAN simulate VoIP calls
between Enterprise Agents in branch offices.

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How ThousandEyes Voice Tests Work
2
Monitor on-prem or hosted SIP server from Enterprise Agents in branch offices.
RTP tests require agent on both ends of the call. Requires at least one Enterprise Agent.
To understand performance to external locations, simulate VoIP calls between your
offices and globally located Cloud Agents.

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How ThousandEyes Voice Tests Work
3 Monitor an end-to-end voice call through your UCaaS VoIP provider from both branch
offices and external locations.

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• Monitor ONLY the SIP infrastructure from within the
corporate network or global locations.
• Supports TCP/UDP; SIP Options and SIP Register
• SIP Metrics: Availability, Response Time, Total Time
Choosing the Right Voice Test
• Combines both SIP Server and RTP Stream tests
• Closest to emulating end-user experience
• For eg: Failed SIP INVITE from an agent will not
trigger a RTP call
SIP Server
Voice Call
• Simulate a RTP voice stream between two agents.
• Requires at least one Enterprise Agent
• RTP Metrics: MOS, Latency, Jitter, PDV
RTP Stream

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Demo

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Troubleshooting SIP Infrastructure
Identify errors in SIP
signaling phase
SIP Metrics: Availability,
Response Time, Total
Time
Monitor SIP infrastructure
from your WAN or global
vantage points
Correlate SIP server
availability to the network
topology and
characteristics

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Troubleshooting SIP Infrastructure
Read SIP headers to
diagnose issues

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Monitor VoIP Call Quality
Identify issues in call
quality by tracking key
VoIP call quality metrics
Dallas agent shows a drop
in MOS score
Identify SIP
proxies in the
path
Voice calls combine both
SIP signaling and RTP
audio streams sequentially

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See what you’re missing.
Watch the webinar:
https://www.thousandeyes.com/resources/voip-webinar