VoIP call quality troubleshooting: jitter, packet loss and one-way audio
For service desk and network technicians who take "the phones sound terrible" and "they can't hear me" tickets for customers with SIP phones, hosted PBXs or SIP trunks. You should know basic IP addressing and NAT; Packet capture basics helps but isn't required. You'll learn how SIP sets up a call and RTP carries the audio, how jitter and loss are measured, why NAT and SIP ALG cause one-way audio and calls that drop after about 30 seconds, and how to prove which it is from a capture. Ends with a supervisor-graded call investigation.
- Level
- Intermediate
- Length
- About 75 minutes
- Contents
- 4 lessons · final exam
- Status
- Published · updated 10 Oct 2026
Skills you'll practise
- Identify the SIP requests and responses of a normal call and the responses that signal common failures
- Calculate packet loss and the RFC 3550 jitter estimate from given values, and choose the next diagnostic step for choppy or robotic audio on Wi-Fi, VPN or congested links
- Identify the cause of one-way audio or a call that drops after about 30 seconds from SDP, Contact headers and firewall evidence
- Interpret Wireshark's RTP stream analysis (lost packets, sequence errors, maximum delta, jitter) for each direction of a call
Course outline
- 1.SIP requests and responses: signalling and media take different pathsLesson · 15 min
- 2.Calculating jitter and packet loss, and the next step for choppy or robotic audioLesson · 18 min
- 3.One-way audio and calls that drop at 30 secondsLesson · 17 min
- 4.Proving it with Wireshark's RTP stream analysisLesson · 12 min
- 5.VoIP call quality troubleshooting: jitter, packet loss and one-way audio: knowledge checkKnowledge check · 14 questions
- 6.VoIP call quality troubleshooting: jitter, packet loss and one-way audio: practical exerciseKnowledge check · 1 question
- 7.Final exam8 questions · passing it completes the course, so people who already know the material can test out
Sources it draws on
The lessons and questions are written from these references, so learners can go back to the original.
- RFC 3261: SIP, Session Initiation Protocol
- RFC 3550: RTP, A Transport Protocol for Real-Time Applications (sequence numbers, timestamps, RTCP reception reports, interarrival jitter)
- Twilio Elastic SIP Trunking: Troubleshooting your Trunk (one-way audio, private IPs in SDP and Contact, calls dropping after 20 to 30 seconds)
- Microsoft Learn: Prepare your organization's network for Teams (jitter and loss symptoms, VPN, QoS, Wi-Fi, NAT session persistence)
- Wireshark User's Guide 9.11: RTP streams and RTP stream analysis
- Wireshark User's Guide 9.3: VoIP Calls window
- VoIP-Info wiki: Routers SIP ALG (how SIP ALGs rewrite SIP and SDP, and common failures)
See it with your own jobs and topics
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