Being writtenNetwork & voice

VoIP call quality troubleshooting

Why calls break up on a connection that tests clean, and how to prove where the fault is.

LevelPractitioner
Material3 modules, 10 lessons
Reading time1 h 49 m
StatusBeing written — not open

This course is not available yet

It is being written. The outline below is what it will cover, and it is unlikely to change much. There is no enrolment and no date — when it is ready it will be on this page. Several of the lessons are built around tools on this site that already work, so you can practise the measurements today.

If you need this capability now rather than later, UCaaS & voice is the service that covers it, or talk to us.

Who it is for

Anyone who has been told the phone system is broken and suspects the network, or the other way round.

Practitioner. For people who administer these systems day to day.

Assumed beforehand:

  • The addressing and DNS material from Network fundamentals, or equivalent experience

What you will be able to do

  • Explain why bandwidth is rarely the cause of bad call quality
  • Measure jitter, loss and latency under load, and say which figure explains the symptom
  • Calculate what a call really costs on the wire, including headers and any VPN
  • Describe how NAT breaks media and what SIP ALG does about it, usually badly
  • Present evidence that distinguishes the carrier, the LAN and the platform

Outline

3 modules, 10 lessons, 1 h 49 m of material. The times are the sum of the lessons rather than a round figure.

01What actually ruins a call

Four impairments, each with a different cause and a different fix.

Jitter and the buffer that hides it12 m

Why variation matters more than delay, and what a jitter buffer costs.

Loss, and why concealment stops working10 m

One percent, bursts against random loss, and what the codec can paper over.

Latency under load12 m

Bufferbloat: the connection that tests fine until somebody uploads something.

One-way delay and the 150 ms budget10 m

Where the budget goes, including the parts you control.

02Codecs and capacity

The codec rate is the audio. The circuit carries considerably more than that.

What a call costs on the wire12 m

RTP, UDP, IP and framing, and why a compressed codec saves less than its bitrate.

Packetisation, and trading delay for bandwidth10 m

The interval, the packet rate, and the delay each choice adds.

Sizing a circuit, and prioritisation12 m

Concurrency, headroom, and why a full link queues.

03NAT, SIP and the things in between

Signalling and media take different paths, and NAT understands neither.

Why NAT breaks media12 m

Address rewriting, symmetric RTP, STUN and TURN.

SIP ALG, and turning it off9 m

What it tries to do, and why it is the first thing to disable.

Reading an estimated MOS honestly10 m

The E-model, its inputs, and what an estimate is not.

Format. Written lessons paired with this site’s VoIP test, bufferbloat test and bandwidth calculator, so each measurement is one you have taken yourself. Delivery mechanism is not built yet.

Would you rather we just did it?

This course exists because the work behind it is work we do. If you need the result rather than the skill, ucaas & voice is the service that covers it.