Here is a list of things to go through when dealing with VoIP Quality issues with VoIP calls, eg: calls breaking up, garbled audio, silence or gaps in the call.

This page doesn't have all (or many) answers, but poses a whole list of things to try and do to pin-point the cause of audio problems. The aim is that by going through these points the customer and Support Staff will have a good understanding on how the problem is manifesting and what's been tried to resolve it so far. It will then make a base for further investigations.

Assumptions:

We assume calls are working but there are audio problems - the quality of the sound on the call is not as expected
If you have registration problems then this page isn't going to help much
CODECs are not mentioned here, as A&A only support G7.11 A-LAW CODEC

Brief overview

Here is a very brief overview of types of audio problems, their cause and the fix. This is not exhaustive, and usually there isn't a quick fix and diagnosing and investigation is required.

Types of problems that could be reported

VoIP calls audio has echo echo
VoIP calls are too loud or too quiet
VoIP calls sound like they are under water
VoIP calls are silent
Audio starts out ok, but stops after a few seconds or a few minutes
Audio has gaps, missing words

General direction to take with different types of audio problem:

SIP Audio Problems, quick fixes or things to do
Type	Typical Cause	Typical fix
No audio	On mute, handset loose cable?! Firewall/NAT blocking traffic	Investigate firewall/NAT/ALG
Short silent gaps	Packet loss	Check internet quality
Audio garbled Intermittently	Internet quality	Check jitter and quality of internet connection
Audio garbled consistently, perhaps only one way	RTP packet size	Check your system sends 20ms (ptime, packetisation, packet time) More info below.
No audio after a few minutes	Firewall/NAT/session timeouts	Investigate firewall/NAT/ALG

Calling *100 will play a continuous 1Khz tone - if the audio breaks up then there is a problem

Audio transportation

The cause of an audio problem needs to be determined, especially if it's intermittent or only in one direction.

This diagram shows the possible places (each arrow) that could affect audio on a call.

The checks and questions outlined below go to some length to try to pinpoint which step is introducing the audio problem.

A&A Recordings

A&A Call recording can narrow down the cause of the problem, as it captures the audio part way through the path. This can tell if we are receiving bad audio from either party or not and so is a good way to narrow down the cause of the problem.

Checklist:

NAT

If your phones are are on a 192.168.x.x or 10.x.x.x IP address then they will be behind a NAT router. Our VoIP and NAT page has information and tips if your phones are behind a NAT router and you have problems.

Things to figure out

Dealing with audio problems can be tricky, we need to identify as much detail about when the problems happens to aid investigation.

SIP Audio problems, things to figure out
Question	More info
Exactly how is audio affected?	gaps, garbled, delayed, silence after a few minutes, etc.
Heard on call recordings?	this feature can be enabled on the control pages Our call recording software has a large jitter buffer and rearranges out of order packets, so bear in mind it is possible for a garbled call to have a perfect call recording.
Just incoming or outgoing calls,	...Or on both? Knowing the direction us useful
internal to A&A calls OK?	if you call another A&A number such as our technical support. Or is it only affecting calls that are sent outside of A&A
BOTH mobiles and landlines	If only mobiles, is it specific to a particular carrier, eg onle EE, or Three, etc
Are both sides of the call affected	or just one? Which side?
If problem is outbound...	A&A Support can provide details on how to test making calls via the different upstream carriers that we use
What has changed recently?	eg, phone or pbx software upgrade, new broadband router, new ISP, new phone etc...

A Checklist:

When investigating audio problems, especially if it's only on some calls and in one direction, it may be useful to go through a list of call types and describe what happens on each call for each person on either end of the call.

This table may help in these cases, make test calls and fill in the yellow boxes.

Table of questions and test calls to try

General Questions	When did this start?
	Describe the general problem
	Is it every single call that has the problem?
	Can the problem be reproduced on demand?
	Who is the internet provider?
	Does calling *100 give a constant tone?
	Has a different phone/softphone be tried?
	What is the internet router being used?
	ping voicelesss.aa.net.uk -c 1000			send results to A&A support
	mtr --aslookup --show-ips -o LA --report-cycles 1 --report-wide voiceless.aa.net.uk			send results to A&A support
Test calls to make:
		Is it OK?	If not, describe problem	Describe what's heard on the call recording
An incoming call from a mobile to the A&A Service
	Audio from the mobile
	Audio to the mobile
An outgoing call to a mobile
	Audio from the mobile
	Audio to the mobile
An incoming call from a 'land line' to the A&A Service
	Audio from the landline
	Audio to the landline
An incoming call from the A&A Office
	Audio from the office
	Audio to the office

Things to investigate

When did the problem start? - what changed around that time?

Was phone system upgraded?
Was the telephone software upgrades?
Was the internet connection changed?
Was the internet router changed?
etc

Then:

Check the quality of the internet connection - broken audio can be down to a poor internet connection
Calling *100 will play a continuous 1Khz tone - if the audio breaks up then there is a problem
Check the CQM graph (if using A&A for the internet connection)
Check for other traffic on the internet connection - spikes in traffic can affect audio
Check latency/jitter during a call - eg ping voiceless.aa.net.uk 1000 times and look at the min/max levels
A&A can set up a ping graph to your IP address - Let Support staff know your IP address. This will then graph loss and latency

Things to swap/change

The aim here is to change things one by one to target in on the thing that may be the cause.

Disable bonding, if bonding is in use (Bonding can add to jitter)
Try just the phone on the internet connection, plugged directly in to the router - unplug/disable all other devices, disable WiFi (this also takes the LAN out of the equation as a problem on the LAN can affect the traffic)
Try a different phone or SIP Client
(if non-A&A internet) Try a different Internet connection, including using a different ISP
Try non-NAT
Disable/Enable any SIP ALG on the broadband router - if it's a feature
If it's only the incoming audio that is broken, look for the jitter buffer setting on the SIP device and increase it bit by bit.
For those using the Grandstream GXP1625, set jitter type from adaptive to fixed. Leave all other settings as default.

Advanced troubleshooting

Get a pcap and use Wireshark to decode it
Wireshark can play audio with different jitter buffer settings so you can sometimes recreate audio problems from the pcap
Wireshark can calculate jitter and delta stats
- Packets can arrive bunched up, with gaps between, or out of order. Jitter is a measurement of this.
- Wireshark calculates jitter as per the definition in RFC3550 so it uses a standard formula.
- Delta is the time difference between arrival of RTP packets. Max delta is the highest gap between arrival of packets. If the max delta is greater than the jitter buffer, audio will start getting dropped. Each packet usually contains 20ms of audio, so for high enough max delta you'll start to lose audio in 20ms chunks.

Choppy or garbled audio heard by the other party

Symptoms

People you call, or who call you, say your voice sounds choppy, robotic or garbled, or as if parts of words are missing.
You can hear them clearly. The problem only affects audio going from your PBX.
It happens on every call, not just now and then.
The A&A call recording may well sound perfect in this circumstance!

Likely cause: the PBX sends the wrong packet size.

Call audio is sent in small packets. At call setup, your PBX and our platform agree how much audio goes in each packet, called the packetisation time or ptime. Our service uses 20 ms, which is the standard for G.711.

Some PBXs agree to 20 ms but then send larger packets, usually 30 ms or 40 ms. Some carriers and networks further along the call can't handle packets that don't match what was agreed, and the far end ends up hearing only part of the audio.

How to fix it

Set your PBX to send 20 ms packets:

In your PBX, find the settings for the SIP trunk that connects to us.
Look for a setting called ptime, packetisation, packet time, RTP packet size or frame size, often under the codec settings.
Set it to 20 ms for G.711 A-law (PCMA).
If there's no per-trunk option, check the global SIP/RTP/codec settings instead.
If your PBX passes audio straight from your phones to the trunk (sometimes called "direct media" or "media passthrough"), also check the handsets or their provisioning templates are set to 20 ms.
Save, apply, and make a test call to someone who can tell you how you sound.

Checking with a packet capture (advanced)

If you can capture traffic on your PBX or router, open the capture in Wireshark and look at RTP packets sent by your PBX:

What to check	Correct (20 ms)	Problem (40 ms)
Frame length (G.711, IPv4 over Ethernet)	214 bytes	374 bytes
RTP payload size	160 bytes	320 bytes
RTP timestamp step between packets	+160	+320
Time between packets	~20 ms	~40 ms