SIP Audio Problems: Difference between revisions

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<indicator name="VoIP">[[File:menu-voip.svg|link=:Category:VoIP_Faults|30px|Back up to the VoIP Faults page]]</indicator>

[[Category:VoIP Faults]]

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.

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= 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 <big>too loud</big> or <small>too quiet</small>

* 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:==

{| class="wikitable"

|+ SIP Audio Problems

|+ SIP Audio Problems, quick fixes or things to do

|-

! Type !! Typical Cause !! Typical fix

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| '''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.

[[File:Voip-audio.drawio.png|800px|frameless]]

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.

[[File:Voip-audio-recording.drawio.png|800px|frameless]]

= Checklist: =

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|-

| '''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...

|}

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|

| Describe the general problem

| colspan="3" style="background-color:#ffffc7;" |

|-

|

| Is it every single call that has the problem?

| colspan="3" style="background-color:#ffffc7;" |

|-

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|-

|

| Who is the internet provider?<br />

| colspan="3" style="background-color:#ffffc7;" |

|-

|

| Does calling *100 give a constant tone?

| colspan="3" style="background-color:#ffffc7;" |

|-

|

| Has a different phone/softphone be tried?

| colspan="3" style="background-color:#ffffc7;" |

|-

|

| What is the internet router being used?

| colspan="3" style="background-color:#ffffc7;" |

|-

|

| colspan="3" | ping voicelesss.aa.net.uk -c 1000

| style="background-color:#ffffc7;" | send results to A&A support

|-

|

| colspan="3" | mtr --aslookup --show-ips -o LA --report-cycles 1 --report-wide voiceless.aa.net.uk

| style="background-color:#ffffc7;" | send results to A&A support

|-

| style="font-weight:bold;" | Test calls to make:

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== Things to investigate ==

⚫

* ~~Check the quality of~~ the internet connection

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)

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* 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==