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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| '''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...

| '''What has changed recently?''' || eg, phone or pbx software upgrade, new broadband router, new ISP, new phone etc...

|}

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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==