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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.
= 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.
{| class="wikitable"
! Type !! Typical Cause !! Typical fix
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| '''No audio''' || On mute, handset loose cable?! Firewall/NAT blocking traffic || Investigate firewall/NAT/ALG
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| '''Short silent gaps''' || Packet loss || Check internet quality
= 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 NAT|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.
* Exactly how is audio affected? gaps, garbled, delayed, silence after a few minutes, etc.▼
** 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.▼
* Is the audio problem present on 'internal to A&A' calls - eg, 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▼
* If only mobiles, is it specific to a particular carrier, eg onle EE, or Three, etc▼
* Is the audio problem heard on both sides of the call, or just one? Which side?▼
* If problem is outbound, then A&A Support can provide details on how to test making calls via the different upstream carriers that we use.▼
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|+ SIP Audio problems, things to figure out
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! Question !! More info
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| '''Just incoming or outgoing calls,''' || ...Or on both? Knowing the direction us useful
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== Things to investigate ==
* (if non-A&A internet) Try a different Internet connection, including using a different ISP
* Try non-NAT
*
* 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.
* 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 [
** 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 [
** 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.
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