SIP Audio Problems

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.

SIP Audio Problems

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	Internet quality	Check jitter and quality of internet connection
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

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

Things to investigate

• Check the quality of the 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.

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.