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SIP Audio Problems: Difference between revisions
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! 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
| '''No audio after a few minutes''' || Firewall/NAT/session timeouts || Investigate firewall/NAT/ALG
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* 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 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 ==
== 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
* (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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