We are experiencing a random problem with audio delays on calls. It’s not an echo, but the audio each direction will be delayed be about 2-3 seconds periodically. I am thinking it might be a codec issue, but I’m having a little bit of trouble figuring out which codec is being used in each call. I’m currently running this:
watch "asterisk -vvvvvrx 'core show channels verbose'"
But it doesn’t seem to show the codec, unless I’m missing something. We’ve had this same FreePBX server for about 3 years with no other issues. SIP trunks are from Flowroute. Any feedback on where I should drill down into troubleshoot would be appreciated.
A codec issue would not introduce delay like this, unless for some reason transcoding was going on and it somehow took 2-3 seconds to transcode. If that was happening you’d have bigger problems.
Delays like this are usually network related or upstream carrier issues.
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Indeed. VoIP is based on RTP (Real Time Protocol). So, If you’ve got any issue on your network, you could have lots of audio problems.
Some stuffs:
- Check QoS. (DiffServ 46) it should be handled evry where in your network. (Switch, Router).
- Check an eventual loopback in your network.
- Test ping: It’s not 100% efficient, but it can show some latencies at first.
- Sometimes, needs to restart some switch or router
- Check if the switch port is set 100 Full Duplex and your NIC card too. If your switch port is set 100 Half Duplex and your NIC card is set 100 Full Duplex, that can works but you will have lots of packets dropped.
So side your switch 100 Full, or Auto.
Just an idiea like that.
Can you try adding some trunks from another provider to test via Outbound Routes ?
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