Choppy audio for bridged remote extensions

Hi, I’m stumped.
I have FreePBX behind a router using NAT. Extensions on the LAN can call remote extensions fine and vice versa. If two remote extensions call each other or make a PSTN via our SIP provider, the audio is VERY choppy. I have enabled the jitter buffer. The problem only seem to occur when two remote channels are bridged.

FreePBX 13.0.197.22 and Asterisk 13.28.1

Any help or suggestions appreciated.

Can you give us some more details about the devices being used? For example, what are the make and model of the remote phones? Are the remote phones in separate locations? Are there any QoS settings enabled on any of the routers? If so, can you try disabling that, as well as any jitter buffer settings if they didn’t make a difference.

Hi, Thanks for the interest.
The remote devices are Google Pixel 2 Phones with Google standard SIP account settings operating over relatively strong mobile 4G internet. FreePBX is connected to the internet on a 100/40 Mbit/s HFC and also work well. I’ve turned off the jitter setting in FreeBPX. I’m using ChanSIP driver.
What has me stuck is that it is only calls that traverse the NAT twice that has the issue. I thought it might be the mobile internet that is causing the issue, but calls to and from the LAN devices (on Yealink W52P) work perfectly. When the remote extension calls out to our SIP service provided are also choppy, this would have only one mobile internet connection.

I’ve looked at the asterisk channel details of these calls and each leg uses G.711a (default in Australia), so there is not transcoding happening in the FreePBX bridge.

I agree that it really doesn’t seem like a bandwidth issue since the remote calls to local extensions sound fine. I’m also not familiar with using Google sip accounts. Do you have any other remote devices/softphones you can set up a test with to see if the problem is unique to the Pixels?

Hi,

I have not a chance to try other handsets/softphones yet, but I was able to turn on recording in FreePBX. Interestingly, the recordings in both directions is sounds clean.

This seems to indicate that is incoming streams from both channels is received fine (and recorded fine) but is being "corrupted" in transmission.

More soon I hope.

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.