I have an intermittent problem where three to 5 second dropouts are occuring during calls. Wireshark shows no packet loss
, but a huge number for jitter. It will go from .05ms throughout the call and suddenly spike to 33554288.93. For the particular call example, this happened on the reverse leg between the PBX and the Yealink SIP-T48G phone. This is FreePBX 14 distro, BTW.
I have QoS enabled on the switches with DSCP. The Yealink phones use 46 for RTP and I have mapped that to COS 7 in the switches. The trunk is a Spectrum in house fiber SIP trunk, but as I said this occurred between the PBX in house and the phone. Iperf3 between the PBX and the PC that is downstream from the phone shows .045 jitter and there are no obvious network problems that I can observe.
What I’m hoping for here is some experienced troubleshooters to give me some suggestions on how to figure out why the jitter suddenly goes off the scale. I included a screenshot of the RTP Stream analysis from wireshark.
This looks like a deep problem with Asterisk that is discussed in https://community.asterisk.org/t/wireshark-incorrect-timestamp-and-audio-shift/81177 although I’m surprised you are not seeing the marker bit set when the timestamps jump.
Most devices have no problem with these timestamp jumps, but a few can take some time to recover.
Note, although wireshark calculates a jitter from this, it isn’t really jitter. real jitter would result in large variations in delta ms.
That sure does look similar. In reading the link, the one guy kept asking the question what problem are you having. In my case customer reports 4-5 seconds of missing audio during random calls. These are Yealink phones. I don’t know if that is relevant.
So is there a phone or freepbx setting that I can tweak to work around this issue?
Anyone? Any suggestions on a setting that I can tweak for the extension or Yealink phone to make it less sensitive to timestamp jumps?
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