SSRC changing every 30 seconds

We use Asterisk 13 in our commercial PBX product, pretty successfully. I am the company president, so forgive me if I am asking a naive question, but I do know enough to be dangerous :slight_smile: I would ordinarily task my developer with this, but I decided to take the opportunity to register here myself and maybe learn something.
We’ve been troubleshooting a particular customer issue, where a certain model phone (VTech SIP Conference phone) will drop out incoming audio, very briefly, at regular 30-second intervals. In analyzing the Wireshark captures, I am seeing the PBX change the SSRC header in the RTP being sent to the phone, at approximately every 30 seconds. Meanwhile, the reverse RTP stream from the phone to the PBX - and both streams between the PBX to the SIP server - keep the same SSRC for the entire call.
Is there a configuration line that we are missing, or have incorrect, that is telling Asterisk to re-assign the source identifier?

Hi, could you ever figure out what was causing the change of the SSRC?

No, I never got a reply to help me understand it.

What channel driver are you using chan_sip or pjsip, also I dont think there is an option on chan_sip who allows you to control the The SDP media attribute ssrc, also post you sip configuration will help