What is the cause of the "translate.c: "Sample size different 320 vs 160"" error?

Question

Asterisk

Asterisk 13.8.1.
A call is made from a SIP phone using the G722 codec to a SIP phone using the alaw codec within the local network. The subscriber of the G722 cannot hear the other party.
DEBUG fails:

res_rtp_asterisk.c: Unsupported payload type received
translate.c: "Sample size different 320 vs 160"

At the same time, if you make a call back from alaw to G722, everything works.
2 solutions help: use the same codec on all devices, or disable bridge_native_rtp.
But I would like to understand what causes this error?