I experience random packet loss in bridged calls (seen with the pjsip show channelstats command). When a subscriber just listens to IVR menus, everything is OK.
When I analyze rtp packes in Wireshark, I see splashes of jitter on the outgoing channels. The ToS settings are the same. The differences I see is the flags and identification IP protocol parameters. In asterisk the DF (don’t fragment) parameter is set to 1, whereas the peer’s value is set to 0. And the identification param of the incoming channel is 0, whereas that of the outgoing channel is above zero.
Can these parameters cause packet loss or they have little influence on audio quality?