With FreePBX distro 12.0.37 and Asterisk 13.2, I can’t find an easy way to mark outgoing audio packets for QoS. We have already QoS working in one side (phones to FreePBX) but not in both directions.
1- Where can we set a default value in pjsip.conf for the parameter tos_audio? I would want to set this setting for all endpoints instead of modifying each one by one.
2-Or is that true the correct way to achieve QOS for outbound packets is by using iptables? I’m not sure but I saw that on this thread: Does latest freepbx uses QoS?.
I did a quick search on this forum but I can’t find the information for PJSIP channel. Does that means QOS is not implemented yet for PJSIP in FreePBX?
Thanks!