The problem is:
When i call using my voip phone, the other person listen me very slowly and with echo..., is because these traffic NOT PRIORIZE because not entry in my mangle...
This is a torch when speaking or calling (voip server ip is: 212.230....):
If i know dst address ( its a static ip ), but random port, ¿how to priorize or mark in these mangle?
All statistics are empty packets...
Actually, my VOIP traffic not entry in this mangle.
->
Code: Select all
add action=mark-connection chain=VoIP comment="Flujo VoIP" disabled=no \
new-connection-mark=VoIP passthrough=yes port=5060 protocol=udp
add action=mark-connection chain=VoIP comment="" disabled=no layer7-protocol=\
sip new-connection-mark=VoIP passthrough=yes
add action=mark-connection chain=VoIP comment="" disabled=no layer7-protocol=\
skypetoskype new-connection-mark=VoIP passthrough=yes
add action=mark-connection chain=VoIP comment="" disabled=no layer7-protocol=\
h323 new-connection-mark=VoIP passthrough=yes
add action=mark-connection chain=VoIP comment="" disabled=no dscp=46 \
new-connection-mark=VoIP passthrough=yes
add action=mark-connection chain=VoIP comment="" disabled=no dscp=11 \
new-connection-mark=VoIP passthrough=yes
add action=mark-connection chain=VoIP comment="" disabled=no dscp=26 \
new-connection-mark=VoIP passthrough=yes
add action=mark-connection chain=VoIP comment="" disabled=no \
dst-address-list=VoIP new-connection-mark=VoIP passthrough=yes
add action=mark-connection chain=VoIP comment="" connection-type=sip \
disabled=no new-connection-mark=VoIP passthrough=yes
add action=mark-connection chain=VoIP comment="" connection-type=h323 \
disabled=no new-connection-mark=VoIP passthrough=yes
add action=change-dscp chain=VoIP comment="" connection-mark=VoIP disabled=no \
new-dscp=46
add action=mark-packet chain=VoIP comment="" connection-mark=VoIP disabled=no \
new-packet-mark=VoIP_P passthrough=no