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barto95
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Posts: 8
Joined: Mon Dec 20, 2021 4:51 pm

HELP CONFIGURATION QOS FOR VOIP

Wed Jan 19, 2022 4:59 pm

HI,

My network is
192.168.1.0/24 = DATA = bridge1
192.168.2.0/24 = VOIP = VOIP (vlan tag2)
Internet = pppoe-out1 = adsl up/down (900k/11LM)

this is my configuration :
/interface bridge port
add bridge=bridge1 ingress-filtering=no interface=ether2
add bridge=bridge1 ingress-filtering=no interface=ether3
add bridge=bridge1 ingress-filtering=no interface=ether4
add bridge=bridge1 ingress-filtering=no interface=ether5
add bridge=bridge1 ingress-filtering=no interface=wlan1
add bridge=bridge1 ingress-filtering=no interface=wlan2
/interface list member
add interface=bridge1 list=LAN
add interface=VOIP list=LAN
add interface=pppoe-out1 list=WAN
My sip proxy/sip server is accessible on domain name xxx.yyyy.com (TRUNK) and on aaa.bbbb.com (Centrex)
port used for sig = 5060 and for RTP = 10000-65000
this is my config for adress list
/ip firewall address-list:
add address=aaa.bbbb.com  list="VOIP_SBC"
add address=xxx.yyyy.com list="VOIP_SBC"
and for mangle:
/ip firewall mangle
add action=mark-connection chain=forward dst-address-list="VOIP_SBC" dst-port=5060 in-interface=VOIP new-connection-mark=SIP-SIGNALING-IPCX-Conn passthrough=yes protocol=\
    udp
add action=mark-connection chain=forward dst-address-list="VOIP_SBC" dst-port=5060 in-interface=VOIP new-connection-mark=SIP-SIGNALING-TRUNK-Conn passthrough=yes protocol=\
    udp
add action=mark-packet chain=forward connection-mark=SIP-SIGNALING-Conn new-packet-mark=SIP-SIGNALING-IPCX-Packet passthrough=yes
add action=mark-packet chain=forward connection-mark=SIP-SIGNALING-TRUNK-Conn new-packet-mark=SIP-SIGNALING-TRUNK-Packet passthrough=yes
add action=mark-connection chain=forward dst-address-list="VOIP_SBC" dst-port=10000-65000 new-connection-mark=SIP-RTP-IPCX-Conn passthrough=yes protocol=udp
add action=mark-packet chain=forward connection-mark=SIP-RTP-IPCX-Conn new-packet-mark=SIP-RTP-IPCX-Packet passthrough=yes
add action=mark-packet chain=forward connection-mark=SIP-RTP-TRUNK-Conn new-packet-mark=SIP-RTP-TRUNK-Packet passthrough=yes
add action=mark-connection chain=forward dst-address-list="VOIP_SBC" dst-port=10000-65000 new-connection-mark=SIP-RTP-TRUNK-Conn passthrough=yes protocol=udp
and my config queue tree:
/queue tree
add max-limit=850k name="Total UPLOAD" parent=global
add limit-at=10k max-limit=10k name="SIP Signalling up" packet-mark=SIP-SIGNALING-IPCX-Packet,SIP-SIGNALING-TRUNK-Packet parent="Total UPLOAD" priority=7
add limit-at=200k max-limit=200k name="SIP RTP up" packet-mark=SIP-RTP-IPCX-Packet,SIP-RTP-TRUNK-Packet parent="Total UPLOAD" priority=6
add max-limit=750k name="Other Traffic" packet-mark=no-mark parent="Total UPLOAD"
My ISP is MY OPERATOR for voip and the packet is tagged with DSCP 26 and 46 in upload (the ip phone send the dscp and the SBC/SIP SERVER send DSCP to phone) the network LNS/BACKBONE prioritize the paquet with DSCP TAG, it's not necessary for me to add queue tree for download because the LNS do that (QOS in Download looks on mikrotik WAN interface)

What's wrong on my configuration for VOIP packet is 100% prioritize of all other packet and do not add jitter/packet loss ?

Thanks for your reply and excuse my English … :)
 
sindy
Forum Guru
Forum Guru
Posts: 10206
Joined: Mon Dec 04, 2017 9:19 pm

Re: HELP CONFIGURATION QOS FOR VOIP

Wed Jan 19, 2022 8:41 pm

Forum rule number one - don't post what you think is related to your issue, post the complete configuration, except sensitive information - see my automatic signature below.

In the absence of the complete configuration, here's my experienced guess: the default configuration of SOHO devices, including hAP ac², contains an action=fasttrack-connection rule in chain forward of /ip firewall filter; in order than mangling and queues worked the way you want, you have to disasble that rule, as the very essence of fasttracking is that packets belonging to fasttracked connections bypass mangling.

BTW, what's your native language if you can reveal that?

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