VoIP and QoS

Linux Advanced Routing and Traffic Control

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Title: VoIP and QoS

There are two Linux routers, a 64K leased line is used
to connect these two machines. Now, we would like to run
Voice over IP between two machines. We have to run QoS and
tag voice packet's TOS field,
otherwise, the voice quality is terrible if we transfer file
and make voice call at the same time.

I am not familiar with tc in Linux. But I have copied a script
from LARTC archives. If the following config is used, the situation
is better than nothing used. The voice packets can be tagged as
TOS=0x21:

tc qdisc add dev hdlc0 parent 1:1 handle 2: pfifo limit 512
tc qdisc add dev hdlc0 parent 1:2 handle 3: pfifo limit 512
tc qdisc add dev hdlc0 parent 1:3 handle 4: pfifo limit 512
tc filter add dev hdlc0 parent 1:0 protocol ip prio 1 u32 divisor 1
tc filter add dev hdlc0 parent 1:0 prio 1 u32 match ip tos 0x21 0xff flowid 1:1

I don't know what it means exactly. But this is used, the voice quality
is bad at the beginning when file is transferred. The voice quality will
"converge" to be good after about 30 seconds.

However, this is not acceptable. Can anyone tell me what is the best
tc config for running VoIP? Thanks very much.


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