dear experts,
I have my mikrotik server with 1 Real internet Ip address, and 1 Fake Ip address ( 192.168.0.0/24 )...
am facing a problem here, each program for voice like skype, msn messenger, yahoo messenger, voiplowrate, or net2phone or each calling cards programs, when I connect with these programs, the clients on other side can hear me, but me no... I cant hear them, and the voice is so bad... its like cuted voice...
anyone can know this problem and how to deal with it or fix it plz??
by z way, in my firewall filters, am opening all the ports of UDP and TCP from ( 1 - 65535 )
so any Idea plz?

How is your Nat set up? Do you have queues? What is the rate/technology to your ISP? How much traffic are you running?

I have no problems with a similar setup on even a 133c

here is my nat:
the 11.11.11.0/24 is my LAN range, and the 192.168.60.1 is my squid server
chain=srcnat action=masquerade out-interface=Real
src-address=11.11.11.0/24 protocol=tcp

1 chain=dstnat action=redirect to-ports=8080 in-interface=Fake
src-address=11.11.11.0/24 dst-port=80 protocol=tcp

2 chain=dstnat action=dst-nat to-addresses=192.168.60.1 to-ports=800
dst-address=11.11.11.0/24 dst-port=8080 protocol=tcp

3 X chain=dstnat action=dst-nat to-addresses=192.168.60.1 to-ports=800
in-interface=Fake dst-port=8080 protocol=tcp



I have my simple queues: each client has 24k Upload and 64k Download
MY bandwidth is 1280 K download with 384k Upload...

NB: my real MRTG is not fully complete on my upload , the maximum upload I got is about 250k only

hi there we are facing the same problem here what i found was the best way is to set priority to that trafic but unfortunatly i beeing a new be was a bit stuck with that

any help will be nice

How is your Nat set up? Do you have queues? What is the rate/technology to your ISP? How much traffic are you running?

I have no problems with a similar setup on even a 133c

from what i can see is that its not the nat rules but issue of priority set on the packet and treatment of the same when it hits the box

Greetings!

Maybe turning up the bandwidth a little wouldn't hurt. 24K up and 64K down is very low. I use 384K/1M.

Greetings!

Maybe turning up the bandwidth a little wouldn't hurt. 24K up and 64K down is very low. I use 384K/1M.

i dont know abt the rest of the world but bandwith is at a very high premium here in africa and cant afford to give that sort of bandwidth as cant buy it so have to manage with what we have in hand

it would be a grate help if you can suggest how to go abt marking say net2phone trafic and give it higher priority we can replicate the same with other voice providers

regards

I was not trying to say go to 384K/1M. Just increase it a little. That upload speed is not even AOL dialup. You get 38K from AOL. Use 64K and 64K and see if it is "kinda bad" both ways, just as a trial.

I was not trying to say go to 384K/1M. Just increase it a little. That upload speed is not even AOL dialup. You get 38K from AOL. Use 64K and 64K and see if it is "kinda bad" both ways, just as a trial.

the problem is when i am on full load say 100 concuratnt users and my total available bandwith is 512/1M what is the best config you can suggest for say max spped browsing and voice call with bandwith dropage for downloads
regards

Just boost up ur bandwidth,
i think it 'll solve ur problem.





thnxs,

Just boost up ur bandwidth

Dear Raktim, how anyone can boost up bandwidth without buying more!!??

I think the problem is about upload bandwidth here, buy more upload bandwidth if you can and you can mark your voice packets and prioritized it in the queue tree to the higher level.

I'm currently running commercial VOIP in Africa using Mikrotik. By setting voice priority slightly higher in the queue tree and by managing the bandwidth carefully, I've got 12 voip users and about 130 standard internet users on 2m down 1m up. It can be done.

for v3 + disable SIP in ip->firewall->service ports. That will solve the one-way audio issue in voip applications.

for v3 + disable SIP in ip->firewall->service ports. That will solve the one-way audio issue in voip applications.

Thanks for the tip. Our VOIP has been good, with a few issues at random. We'll see what this does for us.

kathemeen,

Have you had this cause problems even in routed networks without NAT? Does this setting only affect NATed traffic?

Thanks
Scott

I think 24k is to low for Skype, have a look at the website, as I can recall you need about 30k or 35k to work good. Also there is newer version that is more efficient. Let both sides download that.