SIP one way audio issues

I have srcnat action=same , also have tried masquerade, with and without SIP helper app, 3.10-3.11

I have an OpenSER box that a Polycom(multiple actually) registers to. This works fine. The OpenSER talks to an Asterisk box on another (Also public) IP. I am getting some strange one way audio issues.

I am seeing alot of unreplied UDP connections - basically for some reason the tik is not allowing the UDP in from the Asterisk box in…

Hopefully this will make sense…

192.168.254.254 is the phone behind the tik
x.x.x.x is the public IP of the Asterisk server
: public IP of the Mikrotik

unreplied entry in connection track appears
udp 192.168.254.254:2226 x.x.x.x:18330 19s


WAN port torch shows
source address x.x.x.x:18330 udp going to :1138
and also
source address x.x.x.x:18330 udp going to :2226 (which is not being NATed to the phone since it’s showing up unreplied)

the RTP audio should be going to :2226 and being passed to 192.168.254.254:2226

but it’s not allowing it through - as the unreplied in connection track shows

why is this?

here’s when I log it, it’s just coming into the tika nd getting lost
23:34:24 firewall,info input: in:ether1 out:(none), src-mac 00:60:e0:44:2b:e4, proto UDP, x.x.x.x:18330->:2226, len 180
23:34:24 firewall,info input: in:ether1 out:(none), src-mac 00:60:e0:44:2b:e4, proto UDP, x.x.x.x:18330->:2226, len 180
this repeats as long as the call is up

A simple search turns this up.
http://forum.mikrotik.com/t/sip-one-way-audio-how-to-enable-ports-on-nat-firewall/14925/1

-Louis

yes I have read that and I have disabled SIP helper. It’s not reasonable to do 1:1 NATs as there may be multiple phones behind the Tik.

you have to nat 1:1, or nat a separate port and rtp range for each phone.

phone 1: 5060, 10000-10199
phone 2: 5062, 10200-10299
phone 3: 5064, 10400-10399
etc.

SIP encodes the private IP inside of the payload of the control packet. NAT doesnt change that.

NOTE: This is only if the phones are talking directly to the outside world.

Sam