Hi Gert,

I'm trying to use a fresh ASG 6.993 for SIP VoIP test.
To do that, I configured my ASG with a NAT masquerading rule and I enabled the "VoIP Security" -> "SIP":

SIP server networks: Any
SIP client networks: <SIP phone IP>

I'm not using a SIP server behind my ASG so I believe there is not necessary to define a DNAT rule...

However, when I call from my SIP-phone, I can establish a connection but there is no voice traffic.
Here a packet filter log line:

2007:01:19-16:29:36 (none) ulogd[2493]: id="2016" severity="info" sys="SecureNet" sub="packetfilter" name="SIP call RTP" action="SIP call RTP" fwrule="60018" initf="eth1" outitf="eth0" dstmac="00:02:b3:3a:5c:08" srcmac="00:17:c2:ef:c4:2d" srcip="192.168.60.5" dstip="x.y.z.w" proto="17" length="60" tos="0x00" prec="0xe0" ttl="126" srcport="20000" dstport="58786"

Here, 192.168.60.5 is the SIP-phone IP address and x.y.z.w is the public IP of the SIP proxy.
The RTP port on the phone is 20000.

Why doesn't it work?