[Serusers] [Newbie question] - NAT/port messed up

[Thomas]

Hello!

I'm trying to make my 1st SIP-based system working. I use SER-0.8.14
on Linux Sarge. I have usually UAs behind NAT firewalls.

I found a case, when UAs are working very oddly, I can't call from one
of them, after sime time I can't reach one of them from outside, after
some time when I call one UA, the 2nd is answering (and the same time
when I call 2nd UA, the 2nd is answering).

I have 2 UAs [Grandstream BT100 & X-Lite v2.0] behind a linux NAT
firewall. Both UAs use the same STUN and SER server.
I did some tcpdumps and I found an interesting thing:

Internet Protocol, Src Addr: <FWPUBLICIP> (<FWPUBLICIP>), Dst Addr:  <SERIP> (<SERIP>)
User Datagram Protocol, Src Port: 1024 (1024), Dst Port: 5060 (5060)
Session Initiation Protocol
    Request-Line: REGISTER sip:<MYDOMAIN> SIP/2.0
        Method: REGISTER
        Resent Packet: True
        Suspected resend of frame: 31
    Message Header
        Via: SIP/2.0/UDP <FWPUBLICIP>;branch=z9hG4bK0dbeeb499dbd416e
        From: "M70" <sip:70@<MYDOMAIN>;user=phone>;tag=d4de3110a63ddb36
            SIP Display info: "M70" 
            SIP from address: sip:70@<MYDOMAIN>
            SIP tag: d4de3110a63ddb36
        To: <sip:70@<MYDOMAIN>;user=phone>
            SIP to address: sip:70@<MYDOMAIN>
        Contact: *
        Call-ID: 6ead5eb5744d407d@<IP PHONE LAN IP>
        CSeq: 100 REGISTER
        Expires: 0
        User-Agent: Grandstream BT100 1.0.5.16
        Max-Forwards: 70
        Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
        Content-Length: 0

Internet Protocol, Src Addr: <SERIP> (<SERIP>), Dst Addr: <FWPUBLICIP>77.150.18 (<FWPUBLICIP>)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 200 OK
        Status-Code: 200
        Resent Packet: False
    Message Header
        Via: SIP/2.0/UDP <FWPUBLICIP>;branch=z9hG4bK0dbeeb499dbd416e
        From: "M70" <sip:70@<MYDOMAIN>;user=phone>;tag=d4de3110a63ddb36
            SIP Display info: "M70"
            SIP from address: sip:70@<MYDOMAIN>
            SIP tag: d4de3110a63ddb36
        To: <sip:70@<MYDOMAIN>;user=phone>;tag=9ed85684aa54afbffef0a2d40b9935f4.71f6
            SIP to address: sip:70@<MYDOMAIN>
            SIP tag: 9ed85684aa54afbffef0a2d40b9935f4.71f6
        Call-ID: 6ead5eb5744d407d@<IP PHONE IP>
        CSeq: 100 REGISTER
        Server: Sip EXpress router (0.8.14 (i386/linux))
        Content-Length: 0
        Warning: 392 <SERIP>:5060 "Noisy feedback tells:  pid=25451 req_src_ip=<FWPUBLICIP> req_src_port=1024 in_uri=sip:<MYDOMAIN> out_uri=sip:<MYDOMAIN> via_cnt==1"

What is interesting is that REGISTER message went out from port 1024
but 200 OK came to port 5060. Is this OK? I think here is some problem
and this can make the phones working oddly.

SER is configured with nathelper+rtpproxy.

...
        if (nat_uac_test("3")) {
                # Allow RR-ed requests, as these may indicate that
                # a NAT-enabled proxy takes care of it; unless it is
                # a REGISTER

                if (method == "REGISTER" || ! search("^Record-Route:")) {
                    log("LOG: Someone trying to register from private IP, rewriting\n");

                    # This will work only for user agents that support symmetric
                    # communication. We tested quite many of them and majority is
                    # smart enough to be symmetric. In some phones it takes a configuration
                    # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is
                    # called "symmetric media" and "symmetric signalling".

                    fix_nated_contact(); # Rewrite contact with source IP of signalling
                    if (method == "INVITE") {
                        fix_nated_sdp("1"); # Add direction=active to SDP
                    };
                    force_rport(); # Add rport parameter to topmost Via
                    setflag(6);    # Mark as NATed
                };
        };
...

It is basically the sample nathelper configuration.

Could anybody give me a hint? If it is a well-known situation, please
point me to some other resources, I'm really a new-comer in SIP/SER.

Thank you in advance!

Kind regard,
        Tamas J.