Thanks for the detailed trace, I wouldn't have spent on this without it ;-)

Its' the ACK for the OK that is not done fix_nated_contact for. In fact, checking the getting started rtpproxy file, it seems that this error is also found there. All messages coming from behind NAT should have the contact fixed. Strictly speaking, RFC3261 states that the dialog state should be created from the information found in the INVITE, not the ACK on an OK. Your BYE generating entity does this error. I have seen this problem before, but has not really thought about it from the getting started config file perspective. It is thus safer to change Contact on ALL messages behind NAT, regardless of they are loose routed or not. g-)

Fabio Macchi wrote:

Hi Greger,

I attached an ethereal SIP call trace of a test call ( summary and detailed, I simple maskerade final ip numbers ): below only the INVITE relayed from proxy to gateway:

Session Initiation Protocol

Request-Line: INVITE sip:9999001234@194.244.gatewayIP:5060 SIP/2.0

    Method: INVITE

    [Resent Packet: False]

Message Header

    Record-Route: <sip:194.244.Proxy__IP;ftag=12e1e2e19d527792;lr=on>

    Via: SIP/2.0/UDP 194.244.Proxy__IP;branch=z9hG4bKb24f.5133a2c4.0

        Transport: UDP

        Sent-by Address: 194.244.Proxy__IP

        Branch: z9hG4bKb24f.5133a2c4.0

    Via: SIP/2.0/UDP 

1.255.ua_priv__IP;rport=1176;received=213.156.ua_pub_IP;branch=z9hG4bK35ca7adf63e3094f

        Transport: UDP

        Sent-by Address: 1.255.ua_priv__IP

        RPort: 1176

        Received: 213.156.ua_pub_IP

        Branch: z9hG4bK35ca7adf63e3094f

    From: "000003" <sip:000003@194.244.Proxy__IP>;tag=12e1e2e19d527792

        SIP Display info: "000003"

        SIP from address: sip:000003@194.244.Proxy__IP

        SIP tag: 12e1e2e19d527792

    To: <sip:9999001234@194.244.Proxy__IP>

        SIP to address: sip:9999001234@194.244.Proxy__IP

    Contact: <sip:000003@213.156.ua_pub_IP:1176>

        Contact Binding: <sip:000003@213.156.ua_pub_IP:1176>

            URI: <sip:000003@213.156.ua_pub_IP:1176>

                SIP contact address: sip:000003@213.156.ua_pub_IP:1176

    Supported: replaces

    Call-ID: 953e8996cfcc4ccc@1.255.ua_priv__IP

    CSeq: 60577 INVITE

        Sequence Number: 60577

        Method: INVITE

    User-Agent: Grandstream HT386 1.0.3.64 FXS0

    Max-Forwards: 16

    Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE

    Content-Type: application/sdp

    Content-Length: 327

Message body

As you can see, contact informations are correctly fixed with the pubblic UA address, but when the callee hungs up, the BYE is relayed to the private IP address: am I missing something ?

In this invite I see UA private address only in VIA: does BYE look to this parameter ?

Later caller hangs up too, and the OK is relayed to the correct IP/port.

Any help would be high appreciate, thanks in advance.

Fabio