Re: R: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem - sr-users

20 Mar 2007

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(a)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