We receive an invite from 192.168.0.132 with a public IP in the SDP, I'm going to assume thats via OpenSIPS? 192.168.0.173 is FS, 

Try turning aggressive nat detection off, you should NEVER need that except in some weird extreme cases.

/b



On Thu, Apr 20, 2017 at 8:08 AM, Andrew Paul <andrew.paul85@gmail.com> wrote:
Hi all,

we have FREESWITCH behind opensips and freeswitch is running in AWS. Both Opensips and Freeswitch is in same network + Softphone SDP have public IP address. 

SOFTPHONE --> OPENSIPS --> ( profile : internal)  FREESWITCH  ( external) --> CARRIER --> CUSTOMER 

Freeswitch is action as mediaserver here , I can see the extern-rtp-ip is setting on the calls to CARRIER, But when freeswitch 200Ok to Opensips, it is not putting external IP. It is setting the freeswitch local ip. 

I have followed the instruction in https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 , but not sure why FS is not setting external IP for internal profile. 

Is there any way to force the EIP ( public IP ) to SDP to opensips?. I can see the Contact address have Public IP but not the SDP Connection IP. 

Internal.xml

  <param name="inbound-zrtp-passthru" value="false"/>
    <param name="nonce-ttl" value="60"/>
    <param name="disable-register" value="true"/>
    <param name="auth-calls" value="$${internal_auth_calls}"/>
    <param name="inbound-reg-force-matching-username" value="true"/>
    <param name="auth-all-packets" value="false"/>

    <param name="sip-ip" value="192.168.0.173"/>
    <param name="rtp-ip" value="192.168.0.173"/>
    <param name="ext-sip-ip" value="54.81.198.142"/>
    <param name="ext-rtp-ip" value="54.81.198.142"/>
    <param name="aggressive-nat-detection" value="true"/>


send 1379 bytes to udp/[192.168.0.132]:5060 at 08:53:33.106380:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.0.132:5060;branch=z9hG4bKe04b.c7469955.0;rport=5060
   Via: SIP/2.0/UDP 182.75.26.194:58618;rport=58618;received=182.75.26.194;branch=z9hG4bK-524287-1---8cfb5fbe8c5d0f23
   Record-Route: <sip:192.168.0.132;r2=on;lr;ftag=7325490b;vsf=AAAAAH1uFG18am1hcXBAIRUIFgwREU0YRxoMAAoOQRtBDlQZcmFuc3BvcnQ9VURQ;did=ad3.3f713b01>
   Record-Route: <sip:54.175.200.193:5060;r2=on;lr;ftag=7325490b;vsf=AAAAAH1uFG18am1hcXBAIRUIFgwREU0YRxoMAAoOQRtBDlQZcmFuc3BvcnQ9VURQ;did=ad3.3f713b01>
   To: <sip:14154839556@calldirect.piba.com;transport=UDP>;tag=7Ury0X9HXp68j
   Call-ID: 7sgGUmlws7Bg64kCbksywQ..
   CSeq: 2 INVITE
   Contact: <sip:14154839556@54.81.198.142:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
   Supported: path, replaces
   Allow-Events: talk, hold, conference, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 222
   Remote-Party-ID: "Outbound Call" <sip:14154839556@calldirect.piba.com>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1492664761 1492664762 IN IP4 192.168.0.173
   s=FreeSWITCH
   c=IN IP4 192.168.0.173
   t=0 0
   m=audio 28052 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16


Attaching the logs for reference. 

Regards

Andrew 


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