Thanks Brian for the quick response. Yes 192.168.0.132 is opensips and FS is 192.168.0.173. The EIP of FS instance is  54.81.198.142. I turned off the aggressive NAT to false , but unfortunately same result.

 Do we have any settings for forcing the SDP to extern-rtp-ip ?. Just wondering how it worked for the contact address but not the SDP. 

freeswitch@testenv100.aws.piba.net> sofia status profile internal  

=================================================================================================
Name                    internal
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_internal
Pres Hosts              192.168.0.173,192.168.0.173
Dialplan                XML
Context                 lvcall_bridge
Challenge Realm         auto_from
RTP-IP                  192.168.0.173
Ext-RTP-IP              54.81.198.142
SIP-IP                  192.168.0.173
Ext-SIP-IP              54.81.198.142
URL                     sip:mod_sofia@54.81.198.142:5060
BIND-URL                sip:mod_sofia@54.81.198.142:5060;maddr=192.168.0.173;transport=udp,tcp
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               PCMU
CODECS OUT              PCMU
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     0
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                true
PROXY-MEDIA             false
ZRTP-PASSTHRU           true
AGGRESSIVENAT           false
CALLS-IN                1
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           0

Regards

Andrew


On Thu, Apr 20, 2017 at 7:17 PM, Brian West <brian@freeswitch.org> wrote:
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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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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