It should already be doing that in that setup. On Thu, Apr 20, 2017 at 10:49 AM, Andrew Paul <andrew.paul85@gmail.com> wrote:
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/conf luence/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=z9hG 4bKe04b.c7469955.0;rport=5060 Via: SIP/2.0/UDP 182.75.26.194:58618;rport=5861 8;received=182.75.26.194;branch=z9hG4bK-524287-1---8cfb5fbe8c5d0f23 Record-Route: <sip:192.168.0.132;r2=on;lr;ft ag=7325490b;vsf=AAAAAH1uFG18am1hcXBAIRUIFgwREU0YRxoMAAoOQRtB DlQZcmFuc3BvcnQ9VURQ;did=ad3.3f713b01> Record-Route: <sip:54.175.200.193:5060;r2=on ;lr;ftag=7325490b;vsf=AAAAAH1uFG18am1hcXBAIRUIFgwREU0YRxoMAA oOQRtBDlQZcmFuc3BvcnQ9VURQ;did=ad3.3f713b01> From: "14154839556 <(415)%20483-9556>" <sip:14154839556@calldirect.pi ba.com>;tag=7325490b To: <sip:14154839556@calldirect.piba.com;transport=UDP>;tag=7Ury 0X9HXp68j 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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