Hello experts,
Is there any configuration by which in INVITE message I can send Public IP address to provider.
As of now my SIP server is sending private IP in SDP which I guess is causing 2 way audio issue.
My host is on AWS and we have opened all the connection on AWS firewall setting.
Any suggestion?
Below is the log snippet. The call disconnects in 32 seconds always:
12:25:10.984: Sending [0,UDP] 1176 bytes to 54.172.60.1:5060 >>>>>
INVITE sip:+918605110351@vf-test.pstn.twilio.com SIP/2.0
From: ;tag=00948BE4-2518-1E2B-8BBA-392F1FACAA77-869
To:
Call-ID: 00948BD0-2518-1E2B-8BBA-392F1FACAA77-793@172.31.47.57
CSeq: 1 INVITE
Content-Length: 400
Content-Type: application/sdp
Via: SIP/2.0/UDP 172.31.47.57:5060;branch=z9hG4bK00948BEE-2518-1E2B-8BBA-392F1FACAA77-29
Contact:
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
User-Agent: 3CXPhone 6.0.26523.0
Max-Forwards: 69
X-Genesys-CallUUID: 0023ISH534F2N2TQ74NHVB5AES00000A
X-ISCC-CofId: location=SIP_Switch;cofid=16777236
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: uui,replaces,timer
v=0
o=3cxVCE 1579886013 1 IN IP4 172.31.27.129
s=3cxVCE Audio Call
c=IN IP4 172.31.27.129
t=0 0
m=audio 40006 RTP/AVP 0 8 3 101
a=ptime:20
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 40004 RTP/AVP 34
c=IN IP4 172.31.27.129
a=sendrecv
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
12:25:11.071: $+NET:SIP::0:0
12:25:11.071: SIPTR: Received [0,UDP] 406 bytes from 54.172.60.1:5060
