Hello!

I have the following architecture:

- Kamailio with RTPEngine (with public access on IP *YY.YY.42.207*) - Asterisk (with no public access, internal IP *172.31.69.198*)

I am using a Voip SIP (with public access on IP *54.XXX.XXX.44*) configured as a Trunk on Kamailio using UAC Module.

The issue I am facing is that the SIP messages replies I get from the VOIP provider are being destined to Kamailio's public IP (YY.YY.42.207) instead of Asterisk's IP (as sent on the message).

For example, this is the 200 message I sent to the VOIP provider (I clearly state that the contact is Asterisk sip:172.31.69.198:5080):

2022/04/07 14:17:19.260864 172.31.32.7:5060 -> 54.XXX.XXX.44:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 54.XXX.XXX.44:5060;rport=5060;branch=z9hG4bK90b6.f7a6b0a4.0 Record-Route: sip:YY.YY.42.207;lr;ftag=p0BjH21gpa3rc Call-ID: 464f21c3-3120-123b-91b2-026ef2fede4c From: "419XXXXX998" sip:38XXXX02@172.31.32.169;tag=p0BjH21gpa3rc To: sip:38XXXX02@54.XXX.XXX.44;tag=8c3bb09e-a909-4cb4-99a7-3c57aa8f946e CSeq: 50116727 INVITE Server: Asterisk PBX 18.11.0 *Contact: <sip:172.31.69.198:5080 http://172.31.69.198:5080>* Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 239

v=0 o=- 1649321756 1649321759 IN IP4 172.31.69.198 s=Asterisk c=IN IP4 172.31.69.198 t=0 0 m=audio 10010 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv

But the ACK I get from them is destined to Kamailio instead of Asterisk:

2022/04/07 14:17:19.263469 54.XXX.XXX.44:5060 -> 172.31.32.7:5060 *ACK sip:YY.YY.42.207:5060 SIP/2.0* Route: sip:YY.YY.42.207;lr;ftag=p0BjH21gpa3rc Via: SIP/2.0/UDP 54.XXX.XXX.44:5060;branch=z9hG4bK90b6.f7a6b0a4.2 Max-Forwards: 69 From: "419XXXXX998" sip:38XXXX02@172.31.32.169;tag=p0BjH21gpa3rc To: sip:38XXXX02@54.XXX.XXX.44;tag=8c3bb09e-a909-4cb4-99a7-3c57aa8f946e Call-ID: 464f21c3-3120-123b-91b2-026ef2fede4c CSeq: 50116727 ACK Contact: sip:voip.voip-sbc.6792.fab37901@54.XXX.XXX.44:5060 Content-Length: 0

This causes Kamailio to not know where to forward the message to... Asterisk never gets the reply.

This is happening to one of the SIP Trunks I am using, with the other, everything is fine.

Is there anything I can do to work around it (other than contacting the provider to fix on their end)?

Thanks!

ps: I have attached the full SIP messages trail to help.