Wrong Contact Header
Fernando Lopes via sr-users <[email protected]>
| Newsgroups | gmane.comp.voip.ser |
|---|---|
| Message-ID | <[email protected]> |
Hello everyone, I’m running into an issue with Kamailio (exposed to the public) talking to Asterisk. The problem only happens with UDP/TCP SIP (not with WebRTC), and it’s related to the 200 OK response from Asterisk. Asterisk is sending the 200 OK with its internal IP address in the Contact header, which Kamailio forwards to the client. As a result, the client tries to send the ACK directly to the internal IP, and it fails. Logs (WebRTC → Kamailio → Asterisk) || In sngrep, the connection shows as: 10.5.0.8:5060 (Kamailio) <-> 10.5.0.2:5060 (Asterisk) SIP/2.0 200 OK Via: SIP/2.0/UDP 172.31.217.74:5060;rport=5060;received=10.5.0.8;branch=z9hG4bK0186.15eb7eeb80260f12a3a4d45264371b16.0 Via: SIP/2.0/WSS 05gol16panaa.invalid;rport=65443;received=172.31.208.1;branch=z9hG4bK2464299 Call-ID: i0c08fra5hdplpdj4pq4 From: "916c67e8-652f-49ea-b288-2bbbee8dbf21" <sip:[email protected]>;tag=86ca2rhiqs To: <sip:[email protected]>;tag=b012b645-bf96-4fba-9166-f7a00b10290b CSeq: 2 INVITE Server: Asterisk PBX 18.24.3 Contact: <sip:10.5.0.2:5060> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 667 ACK sip:10.5.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 172.31.217.74:5060;branch=z9hG4bK0186.94970c4407431a3133bec46208b4be06.0 Via: SIP/2.0/WSS 05gol16panaa.invalid;rport=65443;received=172.31.208.1;branch=z9hG4bK3452625 To: <sip:[email protected]>;tag=b012b645-bf96-4fba-9166-f7a00b10290b From: "916c67e8-652f-49ea-b288-2bbbee8dbf21" <sip:[email protected]>;tag=86ca2rhiqs CSeq: 2 ACK Call-ID: i0c08fra5hdplpdj4pq4 Max-Forwards: 69 Supported: outbound User-Agent: SIP.js/0.21.1 Content-Length: 0 Logs (TCP call using MicroSIP → Kamailio → Asterisk) SIP/2.0 200 OK Via: SIP/2.0/UDP 172.31.217.74:5060;rport=5060;received=10.5.0.8;branch=z9hG4bK82df.7f15d0606c2de9f083b2757842de918f.0;i=1 Via: SIP/2.0/TCP 172.31.208.1:57670;rport=57670;received=172.31.208.1;branch=z9hG4bKPj379b3f941943461182af8d639cd93122;alias Call-ID: 5fbb1276c10e4041b9d5bc746bd48035 From: "6001" <sip:6001@localhost>;tag=009616b957db491cbea95fe3bfc123ee To: <sip:100@localhost>;tag=7345097c-95c0-42da-9b8e-1f39c5ec7f53 CSeq: 4449 INVITE Server: Asterisk PBX 18.24.3 Contact: <sip:10.5.0.2:5060> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800;refresher=uac Require: timer Content-Type: application/sdp Content-Length: 481 Any suggestions on how to fix this problem. I'm using a modified version of this: https://github.com/davidcsi/kamailio-private-public/blob/master/kamailio.cfg Thank you. __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- [email protected] To unsubscribe send an email to [email protected] Important: keep the mailing list in the recipients, do not reply only to the sender!