Re: Wrong Contact Header
Fred Posner via sr-users <[email protected]>
| Newsgroups | gmane.comp.voip.ser |
|---|---|
| Message-ID | <[email protected]> |
On Sep 16, 2025, at 5:45 PM, Fernando Lopes via sr-users <[email protected]> wrote: > > 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. Your example shows wss. Regardless, please see the nathelper module and set_contact_alias: https://www.kamailio.org/docs/modules/stable/modules/nathelper.html#nathelper.set_contact_alias The default kamailio config (also available here: https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg) has a NATDETECT route with an example. Fred Posner Tel: +1 (352) 664-3733 Alt: +1 (224) 334-3733 Contact info at https://fredp.xyz __________________________________________________________ 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!