Hello Everyone, I'm having some trouble getting my SIP setup to work properly.
📞 My Call Flow: SIP Client initiates a call The call reaches my router, which forwards it to Kamailio + RTPengine running on 192.168.1.75:5060, which then forwards it to A Kubernetes machine at 192.168.1.190:32210, where Asterisk is running
🔧 Network Setup: Router has the following ports forwarded: UDP 5060 UDP 10000–20000 (for RTP) STUN is enabled on the SIP client
✅ What Works: When I’m connected through a VPN (i.e., inside the local network), everything works fine — both incoming and outgoing audio.
❌ The Problem: When calling from outside the network, signaling works and the call connects, but I can’t receive or send any audio.
Here is a little of my config of kamailio modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") modparam("rtpengine", "setid_avp", "$avp(setid)") listen=udp:192.168.1.75:5060 advertise 62.2x.xxx.xxx:5060 listen=tcp:192.168.1.75:5060 advertise 62.2x.xxx.xxx:5060 listen=udp:127.0.0.1:5060 #Nat Helper modparam("nathelper", "force_socket", "192.168.1.75:5060") modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "nat_addr_mode", 1)
request_route{ ... route(init); ... if(is_method("INVITE|ACK|BYE|CANCEL|OPTIONS|INFO|UPDATE|PRACK")){ route(handle_sip); exit; } } route[init]{ ... if (nat_uac_test("19")){ setflag(FLT_NAT); fix_nated_contact(); force_rport(); } } route[handle_sip]{ if ($rm == "INVITE"){ if (isflagset(FLT_NAT)){ route(natmanage); }
if (!ds_select_dst("1", "4")){ sl_send_reply("500", "No destination"); exit; }
$ru = "sip:" + $rU + "@" + $rd; # Keeps sip:100@domain $du = "sip:192.168.1.190:32210"; # Matches your dispatcher list
route(relay); }
if ($rm == "BYE" || $rm == "CANCEL"){ if (isflagset(FLT_NAT)){ rtpengine_delete(); }
route(relay); }
if ($rm == "ACK"){ if (isflagset(FLT_NAT)){ route(natmanage); }
route(relay); } route(relay); } route[natmanage]{ if (is_request()){ if (has_body("application/sdp")){ rtpengine_offer("replace-origin replace-session-connection ICE=remove"); } force_rport(); fix_nated_contact(); }else{ rtpengine_answer("replace-origin replace-session-connection ICE=remove"); } } rtpengine.conf
[rtpengine] table = 0 interface = internal/192.168.1.75 listen-ng = 127.0.0.1:2223 log-level = 7
timeout = 60 silent-timeout = 3600 tos = 184
port-min = 10000 port-max = 40000
And here are some logs when I call from outside
2025/04/13 15:58:33.407265 2.8x.xx.xxx:59396 -> 192.168.1.75:5060 INVITE sip:100@192.168.1.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:59396;rport;branch=z9hG4bKPj92490d6b6d0c4cd88dac9f0043b1642e Max-Forwards: 70 From: "6001" sip:6005@192.168.1.75:5060;tag=0397d2bdc3ef47f5b6e498126a360b0f To: sip:100@192.168.1.75:5060 Contact: "6001" sip:6005@192.168.1.116:59396;ob;+sip.ice Call-ID: 83a804569c3f489088607c729485cdbd CSeq: 8963 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.21.5 Content-Type: application/sdp Content-Length: 537
v=0 o=- 3953552313 3953552313 IN IP4 192.168.1.116 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4032 RTP/AVP 8 0 101 c=IN IP4 192.168.1.116 b=TIAS:64000 a=rtcp:4017 IN IP4 192.168.1.116 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1567952351 cname:73284f1717045a81 a=ice-ufrag:37e51dc0 a=ice-pwd:49f7442b5078148140877b44 a=candidate:Hc0a80174 1 UDP 2130706431 192.168.1.116 4032 typ host a=candidate:Hc0a80174 2 UDP 2130706430 192.168.1.116 4017 typ host
2025/04/13 15:58:33.412683 192.168.1.75:5060 -> 192.168.1.190:32210 INVITE sip:100@192.168.1.75 SIP/2.0 Via: SIP/2.0/UDP 62.2x.2xx.xxx:5060;branch=z9hG4bK6362.26a75297a4ccf4984dd1f68644c2148f.0 Via: SIP/2.0/UDP 192.168.1.116:59396;received=2.8x.xx.xxx;rport=59396;branch=z9hG4bKPj92490d6b6d0c4cd88dac9f0043b1642e Max-Forwards: 69 From: "6001" sip:6005@192.168.1.75:5060;tag=0397d2bdc3ef47f5b6e498126a360b0f To: sip:100@192.168.1.75:5060 Contact: "6001" sip:6005@2.8x.xx.xxx:59396;ob;+sip.ice Call-ID: 83a804569c3f489088607c729485cdbd CSeq: 8963 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.21.5 Content-Type: application/sdp Content-Length: 322
v=0 o=- 3953552313 3953552313 IN IP4 192.168.1.75 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 12156 RTP/AVP 8 0 101 c=IN IP4 192.168.1.75 b=TIAS:64000 a=ssrc:1567952351 cname:73284f1717045a81 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:12157
2025/04/13 15:58:33.428169 2.8x.xx.xxx:59396 -> 192.168.1.75:5060 INVITE sip:100@192.168.1.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:59396;rport;branch=z9hG4bKPj14c6f24b814e4d158014cc80b1fa5f8f Max-Forwards: 70 From: "6001" sip:6005@192.168.1.75:5060;tag=0397d2bdc3ef47f5b6e498126a360b0f To: sip:100@192.168.1.75:5060 Contact: "6001" sip:6005@192.168.1.116:59396;ob;+sip.ice Call-ID: 83a804569c3f489088607c729485cdbd CSeq: 8964 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.21.5 Authorization: Digest username="6005", realm="asterisk", nonce="1744559913/dab8894e3f31e038d7a969a6349a11c2", uri="sip:100@62.2x.2xx.xxx", response="2ce500d6a1583f05f6344bdf1f712fc7", algorithm=MD5, cnonce="808fc766135d40589dd68032bcaab5c6", opaque="2232aa9c42080f12", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 537
v=0 o=- 3953552313 3953552313 IN IP4 192.168.1.116 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4032 RTP/AVP 8 0 101 c=IN IP4 192.168.1.116 b=TIAS:64000 a=rtcp:4017 IN IP4 192.168.1.116 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1567952351 cname:73284f1717045a81 a=ice-ufrag:37e51dc0 a=ice-pwd:49f7442b5078148140877b44 a=candidate:Hc0a80174 1 UDP 2130706431 192.168.1.116 4032 typ host a=candidate:Hc0a80174 2 UDP 2130706430 192.168.1.116 4017 typ host
2025/04/13 15:58:33.430593 192.168.1.75:5060 -> 2.8x.xx.xxx:59396 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.1.116:59396;rport=59396;branch=z9hG4bKPj14c6f24b814e4d158014cc80b1fa5f8f;received=2.8x.xx.xxx From: "6001" sip:6005@192.168.1.75:5060;tag=0397d2bdc3ef47f5b6e498126a360b0f To: sip:100@192.168.1.75:5060 Call-ID: 83a804569c3f489088607c729485cdbd CSeq: 8964 INVITE Server: kamailio (5.7.4 (x86_64/linux)) Content-Length: 0
2025/04/13 15:58:33.461894 192.168.1.75:5060 -> 192.168.1.190:32210 INVITE sip:100@192.168.1.75 SIP/2.0 Via: SIP/2.0/UDP 62.2x.2xx.xxx:5060;branch=z9hG4bK3362.72f5db87deeeccee7fea924753af3da0.0 Via: SIP/2.0/UDP 192.168.1.116:59396;received=2.8x.xx.xxx;rport=59396;branch=z9hG4bKPj14c6f24b814e4d158014cc80b1fa5f8f Max-Forwards: 69 From: "6001" sip:6005@192.168.1.75:5060;tag=0397d2bdc3ef47f5b6e498126a360b0f To: sip:100@192.168.1.75:5060 Contact: "6001" sip:6005@2.8x.xx.xxx:59396;ob;+sip.ice Call-ID: 83a804569c3f489088607c729485cdbd CSeq: 8964 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.21.5 Authorization: Digest username="6005", realm="asterisk", nonce="1744559913/dab8894e3f31e038d7a969a6349a11c2", uri="sip:100@62.2x.2xx.xxx", response="2ce500d6a1583f05f6344bdf1f712fc7", algorithm=MD5, cnonce="808fc766135d40589dd68032bcaab5c6", opaque="2232aa9c42080f12", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 322
v=0 o=- 3953552313 3953552313 IN IP4 192.168.1.75 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 12156 RTP/AVP 8 0 101 c=IN IP4 192.168.1.75 b=TIAS:64000 a=ssrc:1567952351 cname:73284f1717045a81 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:12157
Please Help me, Thank you.