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37465 discussions

How to place a delay between cancelling branch and retrying next one on timer expiry?
by Alex A 15 Apr '25

15 Apr '25

Hi All, I have an interesting use-case where I need to place a delay between cancelling branch and retrying next one. *Background: * Running Kamailio 5.6.x Kamailio routes requests via a freeswitch B2BUA server on select call flows (where B2BUA functionality is required). Call flow examples: PSTN --> Kamailio --> Freeswitch --> appserver1 PSTN --> Kamailio --> Freeswitch --> appserver2 On attempt failures, a branch retry is required from appserver1 to appserver2; however … [View More]freeswitch B2BUA remains the same Since Freeswitch does not quite follow the RFC 3261 timer K specifications (should be 0 seconds for reliable transports, but FS keeps transactions for T4 (set to 2 seconds)); I have implemented an artificial delay in failure route before retrying (as per recommendation described here: https://freeswitch-users.freeswitch.narkive.com/TsSye66D/482-request-merged… ) *Script config: * onreply_route[MANAGE_REPLY] { xlog("L_INFO", "reply $T_reply_code received"); if (t_check_status( "1[1-9][0-9]")) { t_reset_fr(); } else { t_set_fr(120000); } } route(NATDETECT); #!ifdef WITH_RTPENGINE route(RTPMANAGE); #!endif } route[RELAY] { if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { setflag(FLT_DLGINFO); } if (!t_relay()) { sl_reply_error(); } exit; } route[TOCARRIER] { process bunch of things, attach custom headers, generate request URI t_on_failure("MANAGE_FAILURE_CARRIER"); route(RELAY); } failure_route[MANAGE_FAILURE_CARRIER] { xlog("L_INFO", "reply $T_reply_code received"); if ($avp(inbound_attempt)<=1){ if ($dlg_var(b2bua)=="true"){ usleep(1900000); } route("TOCARRIER"); } *Call Flows: * The above fix works well when an explicit error reply is received back. *error reply SIP exchange: * SBC --> INVITE(branch0) ---> Freeswitch SBC <-- 503 <-- Freeswitch failure_route executed (sleep runs here) SBC --> INVITE(branch1) ---> Freeswitch call continues as expected. however does not work when NO reply is received at all and the timer expires. *Current timer expiry SIP exchange: * SBC --> INVITE(branch0) ---> Freeswitch SBC <-- 100/180 <-- Freeswitch SBC (timer expires) failure_route executed (sleep runs here) SBC --> INVITE(branch1) ---> Freeswitch SBC <-- 482 Merged <-- Freeswitch SBC --> CANCEL(branch0) ---> Freeswitch I am looking to CANCEL branch0 before the sleep delay runs Based on source code, t_cancel_branches("this") can only be executed in reply_route; however the chellenge is that reply_route never sees the "faked 408" on timer expiry(although the failure route does see $T_reply_code 408). Any suggestions on how to handle this would be greatly appreciated. TIA [View Less]

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RTPEngine pool selection for in-dialog requests
by Ben Kaufman 15 Apr '25

15 Apr '25

I'm looking for information on the correct way to handle calls to RTPEngine when processing in-dialog requests. I currently have several setid's configured for varying usages. During processing of an initial INVITE, a determination is made as to if the call will require the use of RTPEngine, and if so, which of the sets should be used. The following is done in code: * setid is assigned to AVP $avp(rtpengine_pool_num) * set_rtpengine_set("$avp(rtpengine_pool_num)") is called, * … [View More]rtpengine_offer is then called. * Additionally, a record-route parameter of proxy_media=yes is added. * The flag PROXY_MEDIA is set In the reply route, the PROXY_MEDIA flag is checked, and if set, set_rtpengine_set("$avp(rtpengine_pool_num)") is called, followed by rtpengine_answer. When handling an in-dialog message my code didn't have anything to evaluate for the setid. rtpengine_offer was just called. Looking at the debug from the rtpengine module, I can see that the record is found in the hashtable, however the node indicated in that result is not necessarily used - rtpengine will select an entry from the default setid instead. From the rtpengine module logs, IP addresses redacted: 2(9) DEBUG: {time=1744651756.968228} rtpengine [rtpengine_funcs.c:139]: check_content_type(): type <application/sdp> found valid 2(9) DEBUG: {time=1744651756.968228} rtpengine [rtpengine.c:2249]: build_rtpp_socks(): same rtpengines list version: 1 (1744651613) 2(9) DEBUG: {time=1744651756.968228} rtpengine [rtpengine.c:3818]: select_rtpp_node_old(): rtpengine hash table lookup find node=udp:A.B.C.D:1234 for calllen=36 callid=ac66f56b-93f8-123e-1b99-0242ac120002 viabranch= 2(9) DEBUG: {time=1744651756.968228} rtpengine [rtpengine.c:3702]: select_rtpp_node_new(): sum is = 210 2(9) DEBUG: {time=1744651756.968228} rtpengine [rtpengine.c:3198]: rtpp_function_call(): proxy reply: d3:sdp288:v=0 o=FreeSWITCH 1744611570 1744611571 IN IP4 W.X.Y.Z s=FreeSWITCH c=IN IP4 W.X.Y.Z My question is really: **SHOULD** rtpengine_offer (and rtpengine_manage, etc) use the node that it finds in the hash table lookup automatically, or should I have to explicitly set the setid? For now I've solved the problem by adding the setid to the record-route headers so I can easily retrieve it in dialog, but it seems that if the module finds a record for a node that there should be some way to rely on that explicitly. Kaufman Senior Voice Engineer E: bkaufman(a)bcmone.com SIP.US Client Support: 800.566.9810 | SIPTRUNK Client Support: 800.250.6510 | Flowroute Client Support: 855.356.9768 [img]<https://www.sip.us> [img]<https://www.siptrunk.com> [img]<https://www.flowroute.com> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else. [View Less]

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Thank you Guys
by Fernando Lopes 15 Apr '25

15 Apr '25

Just want to say thank you for the support. I spent two weeks working on a config, and even though it didn’t work out, I really appreciate the learning process. Today, I found a config that worked here: 🔗 https://github.com/altanai/kamailioexamples/tree/master/RTPEngine_media_pro…

1 0

Nat Problems Kamailio
by Fernando Lopes 14 Apr '25

14 Apr '25

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 … [View More](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. 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TCP timeout
by Pyry Aaltonen 11 Apr '25

11 Apr '25

Hello, I´ve googled around about tcp timeout in kamailio tcp connections when the tcp connection is broken. Found this old question and to me it feels like I’m having that same situation as original question https://www.mail-archive.com/sr-users@lists.kamailio.org/msg15020.html Also found this https://kamailio.org/mailman3/hyperkitty/list/sr-dev@lists.kamailio.org/thr… and it seems to be the same description pretty much that I´m seeing. When the tcp connection (or tls) is interrupted in the … [View More]

5 6

leave
by Mihail D. 11 Apr '25

11 Apr '25

leave

1 0

STUN Server vs. NAT Configuration with Kamailio + RTPengine Setup
by Fernando Lopes 11 Apr '25

11 Apr '25

Previous Thread: https://kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio.org/t… Hello everyone, Should I use a STUN server, or can I just configure NAT properly? Here’s my current setup: arduino Copy Edit Router (NAT device) ↓ Kamailio + RTPengine machine ↓ Virtual machine running Kubernetes ↓ Asterisk container Kamailio and RTPengine are sitting behind the NAT (the router), and they handle SIP and media. Do I need a STUN server in this case, or can I … [View More]

2 1

IPv6 bind wrong IP. #4199
by vunm.allsign＠gmail.com 11 Apr '25

11 Apr '25

Hi guys, I opened issue on github: https://github.com/kamailio/kamailio/issues/4199 - Description: Kamailio (5.8.x): SIP proxy FreeSWITCH: Media - OS: Debian 12 gateway: aaaa:bbb:cccc:20::1 ipv6: aaaa:bbb:cccc:20::41e - kamailio.cfg: #!define ADDR_IPV4 xxx.xxx.xxx.xxx #!define ADDR_IPV6 [aaaa:bbb:cccc:20::41e] listen=ADDR_IPV4 listen=ADDR_IPV6 - I execute netstat -nlutp: tcp6 0 0 aaaa:bbb:cccc:20:::5060 :::* LISTEN 1691274/kamailio tcp6 0 … [View More]

2 1

Load multiple geoip2 databases (City + ASN)
by Joel Serrano 10 Apr '25

10 Apr '25

Hi guys, Is there a way to have multiple geoip2 databases loaded into Kamailio and then point the geoip2_match to them? If I understand correctly the docs I can only load one: https://www.kamailio.org/docs/modules/6.0.x/modules/geoip2.html#geoip2.p.pa… Is my assumption correct? Thanks, Joel.

3 5

Kamailio + RTP Engine question
by fernandolopes20003＠gmail.com 08 Apr '25

08 Apr '25

Hi everyone, I'm experiencing some issues with Kamailio and RTP Engine. Here's the setup: A user initiates a call, and the traffic goes through a router. The router forwards the traffic to a machine at 192.168.1.75, which runs Kamailio and RTP Engine. Kamailio uses a dispatcher to forward calls to 192.168.1.190, a Kubernetes node running an Asterisk container. The problem: When I'm inside the network, everything works fine — I can send and receive audio. But when I'm outside the network, the … [View More]

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