Hello,
I hope somebody could help me.
This is my scheme. User-agent is behind NAT1. Kamailio and pbx are behind NAT2 [Client ip-phone 192.168.89.213 without stun] - LAN1 - NAT1(46.0.0.30) - (internet) - (51.0.0.60)NAT2 - Local2 - (10.130.0.23:5060)kamailio( 10.130.0.23:5070) - pbx
*Questions*: *1*. The VIA header (with 10.130.0.23 and 51.0.0.60) wasn't included in SIP packets. Why? For example 200OK reply. It came from pbx through kamailio. Which setting could break it?
Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.89.213:5060 ;rport=9570;received=46.0.0.30;branch=z9hG4bK2480172053 Record-Route: sip:c172.19.19.111.8348.call.cgatepro;lr Record-Route: sip:172.19.19.111:5060;lr Record-Route: sip:10.130.0.23:5070 ;r2=on;lr;ftag=4268683942;nat=yes,sip:51.0.0.60;r2=on;lr;ftag=4268683942;nat=yes From: "Aleksey" sip:a.yakimkin@mail.domain.ru:5060;tag=4268683942 To: <sip:2961@mail.domain.ru:5060
;tag=A03E2397-404246-FA7543E4_jizmelr-582D
Call-ID: 6_1903330087@192.168.89.213 [Generated Call-ID: 6_1903330087@192.168.89.213] CSeq: 2 INVITE Contact: sip:signode-404246-FA7543E4_jizmelr-582D@172.19.19.111 ;alias=51.0.0.60~5060~1 Supported: 100rel,timer,replaces,histinfo,precondition Allow: INVITE,BYE,CANCEL,ACK,OPTIONS,INFO,MESSAGE,PRACK,UPDATE,REFER Session-Expires: 1800;refresher=uas Content-Type: application/sdp Content-Length: 1170 Message Body
*2*. About Registrar, Path and $du. Phone set Register with headers: Via: SIP/2.0/UDP 192.168.89.213:5060;branch=z9hG4bK2691182696 From: "Aleksey" sip:a.yakimkin@mail.domain.ru:5060;tag=3926879477 To: "Aleksey" sip:a.yakimkin@mail.domain.ru:5060 Contact: sip:a.yakimkin@192.168.89.213:5060 Kamailio respond Via: SIP/2.0/UDP 192.168.89.213:5060 ;rport=9570;received=46.0.0.30;branch=z9hG4bK617463686 Path: sip:10.130.0.23:5070 ;lr;received=46.0.0.30~9570~1;r2=on,sip:51.0.0.60;lr;received=46.0.0.30~9570~1;r2=on From: "Aleksey" sip:a.yakimkin@mail.domain.ru:5060;tag=3926879477 To: "Aleksey" sip:a.yakimkin@mail.domain.ru:5060;tag=194ED16D Contact: sip:a.yakimkin@192.168.89.213:5060;expires=360 Contact: sip:2447@192.168.89.221:5060;expires=247 Contact: sip:2447@192.168.9.16:5060;expires=2116
I try to make a call from pbx to ip phone. pbx inserts in Invite header Route: sip:10.130.0.23:5070 ;lr;received=46.0.0.30~9570~1;r2=on But kamailio relayed Invite direct to 192.168.89.213. (There is network connectivity between ip-phone and kamailio through vpn). The code below helps me to solve my issue. I saw mail-list with similar trouble. But no setting could get kamailio to relay Invite to "Route-received" ip.
$var(the_route) = $hdr(Route); $var(route0) = $(var(the_route){s.select,0,,}); $var(new_host) = $(var(route0){param.value,received}{s.select,0,~}); $var(new_port) = $(var(route0){param.value,received}{s.select,1,~}); if (!strempty($var(new_host)) && !strempty($var(new_port)) ) { $du = "sip:" + $var(new_host) + ":" + $var(new_port); }
*Kamailio settings:* I have such listeners listen=udp:10.130.0.23:5070 # to local network listen=udp:10.130.0.23:5060 advertise 51.0.0.60:5060 # to internet
#MODULE SETTING #--- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock")
modparam("path", "use_received", 1) modparam("path", "enable_r2", 1) modparam("path", "received_format", 1)
# ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")
# ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "auto_inv_100_reason", "Trying")
# ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 2) modparam("rr", "force_send_socket", 1)
# ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0)
modparam("registrar", "use_path", 1) modparam("registrar", "path_use_received", 1) modparam("registrar", "path_mode", 0) #---
For register I use this code route[REGISTRAR] { ... add_path_received(); set_send_socket("udp:10.130.0.23:5070"); route(DISPATCH); ... } route[RELAY] { ... if ($Ru eq "sip:10.130.0.23:5070") { $fs = "udp:10.130.0.23:5060"; } else { $fs = "udp:10.130.0.23:5070"; } ... } route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;
#send INVITE to IP:PROT from Route:...;received= $var(the_route) = $hdr(Route); $var(route0) = $(var(the_route){s.select,0,,}); $var(new_host) = $(var(route0){param.value,received}{s.select,0,~}); $var(new_port) = $(var(route0){param.value,received}{s.select,1,~}); if (!strempty($var(new_host)) && !strempty($var(new_port)) ) { $du = "sip:" + $var(new_host) + ":" + $var(new_port); } if (client_nat_test("3")) {
if(nat_uac_test("18")) { if ($Ru == "sip:10.130.0.23:5070") { rtpproxy_manage("co", "51.0.0.60"); # fix_nated_sdp } else { rtpproxy_manage("co"); }
if (is_method("REGISTER")) { #if ($Ru == "sip:10.130.0.23:5070") { # fix_nated_contact(); #} else { # set_contact_alias(); #} set_contact_alias(); } else { if(is_first_hop()) { set_contact_alias(); } else { add_contact_alias("51.0.0.60", "5060", "udp"); #fix_nated_contact(); } } } else { if ($Ru == "sip:10.130.0.23:5070") { rtpproxy_manage("cor", "51.0.0.60"); # fix_nated_sdp } else { rtpproxy_manage("cor"); } } } else { rtpproxy_manage("co"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); #fix_contact(); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); #} else { #fix_contact(); #} } } #!endif return; }
Thank you.
Can't tell what exactly is wrong with the whole logic of your script, most likely a couple of things, without spending an entire day troubleshooting it. Instead, can highlight a few things that may help:
1. do yourself a big favor and put Kamailio on a public IP or dst_nat SIP to it, unless you plan on running a VoIP business off a home internet or something 2. for REGISTER, if you final Registrar supports *PATH*, you don't need to also *set_contact_alias()* on top of *add_path_received()*. You either do one approach or the other. 3. Invites and others originating from the PBX, having the preset *Route*, must simply be handled by *loose_route() *in combination with *modparam("path", "use_received", 1)*. All that convoluted *$du* manipulation that you are doing trying to compile it from the Route header is totally unnecessary.
Keep trying, you aren't far from figuring it out.
Regards, --Sergiu
On Fri, Jul 8, 2022 at 10:30 AM Алексей Якимкин ayakimkin@gmail.com wrote:
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Sergiu Pojoga -
Алексей Якимкин