Hello,

do not cross post to sr-dev and sr-users when the question is about kamailio.cfg and using kamailio.

If port is not added, then it is the default one for that transport, respectively 5060 for UDP/TCP and 5061 for TLS.

This is how the SIP specs say, is there any particular reason to have the port always there?

Cheers, Daniel

On 18.03.21 11:52, sudheermyworld@yahoo.com wrote:

Please help me with a way to send replies/forwards to include VIA header with port, because i can see that its just sending IP but not the port in VIA header.

my kamailio cfg file for route & route relay looks like below.

route[RELAY] {

# enable additional event routes for forwarded requests         # - serial forking, RTP relaying handling, a.s.o.         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")) {                 if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");         }         if (is_method("ACK|BYE")) {                 $duri= $ruri;                 t_relay();                 exit;         }

if (!t_relay()) {                 sl_reply_error();         }         exit; }

# Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD         # flood detection from same IP and traffic ban for a while         # be sure you exclude checking trusted peers, such as pstn gateways         # - local host excluded (e.g., loop to self)         if(src_ip!=myself) {                 if($sht(ipban=>$si)!=$null) {                         # ip is already blocked                         xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");                         exit;                 }                 if (!pike_check_req()) {                         xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n");                         $sht(ipban=>$si) = 1;                         exit;                 }         } #!endif         if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {                 # silent drop for scanners - uncomment next line if want to reply                 # sl_send_reply("200", "OK");                 exit;         }

if (!mf_process_maxfwd_header("10")) {                 sl_send_reply("483","Too Many Hops");                 exit;         }

if(is_method("OPTIONS") && uri==myself && $rU==$null) {                 sl_send_reply("200","Keepalive");                 exit;         }

if(!sanity_check("1511", "7")) {                 xlog("Malformed SIP message from $si:$sp\n");                 exit;         }         if (is_method("INVITE")) {                 send_reply("100", "Trying");         } }

# Handle requests within SIP dialogs route[WITHINDLG] {         if (!has_totag()) return;

# sequential request withing a dialog should         # take the path determined by record-routing         if (loose_route()) {                 route(DLGURI);                 if (is_method("BYE")) {                         setflag(FLT_ACC); # do accounting ...                         setflag(FLT_ACCFAILED); # ... even if the transaction fails                 } else if ( is_method("ACK") ) {                         # ACK is forwarded statelessly                         route(NATMANAGE);                 } else if ( is_method("NOTIFY") ) {                         # Add Record-Route for in-dialog NOTIFY as per RFC 6665.                         record_route();                 }                 route(RELAY);                 exit;         }

if (is_method("SUBSCRIBE") && uri == myself) {                 # in-dialog subscribe requests                 route(PRESENCE);                 exit;         }         if ( is_method("ACK|BYE") ) {                 #xlog("started ACK if else condition line 622 \n");                 if ( t_check_trans() ) {                         # no loose-route, but stateful ACK;                         # must be an ACK after a 487                         # or e.g. 404 from upstream server                         route(RELAY);                         exit;                 } else {                         route(RELAY);                         # Added after observing ACK not being sent by CSCF                         exit;                 }         }         sl_send_reply("404","Not here");         exit; }

# Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT         force_rport();         if (nat_uac_test("19")) {                 if (is_method("REGISTER")) {                         fix_nated_register();                 } else {                         if(is_first_hop()) {                                 set_contact_alias();                         }                 }                 setflag(FLT_NATS);         } #!endif         return; }

# RTPProxy control and signaling updates for NAT traversal 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;

if(nat_uac_test("8")) {                 rtpproxy_manage("co");         } else {                 rtpproxy_manage("cor");         }

if (is_request()) {                 if (!has_totag()) {                         if(t_is_branch_route()) {                                 add_rr_param(";nat=yes");                         }                 }         }         if (is_reply()) {                 if(isbflagset(FLB_NATB)) {                         if(is_first_hop())                                 set_contact_alias();                 }         } #!endif         return; }

# URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT         if(!isdsturiset()) {                 handle_ruri_alias();         } #!endif         return; }

# Routing to foreign domains route[SIPOUT] {         if (uri==myself) return;

append_hf("P-hint: outbound\r\n");         route(RELAY);         exit; }

# PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN         $var(uri) = $sel(ruri);         xavp_params_explode("$(var(uri){s.unbracket})", "uri"); $var(cic) = $xavp(uri=>cic[0]);

if (($rU=~"^(+)[0-9]{1,20}$")) {            if(mt_match("ace", "$rU","0")) {                 insert_hf("Route: <sip:"+"$var(mtval)"+ ";lr\r\n");                 route(SYLKSERVERPOST);                 }         }

route(RELAY);         exit; #!endif         return; }

# Sylkserver routing route[SYLKSERVERPOST] {      if(mt_match("ace", "$rU","0")) {          t_relay_to("udp:10.0.0.4", "0x01");          exit;    } }

sample INVITE being forwarded

Session Initiation Protocol (SIP as raw text)     INVITE sip:+440000000@spe01.vodafone.com SIP/2.0\r\n     Record-Route: sip:10.1.1.3;lr\r\n     Route: sip:10.1.1.4:5060;lr\r\n     Via: SIP/2.0/UDP 10.1.1.3;branch=z9hG4bK4b7b.693512744570a1e2ea323648d7423180.0\r\n     From: sip:+4400000000@10.1.1.2:5060;tag=1\r\n     To: sip:+4416300000001@spe01.vodafone.com:5060;user=phone\r\n     Call-ID: 1-9256@10.1.1.4\r\n     Date: TUE, 13 DEC 2016 09:34:57 GMT\r\n     Supported: timer,resource-priority,replaces\r\n     Min-SE: 900\r\n     User-Agent: Cisco-CUCM10.5\r\n     Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY\r\n     CSeq: 101 INVITE\r\n

please help how to change below via header

Via: SIP/2.0/UDP 10.1.1.3;branch=z9hG4bK4b7b.693512744570a1e2ea323648d7423180.0

to

Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK4b7b.693512744570a1e2ea323648d7423180.0

my kamailio server is already listing to <IP>.:5060 port on UDP.

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