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37519 discussions

SBC and HA with Kamailio
by kai.ohnacker＠cbc.de 30 Apr '15

30 Apr '15

Hello, I'm new with Kamailio and have to do a research project with Kamailio server. My task is to realize a SIP infrastructure with NATing and a High Availability Cluster with two Kamailio server. For the NAT problem I found a solution with SEMS, but for security I need authentication on the SBC. I have no idea to synchronize the user authentication information from Kamailio to SEMS. Perhaps there is a more useful solution available? The SIP server is behind the NAT in a private network and the SBC to be placed in the DMZ. A new feature of Kamailio 4.2 is a replication for High Availability (HA), but I did not found any information about this. Is there somewhere a documentation or howto where I can look? I need only short keywords and not a complete description. Many thanks, Kai

2 1

NO VOICE AFTER MSILO
by sscc 30 Apr '15

30 Apr '15

hello i have configured msilo module successful but there isn't any voice with msilo. i debug and compare the call flow with and without msilo. with msilo in call flow it didn't follow to relay and consequently didnt activate rtpproxy befor the call is answered. due to which during call there isn't any voice. please help -- View this message in context: http://sip-router.1086192.n5.nabble.com/NO-VOICE-AFTER-MSILO-tp137429.html Sent from the Users mailing list archive at Nabble.com.

1 0

issue with TLS and 2 NIC interfaces
by Vik Killa 29 Apr '15

29 Apr '15

Hello, if i have two interfaces (eth0 and eth0:0) I set kamailio to listen to the IP on eth0:0 This works great except when I try using TLS, kamailio routes traffic from eth0:0 to eth0 then to correct destination It should be just doing eth0:0 -> correct destination If I use UDP/TCP I am not seeing this behavior. If I set kamailio to listen to eth0 and remove eth0;0, TLS works fine. I'm running kamailio compiled from source git master a few days ago. Is anyone else having this problem? I'm wondering if it's a bug. Thanks, V

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Re: [SR-Users] Running Kamailio with IMS
by Abdul Hakeem 29 Apr '15

29 Apr '15

Mihail, Are you able to share your configurations for each of the elements (P-CSCF, S-CSCF and I-CSCF) ? Cheers, Abdul Hakeem -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Mihail Dakov Sent: Wednesday, April 29, 2015 1:03 PM To: miconda(a)gmail.com; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Running Kamailio with IMS Hi All, I solved the issue by compiling kamailio from src and running different instances for each component P-CSCF, S-CSCF, and I-CSCF with different configuration respectively. Thank you all for the support. On 04/29/2015 12:04 PM, Daniel-Constantin Mierla wrote: > Hello, > > On 29/04/15 10:06, Mihail Dakov wrote: >> Hi all, >> >> I am trying to run Kamailio server with IMS modules for which I have >> gone through the following tutorials: >> >> http://nil.uniza.sk/ngnims/kamailio-ims/preparing-debian-operating-system-k… io-4x-platform >> >> >> http://www.kamailio.org/wiki/tutorials/ims/installation-howto >> >> The problem I have is that using 'service kamailio start' I can only >> run one IMS component, say PCSCF. But I need to run all P/S/I-CSCFs. >> How do I proceed to get running P/S/I-CSCF on the same machine with >> kamailio? > you can run multiple instances of kamailio on the same system, just use > different ports for each instance. Or, if you can allocate many IP > addresses to the system, you can run one per IP. Just specify the ip and > port to listen on via global parameter: > > listen=1.2.3.4:5080 > > If you questions is about combining all those configs, perhaps is > possible, but not something I can assist with. > > Cheers, > Daniel > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Kamailio and rtpengine - client behind NAT
by Martin Schiøtz 29 Apr '15

29 Apr '15

Hi I'm using Kamailio 4.2.4 and rtpengine 4.0.0.0+0~mr4.0.0.0. Kamailio is configured with sip users that can register. I have also configured a sip trunk to PSTN. I'm using rtpengine to handle RTP when a client want's to call to PSTN. When a user behind NAT wants to make a call to PSTN and sends INVITE, then the rtpengine gets the SDP info from kamailio and rtppengine creates the new SDP info to be forwarded to PSTN. The problem is that when the call is established rtpengine sends RTP to 10.0.0.100 instead of sending RTP to the NAT'ed users public IP 80.71.X.X ? I thought that "fix_nated_sdp("2")" would do the job before "rtpengine_offer("asymmetric")" but it dosen't have any effect? I have included kamailio.cfg, rtpengine logs and ngrep between kamailio and rtpengine. Cheers, Martin kamailio.cfg --------------------------------------- #!KAMAILIO ## !define WITH_DEBUG #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS #!define WITH_PSTN # # Kamailio (OpenSER) SIP Server v3.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "trunk.testconnect.dk" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" # loadmodule "rtpengine.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" # loadmodule "rtpproxy.so" loadmodule "rtpengine.so" loadmodule "path.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- 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) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- 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) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- # modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") # single rtproxy #modparam("rtpengine", "rtpengine_disable_tout", -1) #modparam("rtpengine", "rtpengine_sock", "udp:localhost:12222") modparam("rtpengine", "rtpengine_sock", "udp:localhost:22222") modparam("path", "use_received", 1); # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { if($rU=~"^\+[1-9][0-9]{3,20}$") { xlog("L_INFO", "MS: BEFORE STRIP $rm from $fu:$sp) to $ru\n"); strip(3); xlog("L_INFO", "MS: AFTER STRIP $rm from $fu:$sp) to $ru\n"); } else if($rU=~"^00[1-9][0-9]{3,20}$") { xlog("L_INFO", "MS: BEFORE STRIP $rm from $fu:$sp) to $ru\n"); strip(4); xlog("L_INFO", "MS: AFTER STRIP $rm from $fu:$sp) to $ru\n"); } xlog("L_INFO", "MS: BEFORE REQINIT $rm from $fu:$sp) to $ru\n"); # per request initial checks route(REQINIT); xlog("L_INFO", "MS: AFTER REQINIT $rm from $fu:$sp) to $ru\n"); # NAT detection route(NAT); xlog("L_INFO", "MS: AFTER NAT $rm from $fu:$sp) to $ru\n"); # handle requests within SIP dialogs route(WITHINDLG); xlog("L_INFO", "MS: AFTER WITHINDLG $rm from $fu:$sp) to $ru\n"); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); xlog("L_INFO", "MS: AFTER PRESENCE $rm from $fu:$sp) to $ru\n"); # handle registrations route(REGISTRAR); xlog("L_INFO", "MS: AFTER REGISTRAR $rm from $fu:$sp) to $ru\n"); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } xlog("L_INFO", "MS: BEFORE PSTN $rm from $fu:$sp) to $ru\n"); # dispatch destinations to PSTN route(PSTN); xlog("L_INFO", "MS: AFTER PSTN $rm from $fu:$sp) to $ru\n"); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { xlog("L_INFO", "MS: RELAY NAT RTPPROXY $rm from $fu:$sp) to $ru\n"); route(RTPPROXY); } #!endif #if (!isflagset(FLT_NATS) && !isbflagset(FLB_NATB) && is_method("INVITE")) { # xlog("L_INFO", "MS: RELAY ALWAYS RTPPROXY $rm from $fu:$sp) to $ru\n"); # route(RTPPROXY); #} #rtpengine_manage(); /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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 (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } xlog("L_INFO", "MS: route[WITHINDLG] 404 $rm from $fu:$sp) to $ru\n"); sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); xlog("L_INFO", "MS: LOCATION 404 $rm from $fu:$sp) to $ru\n"); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); setflag(FLT_NATS); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } if (method=="INVITE" && has_body("application/sdp")) { xlog("L_INFO", "MS: fix_nated_sdp $rm from $fu:$sp) to $ru\n"); #fix_nated_sdp("11"); #fix_nated_sdp("2"); #fix_nated_sdp("8"); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { ## unforce_rtp_proxy(); rtpengine_delete(); } else if (is_method("INVITE")){ # force_rtp_proxy(); ## rtpproxy_offer(); #rtpengine_manage(); rtpengine_offer("asymmetric"); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing #if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) { # xlog("L_INFO", "MS: PSTN not international $rm from $fu:$sp) to $ru\n"); # xlog("L_INFO", "MS: PSTN not international rU=$rU\n"); # return; #} xlog("L_INFO", "MS: PSTN check myself $rm from $fu:$sp) to $rU\n"); if($rU=~"^201111[0-9][0-9]$") { xlog("L_INFO", "MS: PSTN to myself $rm from $fu:$sp) to $ru\n"); return; } # only local users allowed to call if(from_uri!=myself) { xlog("L_INFO", "MS: PSTN 403 $rm from $fu:$sp) to $ru\n"); sl_send_reply("403", "Not Allowed"); exit; } #if (is_method("INVITE")) { # rtpproxy_offer(); #} xlog("L_INFO", "MS: PSTN before rewrite $rm from $fu:$sp) to $ru\n"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); xlog("L_INFO", "MS: PSTN before rewrite $rm from $fu:$sp) to $ru\n"); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { # force_rtp_proxy(); ## rtpproxy_answer(); rtpengine_answer("asymmetric"); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { #unforce_rtp_proxy(); rtpengine_delete(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } --------------------------------------- rtpengine logs - rtpengine offer --------------------------------------- Apr 23 00:52:30 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] Received command 'offer' from 127.0.0.1:56696 Apr 23 00:52:30 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] Dump for 'offer' from 127.0.0.1:56696: { "sdp": "v=0#015#012o=20111101-jitsi.org 0 0 IN IP4 10.0.0.100#015#012s=-#015#012c=IN IP4 10.0.0.100#015#012t=0 0#015#012m=audio 5034 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level#015#012a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=rtcp-xr:voip-metrics#015#012", "flags": [ "asymmetric" ], "call-id": "1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0", "received-from": [ "IP4", "80.71.X.X" ], "from-tag" ... Apr 23 00:52:30 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] ... : "6bb9cb39", "command": "offer" } Apr 23 00:52:30 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] Creating new call Apr 23 00:52:30 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] Opened ports 10728..10729 for media relay Apr 23 00:52:30 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] Opened ports 10746..10747 for media relay Apr 23 00:52:30 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] Shutting down ICE agent (nothing to do) Apr 23 00:52:30 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] Replying to 'offer' from 127.0.0.1:56696 Apr 23 00:52:30 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] Response dump for 'offer' to 127.0.0.1:56696: { "sdp": "v=0#015#012o=20111101-jitsi.org 0 0 IN IP4 10.0.0.100#015#012s=-#015#012c=IN IP4 188.166.X.X#015#012t=0 0#015#012m=audio 10728 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=rtcp-xr:voip-metrics#015#012a=sendrecv#015#012a=rtcp:10729#015#012a=ice-ufrag:LA1hlEBj#015#012a=ice-pwd:S99AwTSnHtaGnaeCv0DL795QH7#015#012a=candidate:rPBAvp2vmOwOv6BS 1 UDP 2130706431 188.166.X.X 10728 typ host#015#012a=candidate:rPBAvp2vmOwOv6BS 2 UDP 2130706430 188.166.X.X 10729 typ host#015#012", "result": " ... Apr 23 00:52:30 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] ... ok" } rtpengine logs - rtpengine delete --------------------------------------- Apr 23 00:53:16 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] Call branch '6bb9cb39' deleted, no more branches remaining Apr 23 00:53:16 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] Final packet stats: Apr 23 00:53:16 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] --- Tag 'gK04ca43fa', created 0:46 ago, in dialogue with '6bb9cb39' Apr 23 00:53:16 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] ------ Media #1 (audio over RTP/AVP) using PCMU/8000 Apr 23 00:53:16 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] --------- Port 10728 <> 188.226.X.X:10146, 544 p, 93568 b, 0 e, 1429743164 last_packet Apr 23 00:53:16 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] --------- Port 10729 <> 188.226.X.X:10147 (RTCP), 0 p, 0 b, 0 e, 1429743150 last_packet Apr 23 00:53:16 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] --- Tag '6bb9cb39', created 0:46 ago, in dialogue with 'gK04ca43fa' Apr 23 00:53:16 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] ------ Media #1 (audio over RTP/AVP) using PCMU/8000 Apr 23 00:53:16 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] --------- Port 10746 <> 10.0.0.100:5034 , 595 p, 102340 b, 0 e, 1429743166 last_packet Apr 23 00:53:16 sipserver-01 rtpengine[27441]: [1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0] --------- Port 10747 <> 10.0.0.100:5035 (RTCP), 3 p, 228 b, 0 e, 1429743166 last_packet Apr 23 00:53:16 sipserver-01 rtpengine[27441]: ci=1040bcc4797199cf13f2820d7e0bea88@0:0:0:0:0:0:0:0, created_from= 127.0.0.1:56696, last_signal=1429743164, tos=0, ml0_start_time=1429743160.638873, ml0_end_time=1429743166.013947, ml0_duration=5.375074, ml0_termination=REGULAR, ml0_local_tag=gK04ca43fa, ml0_local_tag_type=FROM_TAG, ml0_remote_tag=6bb9cb39, ml0_midx1_rtp_endpoint_ip=188.226.X.X, ml0_midx1_rtp_endpoint_port=10146, ml0_midx1_rtp_local_relay_port=10728, ml0_midx1_rtp_relayed_packets=544, ml0_midx1_rtp_relayed_bytes=93568, ml0_midx1_rtp_relayed_errors=0, ml0_midx1_rtp_last_packet=1429743164, ml0_midx1_rtcp_endpoint_ip=188.226.X.X, ml0_midx1_rtcp_endpoint_port=10147, ml0_midx1_rtcp_local_relay_port=10729, ml0_midx1_rtcp_relayed_packets=0, ml0_midx1_rtcp_relayed_bytes=0, ml0_midx1_rtcp_relayed_errors=0, ml0_midx1_rtcp_last_packet=1429743150, ml1_start_time=1429743164.196879, ml1_end_time=1429743166.013948, ml1_duration=1.817069, ml1_termination=REGULAR, ml1_local_tag=6bb9cb39, ml1_local_tag_type=TO_TAG, ml1_remote_tag=gK04ca43fa, ml1_midx1_rtp_endpoint_ip=10.0.0.100, ml1_midx1_rtp_endpoint_port=5034, ml1_midx1_rtp_local_relay_port=10746, ml1_midx1_rtp_relayed_packets=595, ml1_midx1_rtp_relayed_bytes=102340, ml1_midx1_rtp_relayed_errors=0, ml1_midx1_rtp_last_packet=1429743166, ml1_midx1_rtcp_endpoint_ip=10.0.0.100, ml1_midx1_rtcp_endpoint_port=5035, ml1_midx1_rtcp_local_relay_port=10747, ml1_midx1_rtcp_relayed_packets=3, ml1_midx1_rtcp_relayed_bytes=228, ml1_midx1_rtcp_relayed_errors=0, ml1_midx1_rtcp_last_packet=1429743166, -------------------------------- ngrep - SDP send between kamailio and rtpengine: -------------------------------- U 127.0.0.1:56696 -> 127.0.0.1:22222 10856_6 d3:sdp303:v=0. o=20111101-jitsi.org 0 0 IN IP4 10.0.0.100. s=-. c=IN IP4 10.0.0.100. t=0 0. m=audio 5034 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level. a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level. a=rtcp-xr:voip-metrics. 5:flagsl10:asymmetrice7:call-id48:1040bcc4797199cf13f2820d7e0bea88@0 :0:0:0:0:0:0:013:received-froml3:IP413:80.71.X.Xe8:from-tag8:6bb9cb397:command5:offere # U 127.0.0.1:22222 -> 127.0.0.1:56696 10856_6 d3:sdp430:v=0. o=20111101-jitsi.org 0 0 IN IP4 10.0.0.100. s=-. c=IN IP4 188.166.X.X. t=0 0. m=audio 10728 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=rtcp-xr:voip-metrics. a=sendrecv. a=rtcp:10729. a=ice-ufrag:LA1hlEBj. a=ice-pwd:S99AwTSnHtaGnaeCv0DL795QH7. a=candidate:rPBAvp2vmOwOv6BS 1 UDP 2130706431 188.166.X.X 10728 typ host. a=candidate:rPBAvp2vmOwOv6BS 2 UDP 2130706430 188.166.X.X 10729 typ host. 6:result2:oke # --------------------------------

2 4

Measuring subscriber latency
by Jon Bonilla (Manwe) 29 Apr '15

29 Apr '15

Hi all I'm replacing an Asterisk based system with a kamailio based one. One of the features the legacy system has is showing the subscriber the latency obtained from the qualify option of sip.conf Now, I'd like to measure the latency but I'm not sure how to do it. AFAIK nathelper module sends the OPTIONS keepalive messages stateless mode and there's no information there. I was thinking on triggering a route send_options via the timer module, save the timestamp of the relay and the timestamp of the response in onreply_route but it doesn't look elegant. Creating my own daemon in an external server and reading the info from the location module seems to be another option. I guess I'm not the first one with this need so I wonder if there's an already existing solution or an elegant way of dealing with it. any ideas? thanks, Jon

4 3

LCR with gateway capabilities
by Grant Bagdasarian 29 Apr '15

29 Apr '15

Hello, >From what I understand about the LCR module is that the rules have to be prioritized by the admin, be it manually or automatically by an application. Let's say the LCR database has 10 gateways, each with their own rules etc. 4 of these gateways support caller id spoofing and the others don't. But the other 6 are cheaper and Kamailio has to route a call using a gateway which support caller id spoofing and is the cheapest of the 4. Would it be possible to tell the LCR module to select a gateway based on certain capabilities and is the cheapest of the ones which support a certain capability?

2 5

Running Kamailio with IMS
by Mihail Dakov 29 Apr '15

29 Apr '15

Hi all, I am trying to run Kamailio server with IMS modules for which I have gone through the following tutorials: http://nil.uniza.sk/ngnims/kamailio-ims/preparing-debian-operating-system-k… http://www.kamailio.org/wiki/tutorials/ims/installation-howto The problem I have is that using 'service kamailio start' I can only run one IMS component, say PCSCF. But I need to run all P/S/I-CSCFs. How do I proceed to get running P/S/I-CSCF on the same machine with kamailio? br, m.dakov

2 2

2 phones behind same NAT, Kamailio on Pub IP One Way Audio
by Todd R. 29 Apr '15

29 Apr '15

This is my first go round' with Kamailio but I have been messing with it off and on for a few weeks. Running latest version on latest CentOS 7.x. Kamailio installed on VM with public IP in one location, 2 SIP phones behind same NAT, both registered to Kamailio fine. I can call one phone to the other and it rings and I can answer it but get one way audio most of the time, sometimes no audio at all. I just did a standard install with MySQL and no other modules. What am I missing, do I need RTPPROXY? Will Kamailio allow extension to extension calls from/to phones behind NATS without any additional modules? I tried installing it with yum, it installed but it won't start. If I need it, I will remove the YUM version and install from source or GIT or some other method. I do NOT want any media passing through this server, that's one big reason I am learning something OTHER than Asterisk which I am very familiar with. I see all these RTP modules, music on hold etc but I specifically don't want to install anything that causes media to pass through the box. Finding the learning curve much steeper than Asterisk back in the day and a REALLY tough time find step by step examples to get started, especially on CentOS. I bought the draft of the book and it will be a great resource but at the moment, it's not helping me get started or get my first instance up and running. I need to add SIP trunks to originate and terminate calls but I can't even think about that since I can't even get audio on extension to extension calls yet. I also see that Kamailio fails to startup at boot because it can't connect to MySQL, when I restart Kamailio it starts fine. I think the issue is that Kamailio is trying to start before MySQL is up and running, I guess I can do a delayed start of Kamailio to fix that later. Any help would be appreciate by this NEWB. Thanks in advance for any assistance.

2 1

publish authentication problem
by Tomas Zanet 28 Apr '15

28 Apr '15

Hello everyone, I've been working on a project where I would like to implement RFC6035 https://tools.ietf.org/html/rfc6035 Basically, after the end of the call, our SIP UAC sends a SIP PUBLISH to Kamailio where there are some useful information about media quality (packet loss, mos, etc..) This information is stored by Kamailio with sql_query command... To achieve this, I implemented a custom route function inside Kamailio cfg file, which handles PUBLISH request, check vq-rtcpxr body and store that information into a database. Here is a snippet code: # AVPF report route route[AVPF] { if(!is_method("PUBLISH")) return; if (has_body("application/vq-rtcpxr")) { $var(x) = "INSERT INTO quality_reporting_raw(body) VALUES (\"" + $(rb{s.escape.common}) + "\");"; sql_query("cb", "$(var(x))"); ...... reply 200 ok and so on.... exit(0) } } Everything works fine (almost fine, because the report is stored twice...) if I do the route(AVPF) before authentication, like this... (main route): # handle AVPF reports route(AVPF); # authentication route(AUTH); Instead if I swap these two functions, doing the authentication before publish processing, I have some problems about publish authentication....not for the other requests (INVITE and REGISTER) As far as I know From, To and Request-URI must match the authentication user, in the PUBLISH REQUEST as described here http://kamailio.org/docs/modules/4.2.x/modules/auth_db.html#auth_db.f.auth_… I confirm that...I'm sending the SIP PUBLISH from user 101 to user 101, which is currently registered. My problem is that Kamailio is continuously answering with 407 to the SIP PUBLISH, even if the PUBLISH request has the right Proxy-Authorization header... Digging into the log file, I found this : Apr 24 14:30:58 bptrnddmzserver kamailio[30886]: DEBUG: auth_db [authorize.c:486]: auth_check(): realm [myhome2.xip.bpt.com] table [subscriber] flags [1] Apr 24 14:30:58 bptrnddmzserver kamailio[30886]: DEBUG: auth [api.c:86]: pre_auth(): auth:pre_auth: Credentials with realm 'myhome2.xip.bpt.com' not found Apr 24 14:30:58 bptrnddmzserver kamailio[30886]: DEBUG: auth_db [authorize.c:252]: digest_authenticate_hdr(): no credentials Just for your better understanding I'm doing authorization with db, using subscriber table where there are all the users with 'myhome2.xip.bpt.com' domain... In fact, all INVITEs and REGISTERs are correctly authenticated... Would you mind address me to find out the problem? Maybe, there's something wrong in my cfg file. Thanks in advance T, Tomas Zanet Software Design Department tzanet(a)came.com [cid:CAME.png]<http://www.came.com/> [cid:LogoExpo.png] [cid:FooterCAME.png] Il messaggio di posta elettronica contiene informazioni di carattere confidenziale specifiche per il destinatario. Nel caso non ne siate il destinatario, segnalatelo immediatamente al mittente ed eliminate dai vostri archivi quanto ricevuto (compresi i file allegati). L'uso, la diffusione, distribuzione o riproduzione del presente messaggio e dei suoi allegati da parte di ogni altra persona costituisce reato. Rif. Decreto legislativo 30 giugno 2003, n. 196 - Codice in materia di protezione dei dati personali. The email message contains confidential information specific to the recipient. If you are not the recipient, write it to the sender immediately and delete from your files as received (including file attachments). Use, dissemination, distribution or reproduction of this message and its attachments by any other person is a criminal offense. References Legislative Decree 30 June 2003, n. 196 - Code for the protection of personal data. Right to opposition: The person concerned who receives the message has the right at any time to oppose its processing for forwarding commercial messages, advertising material or direct sales proposals, by clicking on the e-mail address below (or by traditional postal services by writing to: Came S.p.A., Via Martiri della Libert? 15, 31030 Dosson di Casier (TV) - Italy, or by sending a fax to +39 0422 4941. Furthermore the person concerned may exercise all the rights of access to the personal data as provided by art. 7, Legislative Decree no. 196/2003, including the rights to amendment, updating and deletion, by sending an e-mail to privacy(a)came.com<mailto:privacy@came.com>.

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