Hola, estoy utilizando kamailio 3.2.0 (x86_64/linux) y desde hace unos días estoy intentando comunicar un softphone desde el interior de una red con NAT con el exterior. La configuración es la siguiente:

Softphone (192.168.0.5) <--> Kamailio (192.168.0.3) <--> Router (192.168.0.1) <--> Softphone sobre smartphone

Todos los puertos del router están redireccionados a 192.168.0.3 Tengo instalado rtpproxy 1.2.1-1 con la siguiente configuración:

--------------------------/etc/defaults/rtpproxy-------------------------------- # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222

# Additional options that are passed to the daemon. EXTRA_OPTS="-l candamil.dyndns.org" ----------------------------------------------------------------------------------------

y el funcionamiento es correcto. Este es el mensaje de salida del log de kamailio:

---------------------------------------------------------------------------------------------- Nov 12 20:09:13 condor kamailio[7001]: INFO: rtpproxy [rtpproxy.c:1415]: rtp proxy udp:127.0.0.1:22222 found, support for it enabled -----------------------------------------------------------------------------------------------

Tanto el softphone del interior de la red como el del exterior son linphone. La configuración del del interior de la red es la siguiente:

SIP identity: sip:1001@192.168.0.3 SIP proxy: sip:192.168.0.3 Indico conexión directa a internet.

En la del externo especifico como proxy y domain "candamil.dyndns.org", que es la dirección DNS que apunta a la IP de mi router.

En este caso, los síntomas son los siguientes: La autentificación de ambos, tanto externo como interno, es correcta. Al realizar una llamada en cualquiera de los dos sentidos, el softphone suena y la llamada se contesta y cuelga correctamente, pero no se transmite la señal de voz. En el log figura el siguiente mensaje:

---------------------------------------------------------------------------------------------- Nov 12 20:23:14 condor kamailio[6991]: ERROR: rtpproxy [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy ----------------------------------------------------------------------------------------------

Ocurre lo mismo si en el interno indico que está tras NAT, con la ip del router, y que está tras NAT con un servidor STUN (stunserver.org). En los 3 casos, en el softphone externo, figura una llamada de 1001@192.168.0.3, mientras que en el interno, figura una llamada de 1002@candamil.dyndns.org.

Si cambio en el softphone interno el proxy a sip:candamil.dyndns.org, todo se vuelve a repetir. Si lo que hago es cambiar la SIP identity a sip:1001@candamil.dyndns.org, al realizar una llamada desde interno a externo, el interno no llega a percatarse de que se respondió a la llamada, y aparece el siguiente error en el log:

---------------------------------------------------------------------------------------------------------- Nov 12 20:53:00 condor kamailio[7306]: ERROR: <core> [parser/parse_via.c:2600]: ERROR: parse_via: invalid port number <5060ranch=z9hG4bKc 50f.b4825246.0> Nov 12 20:53:00 condor kamailio[7306]: ERROR: <core> [parser/parse_via.c:2629]: ERROR: parse_via on: <SIP/2.0/UDP 192.168.0.3:5060ranch=z 9hG4bKc50f.b4825246.0;received=87.223.138.84#015#012Via: SIP/2.0/UDP 87.223.138.84:5060;rport=5060;branch=z9hG4bK1021772993#015#012From: sip:1001@candamil.dyndns.org;tag=783852345#015#012To: < sip:1002@candamil.dyndns.org>#015#012Call-ID: 1644787160#015#012CSeq: 21 INVITE# 015#012User-Agent: Linphone/3.4.0 (eXosip2/unknown)#015#012Content-Length: 0#015#012#015#012> ------------------------------------------------------------------------------------------------------------

Por el contrario, haciendo una llamada desde el externo al interno, ocurre lo mismo que antes.

Por último, esta es la configuración relevante de kamailio:

-----------------------------------------kamailio.cfg---------------------------------------------------- #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_NAT

####### 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://openser:openserrw@localhost/openser" #!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 ######### /* 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="candamil.dyndns.org"

/* 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

# life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605

####### Custom Parameters #########

#!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 = "" desc "PSTN GW Address" #!endif

####### Modules Section ########

#!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif

#!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so"

#!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif

# ----------------- setting module-specific parameters ---------------

# ----- 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)

#!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222")

# ----- 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

####### Routing Logic ########

# Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route {

# per request initial checks route(REQINIT);

# NAT detection route(NATDETECT);

# handle requests within SIP dialogs route(WITHINDLG);

### 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);

# handle registrations route(REGISTRAR);

if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }

# dispatch destinations to PSTN route(PSTN);

# user location service route(LOCATION);

route(RELAY); }

route[RELAY] {

# enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); }

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 } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } 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; } } 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_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif

#!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif

$avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: 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 ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } 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[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }

# RTPProxy control 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;

rtpproxy_manage();

if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!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(!($rU=~"^(+|00)[1-9][0-9]{3,20}$")) return;

# only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; }

$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);

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

# route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return;

# check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return;

$ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif

return; }

# manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); }

# manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); }

# manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE);

if (t_is_canceled()) { exit; }

#!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif

#!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } --------------------------------------------------------------------------------------------------------------

Espero que alguien pueda echarme una mano. Gracias.