No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy)
25 Jun
2009
25 Jun
'09
1:01 p.m.
Hola, este es mi primer post en esta lista,
Tengo instalado el kamailio 1.5.1 y estoy utilizando un kamailio.cfg utilizando el que
trae por defecto, luego he modificado el cfg para activar mysql, domain, presence,
nathelper y authentication con md5, todo funciona como se supone que deberia, los clientes
pueden registrarse, enviar mensajes de texto y hablar entre ellos. El unico problema es el
audio cuando dos clientes estan detras de una NAT, los telefonos pueden realizar la
llamada y suena el ring, pero cuando se descuelga no hay audio en ninguna direccion.
cuendo los telefonos tienen una IP publica todo funciona bien, tambien funciona cuando
utilizo un Linksys PAP2T con las opciones "Insert VIA received", "Insert
VIA rport", "Handle VIA received", "Handle VIA rport" y "NAT
mapping enable" encendidas, con el softphone de Qutecom stambien funciona.
Este problema me esta sucediendo con los telefonos fisicos Thomson phones (model ST 2022)
y GrandStream Budge Tone 200, este problema ocurre sin importar que opciones le coloque
para el tipo de nateo dentro de los telefonos, Incluso he utilizado stun con
stunserver.org o con el servidor de stun de ekiga, los telefonos se registran y pueden
hacer y recibir llamadas, pero no hay audio cuando atiendes la llamada.
Con kamctl ul show, puedes ver que han registrado el Contact con su IP local y el Received
con la IP publica y los puertos para el NAT
La unica diferencia con los Linksys que si funcionan es que los Linksys registran el
Contact con la IP publica.
Aqui se puede ver dos telefonos con NAT en el proxy
Domain:: location table=512 records=2 max_slot=1
AOR:: 20000004(a)212.4.107.250
Contact:: sip:20000004@192.168.254.110:5060;transport=udp;user=phone Q=
Expires:: 1150
Callid:: 72ed03f6d2f390f9(a)192.168.254.110
Cseq:: 10003
User-agent:: Grandstream BT200 1.1.6.27
Received:: sip:212.4.97.115:35379
State:: CS_NEW
Flags:: 0
Cflag:: 0
Socket:: udp:212.4.107.250:5060
Methods:: 7807
AOR:: 20000000(a)212.4.107.250
Contact:: sip:20000000@192.168.254.101:5060;user=phone Q=
Expires:: 2945
Callid:: 17fe-c0a80101-5-1(a)192.168.254.101
Cseq:: 6
User-agent:: THOMSON ST2022 hw2 fw3.56 00-18-F6-B5-7E-06
Received:: sip:212.4.97.115:55128
State:: CS_NEW
Flags:: 0
Cflag:: 0
Socket:: udp:212.4.107.250:5060
Methods:: 4294967295
Estoy utilizando rtpproxy y no hay ningun error en el log que indique que el rtpproxy no
esta funcionando, de hecho haciendo un SIP trace muestra al rtpproxy seteando puertos para
el audio.
Ejecuto el rtpproxy con este comando:
rtpproxy -l 212.4.107.250 -s udp:localhost:7722 -F
Cualquier ayuda sera apreciada, llevo dos semanas buscando una solucion
Adjunto mi kamailio.cfg para que puedan mirarlo, al final de este mensaje voy a adjuntar
el SIP Trace de una llamada entre dos telefonos detras de una NAT (un Thomson y un
GrandStream) en caso que puedan ayudarme a decifrar que esta mal aqui:
este es mi cfg
**************************************************************************************************
#
# $Id: kamailio.cfg 5800 2009-04-20 11:01:49Z miconda $
#
# Kamailio (OpenSER) SIP Server - basic configuration script
# - web: http://www.kamailio.org
# - svn: http://openser.svn.sourceforge.net/viewvc/openser/
#
# Direct your questions about this file to: <users(a)lists.kamailio.org>
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
#
# There are comments showing how to enable different features in th econfig
# file. Such commented code starts with #X# where X is a letter to identify
# a feature. Delete entire #X# if you want to enable that feature. Next are
# sed commands that help you enable such features.
#
# *** To enamble mysql execute:
# sed -i 's/#m#//g' kamailio.cfg
#
# *** To enamble authentication execute:
# - enable mysql
# sed -i 's/#a#//g' kamailio.cfg
# - add users using 'kamctl'
#
# *** To enamble persistent user location execute:
# - enable mysql
# sed -i 's/#u#//g' kamailio.cfg
#
# *** To enamble presence server execute:
# - enable mysql
# sed -i 's/#p#//g' kamailio.cfg
#
# *** To enamble nat traversal execute:
# sed -i 's/#n#//g' kamailio.cfg
# - install RTPProxy: http://www.rtpproxy.org
# - start RTPProxy:
# rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#
# *** To enhance accounting execute:
# - enable mysql
# sed -i 's/#c#//g' kamailio.cfg
# - add following columns to database
# 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_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
# ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT
'';
#
####### Global Parameters #########
debug=3
log_stderror=no
log_facility=LOG_LOCAL0
fork=yes
children=4
/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes
/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes
/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no
/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */
#dns_try_ipv6=yes
/* uncomment the next line to disable the auto discovery of local aliases
based on revers DNS on IPs (default on) */
#auto_aliases=no
/* uncomment the following lines to enable TLS support (default off) */
#disable_tls = no
#listen = tls:your_IP:5061
#tls_verify_server = 1
#tls_verify_client = 1
#tls_require_client_certificate = 0
#tls_method = TLSv1
#tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem"
#tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem"
#tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem"
port=5060
/* 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:192.168.1.2:5060
####### Modules Section ########
#set module path
mpath="/usr/local/lib/kamailio/modules/"
/* uncomment next line for MySQL DB support */
loadmodule "db_mysql.so"
loadmodule "mi_fifo.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "uri_db.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "acc.so"
/* uncomment next lines for MySQL based authentication support
NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "auth.so"
loadmodule "auth_db.so"
/* uncomment next line for aliases support
NOTE: a DB (like db_mysql) module must be also loaded */
#loadmodule "alias_db.so"
/* uncomment next line for multi-domain support
NOTE: a DB (like db_mysql) module must be also loaded
NOTE: be sure and enable multi-domain support in all used modules
(see "multi-module params" section ) */
loadmodule "domain.so"
/* uncomment the next two lines for presence server support
NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "presence.so"
loadmodule "presence_xml.so"
loadmodule "presence_mwi.so"#manually added
loadmodule "nathelper.so"
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- 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)
# ----- rr 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)
# ----- uri_db params -----
/* by default we disable the DB support in the module as we do not need it
in this configuration */
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "db_url", "")
# ----- acc params -----
/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
/* 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", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
/* uncomment the following lines to enable DB accounting also */
#c#modparam("acc", "db_flag", 1)
#c#modparam("acc", "db_missed_flag", 2)
#c#modparam("domain", "db_url",
#c# "mysql://openser:openserrw@localhost/openser")
#c#modparam("acc", "db_extra",
#c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
# ----- usrloc params -----
/* uncomment the following lines if you want to enable DB persistency
for location entries */
#u#modparam("usrloc", "db_mode", 2)
#u#modparam("usrloc", "db_url",
#u# "mysql://openser:openserrw@localhost/openser")
# ----- auth_db params -----
/* uncomment the following lines if you want to enable the DB based
authentication */
#a#modparam("auth_db", "calculate_ha1", yes)
#a#modparam("auth_db", "password_column", "password")
#a#modparam("auth_db", "db_url",
#a# "mysql://openser:openserrw@localhost/openser")
#a#modparam("auth_db", "load_credentials", "")
#parametros de autentificacion modificados manualmente
modparam("auth_db", "user_column", "username")
modparam("auth_db", "domain_column", "domain")
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "password_column_2", "ha1b")
modparam("auth_db", "calculate_ha1", 0)
#modparam("auth_db", "use_domain", 0)
modparam("auth_db", "use_domain", 1)#0 encendemos con 1 porque
utilizaremos multi-domain
modparam("auth_db", "load_credentials", "rpid")
modparam("auth_db", "db_url",
"mysql://openser:openserrw@localhost/openser")
# ----- alias_db params -----
/* uncomment the following lines if you want to enable the DB based
aliases */
#modparam("alias_db", "db_url",
# "mysql://openser:openserrw@localhost/openser")
# ----- domain params -----
/* uncomment the following lines to enable multi-domain detection
support */
modparam("domain", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("domain", "db_mode", 1) # Use caching
# ----- multi-module params -----
/* uncomment the following line if you want to enable multi-domain support
in the modules (dafault off) */
modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1)
# ----- presence params -----
/* uncomment the following lines if you want to enable presence */
modparam("presence|presence_xml", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("presence_xml", "force_active", 1)
modparam("presence", "server_address",
"sip:212.4.107.250:5060")
# -- nathelper
modparam("nathelper", "rtpproxy_sock",
"udp:127.0.0.1:7722")
modparam("nathelper", "natping_interval", 15)
modparam("nathelper", "ping_nated_only", 0)
modparam("nathelper", "sipping_bflag", 7)
modparam("nathelper", "sipping_from",
"sip:pinger@212.4.107.250")
modparam("registrar|nathelper", "received_avp",
"$avp(i:80)")
modparam("usrloc", "nat_bflag", 6)
modparam("nathelper", "sipping_method", "OPTIONS")
####### Routing Logic ########
# main request routing logic
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
# NAT detection
route(4);
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
}
route(1);
} else {
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(2);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from
upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ... ignore and discard.\n");
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
#initial requests
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
{
t_relay();
}
exit;
}
t_check_trans();
# authentication
route(3);
# record routing
if (!is_method("REGISTER|MESSAGE"))
{
record_route();
}
# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}
##if (!uri==myself)
/* replace with following line if multi-domain support is used */
if (!is_uri_host_local())
{
append_hf("P-hint: outbound\r\n");
# if you have some interdomain connections via TLS
##if($rd=="tls_domain1.net") {
## t_relay("tls:domain1.net");
## exit;
##} else if($rd=="tls_domain2.net") {
## t_relay("tls:domain2.net");
## exit;
##}
route(1);
}
# requests for my domain
if( is_method("PUBLISH|SUBSCRIBE"))
{
route(2);
}
if (is_method("REGISTER"))
{
if (!save("location"))
{
sl_reply_error();
}
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# apply DB based aliases (uncomment to enable)
##alias_db_lookup("dbaliases");
if (!lookup("location")) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
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
setflag(2);
route(1);
}
route[1] {
if (check_route_param("nat=yes")) {
setbflag(6);
setbflag(7);# sipping
}
if (isflagset(5) || isbflagset(6)) {
route(5);
}
/* example how to enable some additional event routes */
if (is_method("INVITE")) {
#t_on_branch("1");
t_on_reply("1");
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Presence route
/* uncomment the whole following route for enabling presence server */
route[2]
{
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;
# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==null)
{
sl_send_reply("404", "Not here");
exit;
}
return;
}
# Authentication route
/* uncomment the whole following route for enabling authentication */
route[3] {
if (is_method("REGISTER"))
{
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize("", "subscriber"))
{
www_challenge("", "0");
exit;
}
if ($au!=$tU)
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
}
# Auth only on registration
#a# } else {
#a# # authenticate if from local subscriber (uncomment to enable auth)
#a# if (from_uri==myself)
#a# {
#a# if (!proxy_authorize("", "subscriber")) {
#a# proxy_challenge("", "0");
#a# exit;
#a# }
#a# if (is_method("PUBLISH"))
#a# {
#a# if ($au!=$tU) {
#a# sl_send_reply("403","Forbidden auth ID");
#a# exit;
#a# }
#a# } else {
#a# if ($au!=$fU) {
#a# sl_send_reply("403","Forbidden auth ID");
#a# exit;
#a# }
#a# }
#a#
#a# consume_credentials();
#a# # caller authenticated
#a# }
#a# }
return;
}
# Caller NAT detection route
/* uncomment the whole following route for enabling Caller NAT Detection */
route[4]{
force_rport();
if (nat_uac_test("19")) {
if (method=="REGISTER") {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(5);
}
return;
}
# RTPProxy control
/* uncomment the whole following route for enabling RTPProxy Control */
route[5] {
if (is_method("BYE")) {
unforce_rtp_proxy();
} else if (is_method("INVITE")){
force_rtp_proxy();
}
if (!has_totag()) add_rr_param(";nat=yes");
return;
}
branch_route[1] {
xdbg("new branch at $ru\n");
}
onreply_route[1] {
xdbg("incoming reply\n");
if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])")
{
force_rtp_proxy();
}
if (isbflagset(6)) {
fix_nated_contact();
}
}
failure_route[1] {
if (is_method("INVITE")
&& (isbflagset(6) || isflagset(5))) {
unforce_rtp_proxy();
}
if (t_was_cancelled()) {
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();
##}
}
**************************************************************************************************
**************************************************************************************************
Aqui va el SIP Trace para una llamada de telefonos fisicos NATed to NATed:
**************************************************************************************************
U +0.161561 212.4.97.115:35379 -> 212.4.107.250:5060
INVITE sip:20000000@212.4.107.250;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668
From: "20000004"
<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>
Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 332
v=0
o=20000004 8000 8000 IN IP4 192.168.254.110
s=SIP Call
c=IN IP4 192.168.254.110
t=0 0
m=audio 40000 RTP/AVP 4 3 18 0 8 9 97
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:60
#
U +0.000407 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP
192.168.254.110:5060;branch=z9hG4bK8f809670adc00668;rport=35379;received=212.4.97.115
From: "20000004"
<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Server: Kamailio (1.5.1-notls (i386/linux))
Content-Length: 0
#
U +0.000034 212.4.107.250:5060 -> 212.4.97.115:55128
INVITE sip:20000000@192.168.254.101:5060;user=phone SIP/2.0
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004"
<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>
Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 348
v=0
o=20000004 8000 8000 IN IP4 192.168.254.110
s=SIP Call
c=IN IP4 212.4.107.250
t=0 0
m=audio 35752 RTP/AVP 4 3 18 0 8 9 97
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:60
a=nortpproxy:yes
#
U +0.019311 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Content-Length: 0
#
U +0.030480 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Length: 0
#
U +0.000083 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Length: 0
#
U +6.510103 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
#
U +0.000365 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes
#
U +0.034122 212.4.97.115:35379 -> 212.4.107.250:5060
ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bKdf5e0ceed72f3797
Route: <sip:212.4.107.250;lr=on;nat=yes>
From: "20000004"
<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone>
Supported: path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 ACK
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
#
U +0.000245 212.4.107.250:5060 -> 192.168.254.101:5060
ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.2
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bKdf5e0ceed72f3797
From: "20000004"
<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone>
Supported: path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 ACK
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
#
U +0.458031 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
#
U +0.000246 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes
#
U +0.999724 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
#
U +0.000295 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes
25 Jun
25 Jun
1:17 p.m.
En el INVITE reenviado de Kamailio al destino se ve la IP privada en el
SDP.
Tal vez no esté entrando en el route[5] que es donde se hace el
force_rtp_proxy. Prueba a que se ejecute siempre el force_rtp_proxy en
los INVITEs y sus respuestas a ver si así te funciona en este caso que
te da problemas.
G.
On 06/25/2009 01:01 PM, rubenrojas - Trc.es wrote:
Hola, este es mi primer post en esta lista,
Tengo instalado el kamailio 1.5.1 y estoy utilizando un kamailio.cfg utilizando el que
trae por defecto, luego he modificado el cfg para activar mysql, domain, presence,
nathelper y authentication con md5, todo funciona como se supone que deberia, los clientes
pueden registrarse, enviar mensajes de texto y hablar entre ellos. El unico problema es el
audio cuando dos clientes estan detras de una NAT, los telefonos pueden realizar la
llamada y suena el ring, pero cuando se descuelga no hay audio en ninguna direccion.
cuendo los telefonos tienen una IP publica todo funciona bien, tambien funciona cuando
utilizo un Linksys PAP2T con las opciones "Insert VIA received", "Insert
VIA rport", "Handle VIA received", "Handle VIA rport" y "NAT
mapping enable" encendidas, con el softphone de Qutecom stambien funciona.
Este problema me esta sucediendo con los telefonos fisicos Thomson phones (model ST 2022)
y GrandStream Budge Tone 200, este problema ocurre sin importar que opciones le coloque
para el tipo de nateo dentro de los telefonos, Incluso he utilizado stun con
stunserver.org o con el servidor de stun de ekiga, los telefonos se registran y pueden
hacer y recibir llamadas, pero no hay audio cuando atiendes la llamada.
Con kamctl ul show, puedes ver que han registrado el Contact con su IP local y el
Received con la IP publica y los puertos para el NAT
La unica diferencia con los Linksys que si funcionan es que los Linksys registran el
Contact con la IP publica.
Aqui se puede ver dos telefonos con NAT en el proxy
Domain:: location table=512 records=2 max_slot=1
AOR:: 20000004(a)212.4.107.250
Contact:: sip:20000004@192.168.254.110:5060;transport=udp;user=phone Q=
Expires:: 1150
Callid:: 72ed03f6d2f390f9(a)192.168.254.110
Cseq:: 10003
User-agent:: Grandstream BT200 1.1.6.27
Received:: sip:212.4.97.115:35379
State:: CS_NEW
Flags:: 0
Cflag:: 0
Socket:: udp:212.4.107.250:5060
Methods:: 7807
AOR:: 20000000(a)212.4.107.250
Contact:: sip:20000000@192.168.254.101:5060;user=phone Q=
Expires:: 2945
Callid:: 17fe-c0a80101-5-1(a)192.168.254.101
Cseq:: 6
User-agent:: THOMSON ST2022 hw2 fw3.56 00-18-F6-B5-7E-06
Received:: sip:212.4.97.115:55128
State:: CS_NEW
Flags:: 0
Cflag:: 0
Socket:: udp:212.4.107.250:5060
Methods:: 4294967295
Estoy utilizando rtpproxy y no hay ningun error en el log que indique que el rtpproxy no
esta funcionando, de hecho haciendo un SIP trace muestra al rtpproxy seteando puertos para
el audio.
Ejecuto el rtpproxy con este comando:
rtpproxy -l 212.4.107.250 -s udp:localhost:7722 -F
Cualquier ayuda sera apreciada, llevo dos semanas buscando una solucion
Adjunto mi kamailio.cfg para que puedan mirarlo, al final de este mensaje voy a adjuntar
el SIP Trace de una llamada entre dos telefonos detras de una NAT (un Thomson y un
GrandStream) en caso que puedan ayudarme a decifrar que esta mal aqui:
este es mi cfg
**************************************************************************************************
#
# $Id: kamailio.cfg 5800 2009-04-20 11:01:49Z miconda $
#
# Kamailio (OpenSER) SIP Server - basic configuration script
# - web: http://www.kamailio.org
# - svn: http://openser.svn.sourceforge.net/viewvc/openser/
#
# Direct your questions about this file to:<users@lists.kamailio.org>
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
#
# There are comments showing how to enable different features in th econfig
# file. Such commented code starts with #X# where X is a letter to identify
# a feature. Delete entire #X# if you want to enable that feature. Next are
# sed commands that help you enable such features.
#
# *** To enamble mysql execute:
# sed -i 's/#m#//g' kamailio.cfg
#
# *** To enamble authentication execute:
# - enable mysql
# sed -i 's/#a#//g' kamailio.cfg
# - add users using 'kamctl'
#
# *** To enamble persistent user location execute:
# - enable mysql
# sed -i 's/#u#//g' kamailio.cfg
#
# *** To enamble presence server execute:
# - enable mysql
# sed -i 's/#p#//g' kamailio.cfg
#
# *** To enamble nat traversal execute:
# sed -i 's/#n#//g' kamailio.cfg
# - install RTPProxy: http://www.rtpproxy.org
# - start RTPProxy:
# rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#
# *** To enhance accounting execute:
# - enable mysql
# sed -i 's/#c#//g' kamailio.cfg
# - add following columns to database
# 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_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
# ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT
'';
#
####### Global Parameters #########
debug=3
log_stderror=no
log_facility=LOG_LOCAL0
fork=yes
children=4
/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes
/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes
/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no
/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */
#dns_try_ipv6=yes
/* uncomment the next line to disable the auto discovery of local aliases
based on revers DNS on IPs (default on) */
#auto_aliases=no
/* uncomment the following lines to enable TLS support (default off) */
#disable_tls = no
#listen = tls:your_IP:5061
#tls_verify_server = 1
#tls_verify_client = 1
#tls_require_client_certificate = 0
#tls_method = TLSv1
#tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem"
#tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem"
#tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem"
port=5060
/* 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:192.168.1.2:5060
####### Modules Section ########
#set module path
mpath="/usr/local/lib/kamailio/modules/"
/* uncomment next line for MySQL DB support */
loadmodule "db_mysql.so"
loadmodule "mi_fifo.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "uri_db.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "acc.so"
/* uncomment next lines for MySQL based authentication support
NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "auth.so"
loadmodule "auth_db.so"
/* uncomment next line for aliases support
NOTE: a DB (like db_mysql) module must be also loaded */
#loadmodule "alias_db.so"
/* uncomment next line for multi-domain support
NOTE: a DB (like db_mysql) module must be also loaded
NOTE: be sure and enable multi-domain support in all used modules
(see "multi-module params" section ) */
loadmodule "domain.so"
/* uncomment the next two lines for presence server support
NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "presence.so"
loadmodule "presence_xml.so"
loadmodule "presence_mwi.so"#manually added
loadmodule "nathelper.so"
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- 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)
# ----- rr 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)
# ----- uri_db params -----
/* by default we disable the DB support in the module as we do not need it
in this configuration */
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "db_url", "")
# ----- acc params -----
/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
/* 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", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
/* uncomment the following lines to enable DB accounting also */
#c#modparam("acc", "db_flag", 1)
#c#modparam("acc", "db_missed_flag", 2)
#c#modparam("domain", "db_url",
#c# "mysql://openser:openserrw@localhost/openser")
#c#modparam("acc", "db_extra",
#c#
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
# ----- usrloc params -----
/* uncomment the following lines if you want to enable DB persistency
for location entries */
#u#modparam("usrloc", "db_mode", 2)
#u#modparam("usrloc", "db_url",
#u# "mysql://openser:openserrw@localhost/openser")
# ----- auth_db params -----
/* uncomment the following lines if you want to enable the DB based
authentication */
#a#modparam("auth_db", "calculate_ha1", yes)
#a#modparam("auth_db", "password_column", "password")
#a#modparam("auth_db", "db_url",
#a# "mysql://openser:openserrw@localhost/openser")
#a#modparam("auth_db", "load_credentials", "")
#parametros de autentificacion modificados manualmente
modparam("auth_db", "user_column", "username")
modparam("auth_db", "domain_column", "domain")
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "password_column_2", "ha1b")
modparam("auth_db", "calculate_ha1", 0)
#modparam("auth_db", "use_domain", 0)
modparam("auth_db", "use_domain", 1)#0 encendemos con 1 porque
utilizaremos multi-domain
modparam("auth_db", "load_credentials", "rpid")
modparam("auth_db", "db_url",
"mysql://openser:openserrw@localhost/openser")
# ----- alias_db params -----
/* uncomment the following lines if you want to enable the DB based
aliases */
#modparam("alias_db", "db_url",
# "mysql://openser:openserrw@localhost/openser")
# ----- domain params -----
/* uncomment the following lines to enable multi-domain detection
support */
modparam("domain", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("domain", "db_mode", 1) # Use caching
# ----- multi-module params -----
/* uncomment the following line if you want to enable multi-domain support
in the modules (dafault off) */
modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1)
# ----- presence params -----
/* uncomment the following lines if you want to enable presence */
modparam("presence|presence_xml", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("presence_xml", "force_active", 1)
modparam("presence", "server_address",
"sip:212.4.107.250:5060")
# -- nathelper
modparam("nathelper", "rtpproxy_sock",
"udp:127.0.0.1:7722")
modparam("nathelper", "natping_interval", 15)
modparam("nathelper", "ping_nated_only", 0)
modparam("nathelper", "sipping_bflag", 7)
modparam("nathelper", "sipping_from",
"sip:pinger@212.4.107.250")
modparam("registrar|nathelper", "received_avp",
"$avp(i:80)")
modparam("usrloc", "nat_bflag", 6)
modparam("nathelper", "sipping_method", "OPTIONS")
####### Routing Logic ########
# main request routing logic
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
# NAT detection
route(4);
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
}
route(1);
} else {
if (is_method("SUBSCRIBE")&& uri == myself)
{
# in-dialog subscribe requests
route(2);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an
ACK after a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ... ignore
and discard.\n");
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
#initial requests
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
{
t_relay();
}
exit;
}
t_check_trans();
# authentication
route(3);
# record routing
if (!is_method("REGISTER|MESSAGE"))
{
record_route();
}
# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}
##if (!uri==myself)
/* replace with following line if multi-domain support is used */
if (!is_uri_host_local())
{
append_hf("P-hint: outbound\r\n");
# if you have some interdomain connections via TLS
##if($rd=="tls_domain1.net") {
## t_relay("tls:domain1.net");
## exit;
##} else if($rd=="tls_domain2.net") {
## t_relay("tls:domain2.net");
## exit;
##}
route(1);
}
# requests for my domain
if( is_method("PUBLISH|SUBSCRIBE"))
{
route(2);
}
if (is_method("REGISTER"))
{
if (!save("location"))
{
sl_reply_error();
}
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# apply DB based aliases (uncomment to enable)
##alias_db_lookup("dbaliases");
if (!lookup("location")) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
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
setflag(2);
route(1);
}
route[1] {
if (check_route_param("nat=yes")) {
setbflag(6);
setbflag(7);# sipping
}
if (isflagset(5) || isbflagset(6)) {
route(5);
}
/* example how to enable some additional event routes */
if (is_method("INVITE")) {
#t_on_branch("1");
t_on_reply("1");
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Presence route
/* uncomment the whole following route for enabling presence server */
route[2]
{
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;
# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==null)
{
sl_send_reply("404", "Not here");
exit;
}
return;
}
# Authentication route
/* uncomment the whole following route for enabling authentication */
route[3] {
if (is_method("REGISTER"))
{
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize("", "subscriber"))
{
www_challenge("", "0");
exit;
}
if ($au!=$tU)
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
}
# Auth only on registration
#a# } else {
#a# # authenticate if from local subscriber (uncomment to enable auth)
#a# if (from_uri==myself)
#a# {
#a# if (!proxy_authorize("", "subscriber")) {
#a# proxy_challenge("", "0");
#a# exit;
#a# }
#a# if (is_method("PUBLISH"))
#a# {
#a# if ($au!=$tU) {
#a# sl_send_reply("403","Forbidden
auth ID");
#a# exit;
#a# }
#a# } else {
#a# if ($au!=$fU) {
#a# sl_send_reply("403","Forbidden
auth ID");
#a# exit;
#a# }
#a# }
#a#
#a# consume_credentials();
#a# # caller authenticated
#a# }
#a# }
return;
}
# Caller NAT detection route
/* uncomment the whole following route for enabling Caller NAT Detection */
route[4]{
force_rport();
if (nat_uac_test("19")) {
if (method=="REGISTER") {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(5);
}
return;
}
# RTPProxy control
/* uncomment the whole following route for enabling RTPProxy Control */
route[5] {
if (is_method("BYE")) {
unforce_rtp_proxy();
} else if (is_method("INVITE")){
force_rtp_proxy();
}
if (!has_totag()) add_rr_param(";nat=yes");
return;
}
branch_route[1] {
xdbg("new branch at $ru\n");
}
onreply_route[1] {
xdbg("incoming reply\n");
if ((isflagset(5) || isbflagset(6))&&
status=~"(183)|(2[0-9][0-9])") {
force_rtp_proxy();
}
if (isbflagset(6)) {
fix_nated_contact();
}
}
failure_route[1] {
if (is_method("INVITE")
&& (isbflagset(6) || isflagset(5))) {
unforce_rtp_proxy();
}
if (t_was_cancelled()) {
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();
##}
}
**************************************************************************************************
**************************************************************************************************
Aqui va el SIP Trace para una llamada de telefonos fisicos NATed to NATed:
**************************************************************************************************
U +0.161561 212.4.97.115:35379 -> 212.4.107.250:5060
INVITE sip:20000000@212.4.107.250;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>
Contact:<sip:20000004@192.168.254.110:5060;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 332
v=0
o=20000004 8000 8000 IN IP4 192.168.254.110
s=SIP Call
c=IN IP4 192.168.254.110
t=0 0
m=audio 40000 RTP/AVP 4 3 18 0 8 9 97
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:60
#
U +0.000407 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP
192.168.254.110:5060;branch=z9hG4bK8f809670adc00668;rport=35379;received=212.4.97.115
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Server: Kamailio (1.5.1-notls (i386/linux))
Content-Length: 0
#
U +0.000034 212.4.107.250:5060 -> 212.4.97.115:55128
INVITE sip:20000000@192.168.254.101:5060;user=phone SIP/2.0
Record-Route:<sip:212.4.107.250;lr=on;nat=yes>
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>
Contact:<sip:20000004@212.4.97.115:35379;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 348
v=0
o=20000004 8000 8000 IN IP4 192.168.254.110
s=SIP Call
c=IN IP4 212.4.107.250
t=0 0
m=audio 35752 RTP/AVP 4 3 18 0 8 9 97
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:60
a=nortpproxy:yes
#
U +0.019311 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Content-Length: 0
#
U +0.030480 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact:<sip:20000000@192.168.254.101:5060;user=phone>
Record-Route:<sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Length: 0
#
U +0.000083 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact:<sip:20000000@192.168.254.101:5060;user=phone>
Record-Route:<sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Length: 0
#
U +6.510103 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact:<sip:20000000@192.168.254.101:5060;user=phone>
Record-Route:<sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
#
U +0.000365 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact:<sip:20000000@192.168.254.101:5060;user=phone>
Record-Route:<sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes
#
U +0.034122 212.4.97.115:35379 -> 212.4.107.250:5060
ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bKdf5e0ceed72f3797
Route:<sip:212.4.107.250;lr=on;nat=yes>
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Contact:<sip:20000004@192.168.254.110:5060;transport=udp;user=phone>
Supported: path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 ACK
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
#
U +0.000245 212.4.107.250:5060 -> 192.168.254.101:5060
ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.2
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bKdf5e0ceed72f3797
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Contact:<sip:20000004@212.4.97.115:35379;transport=udp;user=phone>
Supported: path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 ACK
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
#
U +0.458031 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact:<sip:20000000@192.168.254.101:5060;user=phone>
Record-Route:<sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
#
U +0.000246 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact:<sip:20000000@192.168.254.101:5060;user=phone>
Record-Route:<sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes
#
U +0.999724 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact:<sip:20000000@192.168.254.101:5060;user=phone>
Record-Route:<sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
#
U +0.000295 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact:<sip:20000000@192.168.254.101:5060;user=phone>
Record-Route:<sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes
_______________________________________________
SR-Users-ES mailing list
SR-Users-ES(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-es
1:21 p.m.
New subject: No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy)
El Jueves, 25 de Junio de 2009, ggb escribió:
En el INVITE reenviado de Kamailio al destino se ve la
IP privada en el
SDP.
Tal vez no esté entrando en el route[5] que es donde se hace el
force_rtp_proxy. Prueba a que se ejecute siempre el force_rtp_proxy en
los INVITEs y sus respuestas a ver si así te funciona en este caso que
te da problemas.
No, te equivocas, el SDP del INVITE que sale de Kamailio *sí* que tiene la IP
de media modificada:
--------------------------
v=0
o=20000004 8000 8000 IN IP4 192.168.254.110 <--- Esto da igual
s=SIP Call
c=IN IP4 212.4.107.250 <--- Esto es la que importa
t=0 0
m=audio 35752 RTP/AVP 4 3 18 0 8 9 97
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:60
a=nortpproxy:yes
--------------------------
--
Iñaki Baz Castillo <ibc(a)aliax.net>
1:18 p.m.
New subject: No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy)
El Jueves, 25 de Junio de 2009, rubenrojas - Trc.es escribió:
# Caller NAT detection route
/* uncomment the whole following route for enabling Caller NAT Detection */
route[4]{
force_rport();
if (nat_uac_test("19")) {
if (method=="REGISTER") {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(5);
}
return;
}
Lo anterior NO se está ejecutando. Fíjate que el Contact del 200 Ok llega con
la IP provada al llamante, por lo que el ACK que envía llega al proxy con el
RURI conteniendo una IP privada del llamado, así que el proxy ruta el ACK a la
IP privada (que obiviamente no llega). Por eso no oyes audio, porque la
negociación del INVITE no acaba (no llega el ACK al llamado, y por eso hay
tantas retransmissiones del 200 OK que no pareces haber observado).
# RTPProxy control
/* uncomment the whole following route for enabling RTPProxy Control */
route[5] {
if (is_method("BYE")) {
unforce_rtp_proxy();
} else if (is_method("INVITE")){
force_rtp_proxy();
}
if (!has_totag()) add_rr_param(";nat=yes");
return;
}
branch_route[1] {
xdbg("new branch at $ru\n");
}
onreply_route[1] {
xdbg("incoming reply\n");
if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])")
{
force_rtp_proxy();
}
if (isbflagset(6)) {
fix_nated_contact();
}
}
failure_route[1] {
if (is_method("INVITE")
&& (isbflagset(6) || isflagset(5))) {
unforce_rtp_proxy();
}
if (t_was_cancelled()) {
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();
##}
}
***************************************************************************
***********************
***************************************************************************
***********************
Aqui va el SIP Trace para una llamada de telefonos fisicos NATed to NATed:
***************************************************************************
*********************** U +0.161561 212.4.97.115:35379 -> 212.4.107.250:5060
INVITE sip:20000000@212.4.107.250;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668
From: "20000004"
<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To:
<sip:20000000@212.4.107.250;user=phone>
Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 332
v=0
o=20000004 8000 8000 IN IP4 192.168.254.110
s=SIP Call
c=IN IP4 192.168.254.110
t=0 0
m=audio 40000 RTP/AVP 4 3 18 0 8 9 97
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:60
#
U +0.000407 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP
192.168.254.110:5060;branch=z9hG4bK8f809670adc00668;rport=35379;received=21
2.4.97.115 From: "20000004"
<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To:
<sip:20000000@212.4.107.250;user=phone>
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Server: Kamailio (1.5.1-notls (i386/linux))
Content-Length: 0
#
U +0.000034 212.4.107.250:5060 -> 212.4.97.115:55128
INVITE sip:20000000@192.168.254.101:5060;user=phone SIP/2.0
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f8096
70adc00668 From: "20000004"
<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To:
<sip:20000000@212.4.107.250;user=phone>
Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 69
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 348
v=0
o=20000004 8000 8000 IN IP4 192.168.254.110
s=SIP Call
c=IN IP4 212.4.107.250
t=0 0
m=audio 35752 RTP/AVP 4 3 18 0 8 9 97
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:60
a=nortpproxy:yes
#
U +0.019311 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f8096
70adc00668 From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:
<sip:20000000@212.4.107.250;user=phone>
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Content-Length: 0
#
U +0.030480 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f8096
70adc00668 From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:
<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,
INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Length: 0
#
U +0.000083 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f8096
70adc00668 From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:
<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,
INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Length: 0
#
U +6.510103 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f8096
70adc00668 From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:
<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,
INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
#
U +0.000365 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f8096
70adc00668 From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:
<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,
INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes
#
U +0.034122 212.4.97.115:35379 -> 212.4.107.250:5060
ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bKdf5e0ceed72f3797
Route: <sip:212.4.107.250;lr=on;nat=yes>
From: "20000004"
<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To:
<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone>
Supported: path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 ACK
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
#
U +0.000245 212.4.107.250:5060 -> 192.168.254.101:5060
ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.2
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bKdf5e0c
eed72f3797 From: "20000004"
<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To:
<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone>
Supported: path
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 ACK
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 69
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
#
U +0.458031 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f8096
70adc00668 From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:
<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,
INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
#
U +0.000246 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f8096
70adc00668 From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:
<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,
INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes
#
U +0.999724 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f8096
70adc00668 From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:
<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,
INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
#
U +0.000295 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f8096
70adc00668 From:
"20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To:
<sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d(a)192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,
INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167
v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes
_______________________________________________
SR-Users-ES mailing list
SR-Users-ES(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-es
--
Iñaki Baz Castillo <ibc(a)aliax.net>
1:25 p.m.
New subject: No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy)
El Jueves, 25 de Junio de 2009, Iñaki Baz Castillo escribió:
El Jueves, 25 de Junio de 2009, rubenrojas - Trc.es
escribió:
Perdona, lo de arriba sí se está ejecutando (el Contact del INVITE cambia). Lo
que no cambia es el de la respuesta, o sea, no se ejecuta esto:
onreply_route[1] {
xdbg("incoming reply\n");
if ((isflagset(5) || isbflagset(6)) &&
status=~"(183)|(2[0-9][0-9])")
{
force_rtp_proxy();
}
if (isbflagset(6)) {
fix_nated_contact();
}
}
Lo que no sé es porqué no se ejecuta puesto que tienes puesto en
t_onreply("1") en route[1]. Eso tendrás que descubrirlo a base de poner
"xlog"
para ver si se están eejcutando los bloques o no. Nadie garantiza que el
config file que viene por defecto sea 100% correcto.
--
Iñaki Baz Castillo <ibc(a)aliax.net>
# Caller NAT detection route
/* uncomment the whole following route for enabling Caller NAT Detection
*/ route[4]{
force_rport();
if (nat_uac_test("19")) {
if (method=="REGISTER") {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(5);
}
return;
}
Lo anterior NO se está ejecutando. Fíjate que el Contact del 200 Ok llega
con la IP provada al llamante, por lo que el ACK que envía llega al proxy
con el RURI conteniendo una IP privada del llamado, así que el proxy ruta
el ACK a la IP privada (que obiviamente no llega). Por eso no oyes audio,
porque la negociación del INVITE no acaba (no llega el ACK al llamado, y
por eso hay tantas retransmissiones del 200 OK que no pareces haber
observado).
5412
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5412
days old
sr-users-es@lists.kamailio.org
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participants (3)
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ggb -
Iñaki Baz Castillo -
rubenrojas - Trc.es