[SR-Users-ES] No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy)
rubenrojas - Trc.es
rubenrojas en trc.es
Jue Jun 25 13:01:47 CEST 2009
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 en 212.4.107.250
Contact:: sip:20000004 en 192.168.254.110:5060;transport=udp;user=phone Q=
Expires:: 1150
Callid:: 72ed03f6d2f390f9 en 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 en 212.4.107.250
Contact:: sip:20000000 en 192.168.254.101:5060;user=phone Q=
Expires:: 2945
Callid:: 17fe-c0a80101-5-1 en 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 en 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 en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>
Contact: <sip:20000004 en 192.168.254.110:5060;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: c177cae013da224d en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>
Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>
Contact: <sip:20000004 en 212.4.97.115:35379;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: c177cae013da224d en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>
Call-ID: c177cae013da224d en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d en 192.168.254.110
CSeq: 29653 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d en 192.168.254.110
CSeq: 29653 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d en 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 en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
Contact: <sip:20000004 en 192.168.254.110:5060;transport=udp;user=phone>
Supported: path
Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
Contact: <sip:20000004 en 212.4.97.115:35379;transport=udp;user=phone>
Supported: path
Call-ID: c177cae013da224d en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d en 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 en 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
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