[Kamailio-Users] No Audio with clients Behind NAT, audio is fine with clients using public IPs (Im using rtpproxy)
25 Jun
2009
25 Jun
'09
12:41 p.m.
Hello everyone, this is my first post on this list,
I have installed kamailio 1.5.1 and set up a vanilla default kamailio.cfg, then I have
modified the cfg to activate mysql, domain, presence, nathelper and authentication with
md5, everything works as supposed to, and the clients can register, send txt messages and
talk to each other. The only problem is with the audio when the two clients are behind a
NAT, the phones can make a call and it does ring too, but when you pick up there is no
audio both ways.
when the phones have a public IP everything goes fine, it also works when I use a Linksys
PAP2T whith the options to "Insert VIA received", "Insert VIA rport",
"Handle VIA received", "Handle VIA rport" and "NAT mapping
enable" turned on, with the Qutecom softphone works too.
This is happening with thomson phones (model ST 2022), and GrandStream Budge Tone 200, it
happens no matter what options I set for NATting on the phones, I've even used stun
with stunserver.org or the ekiga stunserver, the phones register and can make and recieve
calls, but there is no audio when you pick up the call.
With a kamctl ul show, you can see that the phones have registered the Contact with their
local IPs and the Received have the public IPs and ports for the NAT
The only difference with the working Linksys is that they register the Contact with the
public IP.
Here you can see two NATed phones on the 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
Im using rtpproxy and there is no log error that indicates that rttpproxy isn't
working, in fact doing a SIP trace shows rtpproxy setting ports for the audio.
I run rtpproxy with this command:
rtpproxy -l 212.4.107.250 -s udp:localhost:7722 -F
Any help would be greatly appreciated, I've been two weeks looking for a solution
Im attaching my kamailio.cfg so you can take a look, at the end of the message Im gonna
attache the SIP Trace of a call between two NATed phones (a Thomson and a GrandStream) in
case anyone can help me decypher whats wrong here:
this is my cfg file
**************************************************************************************************
#
# $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();
##}
}
**************************************************************************************************
**************************************************************************************************
And here goes the SIP Trace for a NATed to NATed hardphones:
**************************************************************************************************
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
26 Jun
26 Jun
12:32 p.m.
New subject: [Kamailio-Users] No Audio with clients Behind NAT, audio is fine with clients using public IPs (Im using rtpproxy)
I've just setup a similar configuration as yours.
Could you check the SDP payload of the SIP INVITE that goes from kamailio to
the callee ? It should contains the IP address of rtpproxy and the port
allocated by the proxy.
Regards,
Pascal
On Thu, Jun 25, 2009 at 12:41 PM, rubenrojas - Trc.es <rubenrojas(a)trc.es>wrote;wrote:
Hello everyone, this is my first post on this list,
I have installed kamailio 1.5.1 and set up a vanilla default kamailio.cfg,
then I have modified the cfg to activate mysql, domain, presence, nathelper
and authentication with md5, everything works as supposed to, and the
clients can register, send txt messages and talk to each other. The only
problem is with the audio when the two clients are behind a NAT, the phones
can make a call and it does ring too, but when you pick up there is no audio
both ways.
when the phones have a public IP everything goes fine, it also works when I
use a Linksys PAP2T whith the options to "Insert VIA received", "Insert
VIA
rport", "Handle VIA received", "Handle VIA rport" and "NAT
mapping enable"
turned on, with the Qutecom softphone works too.
This is happening with thomson phones (model ST 2022), and GrandStream
Budge Tone 200, it happens no matter what options I set for NATting on the
phones, I've even used stun with stunserver.org or the ekiga stunserver,
the phones register and can make and recieve calls, but there is no audio
when you pick up the call.
With a kamctl ul show, you can see that the phones have registered the
Contact with their local IPs and the Received have the public IPs and ports
for the NAT
The only difference with the working Linksys is that they register the
Contact with the public IP.
Here you can see two NATed phones on the 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
Im using rtpproxy and there is no log error that indicates that rttpproxy
isn't working, in fact doing a SIP trace shows rtpproxy setting ports for
the audio.
I run rtpproxy with this command:
rtpproxy -l 212.4.107.250 -s udp:localhost:7722 -F
Any help would be greatly appreciated, I've been two weeks looking for a
solution
Im attaching my kamailio.cfg so you can take a look, at the end of the
message Im gonna attache the SIP Trace of a call between two NATed phones (a
Thomson and a GrandStream) in case anyone can help me decypher whats wrong
here:
this is my cfg file
**************************************************************************************************
#
# $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<sip%3Apinger@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();
##}
}
**************************************************************************************************
**************************************************************************************************
And here goes the SIP Trace for a NATed to NATed hardphones:
**************************************************************************************************
U +0.161561 212.4.97.115:35379 -> 212.4.107.250:5060
INVITE sip:20000000@212.4.107.250 <sip%3A20000000(a)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<sip%3A20000004@212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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<sip%3A20000004@212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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<sip%3A20000004@212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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
<sip%3A20000004(a)212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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
<sip%3A20000004(a)212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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
<sip%3A20000004(a)212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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
<sip%3A20000004(a)212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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
<sip%3A20000004(a)212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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<sip%3A20000004@212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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<sip%3A20000004@212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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
<sip%3A20000004(a)212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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
<sip%3A20000004(a)212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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
<sip%3A20000004(a)212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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
<sip%3A20000004(a)212.4.107.250>
;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000@212.4.107.250 <sip%3A20000000(a)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
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12:38 p.m.
New subject: [Kamailio-Users] No Audio with clients Behind NAT, audio is fine with clients using public IPs (Im using rtpproxy)
I've just setup a similar configuration as yours.
Could you check the SDP payload of the SIP INVITE that goes from kamailio to
the callee ? It should contains the IP address of rtpproxy and the port
allocated by the proxy.
Regards,
Pascal
On Thu, Jun 25, 2009 at 12:41 PM, rubenrojas - Trc.es <rubenrojas(a)trc.es>wrote;wrote:
Hello everyone, this is my first post on this list,
I have installed kamailio 1.5.1 and set up a vanilla default kamailio.cfg,
then I have modified the cfg to activate mysql, domain, presence, nathelper
and authentication with md5, everything works as supposed to, and the
clients can register, send txt messages and talk to each other. The only
problem is with the audio when the two clients are behind a NAT, the phones
can make a call and it does ring too, but when you pick up there is no audio
both ways.
when the phones have a public IP everything goes fine, it also works when I
use a Linksys PAP2T whith the options to "Insert VIA received", "Insert
VIA
rport", "Handle VIA received", "Handle VIA rport" and "NAT
mapping enable"
turned on, with the Qutecom softphone works too. [...]
6:05 p.m.
New subject: [Kamailio-Users] No Audio with clients Behind NAT, audio is fine with clients using public IPs (Im using rtpproxy)
6:11 p.m.
New subject: [Kamailio-Users] No Audio with clients Behind NAT, audio is fine with clients using public IPs (Im using rtpproxy)
6:12 p.m.
New subject: [Kamailio-Users] No Audio with clients Behind NAT, audio is fine with clients using public IPs (Im using rtpproxy)
5427
days inactive
5428
days old
5 comments
5 participants
participants (5)
-
hh174 -
olivier taylor -
olivier.taylor＠gmail.com
-
Pascal Maugeri -
rubenrojas - Trc.es