Hi all,
I've the following infrastructure
(fake IPs)
62.128.128.68 (Telco SBC Inbound)
|
|
10.19.139.66 (Kamailio + RTPEngine)
|
|
10.19.139.69 (Asterisk 13)
|
|
62.128.128.68 (Telco SBC Outbound)
Test scenario is:
- a mobile calls a destination; the call comes in from Telco SBC Inbound,
goes through Kamailio+RTPengine and then to Asterisk which redirects the
call to Telco SBC Outbound.
- Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it
stays in the path
when a call comes in SBC Inbound, it is correctly redireted to SBC
Outbound. The call is answered, and after 5 or so seconds, SBC Inbound
sends a new INVITE to force codecs, which Asterisk replies correctly.
the problem is that between Kamailio and SBC Inbund, when Asterisk replies
to the second invite, the media port changes, resulting in audio loss:
during the call, audio from destination to originator works OK but
originator to destination is mute.
According to the telco's guys,this is due to the fact that Kamailio is
changing during call the media port from original media port to a new one,
on reply to the second Invite, and this is wrong and reason why sound from
the originator to destination is lost (but not from destination to
originator)
I'm lost too here.....
Any clue? Can anyone help? Resuming I need to make sure that, if during a
call some INVITES are sent from SBC to Asterisk, that the reply keeps the
same IP_ADDR:port from beginning to the end of the call. As it is today,
the second invite changes te RTP Media port and no audi is heard....
If needed I can send PCap files.
please find below my kamailio.cfg file
#!define WITH_NAT
#!define WITH_PSTN
/* enables Accounting Log functions */
#!define FLT_ACC 1
/* enable Accounting of missed or failed calls */
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
/* sets the dispatcher list file path */
#!define DS_LIST "/etc/kamailio/dispatcher.list"
/* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP
Proxy */
#!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000"
// was udp:localhost:60000
/* defines DB connection string */
#!ifndef DBURL
#!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio"
#!endif
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
####### Global Parameters #########
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif
#!define FLT_DISPATCH_SETID 1
#!define FLT_FS 10
#!define FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
fork=yes
children=4
/* comment the next line to enable TCP */
disable_tcp=no
/* uncomment the next line to disable the auto discovery of local aliases
based on revers DNS on IPs (default on) */
auto_aliases=no
/* add local domain aliases */
# alias="mysipserver.com"
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:10.19.139.66:5060
listen=udp:eth1:5060
sip_warning=no;
####### Modules Section ########
#set module path
mpath="/usr/lib64/kamailio/modules/"
loadmodule "db_mysql.so"
loadmodule "jsonrpcs.so"
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
/**************/
loadmodule "usrloc.so"
/**************/
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "acc.so"
loadmodule "path.so"
loadmodule "dispatcher.so"
loadmodule "rtpengine.so"
loadmodule "nathelper.so"
loadmodule "rtimer.so"
loadmodule "sqlops.so"
# ----------------- setting module-specific parameters ---------------
# ----- jsonrpcs params -----
modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo")
modparam("jsonrpcs", "pretty_format", 1)
# ----- 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)
# ----- acc params -----
modparam("acc", "failed_transaction_flag", 3)
modparam("acc",
"log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si")
# ----- acc params -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc",
"log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc",
"db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
//;calltype=$avp(calltype)")
# ----- usrloc params -----
/* enable DB persistency for location entries */
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
# ----- tm params -----
# ----- the TM module enables stateful processing of SIP requests
modparam("tm", "fr_timer", 2000)
modparam("tm", "fr_inv_timer", 40000)
# ----- dispatcher params -----
modparam("dispatcher", "list_file", DS_LIST)
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)")
modparam("dispatcher", "ds_hash_size", 9)
modparam("dispatcher", "dstid_avp", "$avp(dsdstid)")
modparam("path", "use_received", 1)
# ----- rtpproxy params -----
modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR)
modparam("nathelper", "received_avp", "$avp(s:rcv)")
/************************/
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:ping@kamailio.org")
# params needed for NAT traversal in other modules
modparam("usrloc", "nat_bflag", FLB_NATB)
/***********************/
modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;")
modparam("rtimer", "exec", "timer=cdr;route=CDRS")
modparam("sqlops", "sqlcon",
"ca=>mysql://kamailio:**********@*************/kamailio")
####### Routing Logic ########
# main request routing logic
route {
xlog("L_INFO","$rm from $si");
xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} ");
# OPTIONS requests without a username in the Request-URI but one
# of our domains or IPs are addressed to the proxy itself and
# can be answered statelessly.
if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" &&
$fU=="ping")// && strempty((a)ruri.user) && (uri == myself))
{
sl_send_reply("200","OK");
exit;
}
# per request initial checks
route(REQINIT);
rtpengine_manage();
# NAT detection
route(NATDETECT);
# handle requests within SIP dialogs
### only initial requests (no To tag)
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans()){
route(RELAY);
}
exit;
}
# handle retransmissions
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}
route(WITHINDLG);
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
xlog("L_INFO","Removing Headers");
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")){
xlog("L_INFO","Recording Route");
record_route();
if (is_method("INVITE")) {
if (has_body("application/sdp")) {
if (rtpengine_offer())
t_on_reply("1");
} else {
t_on_reply("2");
}
}
}
if (is_method("ACK") && has_body("application/sdp"))
rtpengine_answer();
# account only INVITEs
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
}
//route(SIPOUT);
xlog("L_INFO","Setting PRESENCE");
# handle presence related requests
route(PRESENCE);
# handle registrations
xlog("L_INFO","Handling REGISTRAR");
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# dispatch destinations
route(DISPATCH);
xlog("L_INFO","**********END*************");
}
onreply_route[1]
{
if (has_body("application/sdp"))
rtpengine_answer();
}
onreply_route[2]
{
if (has_body("application/sdp"))
rtpengine_offer();
}
# Per SIP request initial checks
route[REQINIT] {
xlog("L_INFO","REQINIT Starting");
if (!mf_process_maxfwd_header("10")) {
xlog("L_INFO","483 - Too Many Hops");
sl_send_reply("483","Too Many Hops");
exit;
}
if(!sanity_check("1511", "7"))
{
xlog("Sanity Check -> Malformed SIP message from
$si:$sp\n");
exit;
}
xlog("L_INFO","REQINIT Finishing");
}
/***********************/
# Caller NAT detection
route[NATDETECT] {
xlog("L_INFO","Entering NATDTECT");
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
xlog("L_INFO","Fix Nated Register");
fix_nated_register();
} else {
if(is_first_hop())
{
xlog("L_INFO","Set Contact Alias");
set_contact_alias();
}
}
xlog("L_INFO","Set FLT_NATS" + FLT_NATS);
setflag(FLT_NATS);
}
#!endif
xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS);
xlog("L_INFO","Finishing NATDETECT");
return;
}
/***********************/
# Handle requests within SIP dialogs
route[WITHINDLG] {
xlog("L_INFO","Entering WITHDLG");
if (!has_totag()) return;
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the
transaction fails
}
else if ( is_method("ACK") ) {
# ACK is forwarded statelessy
xlog("L_INFO","Going to NATMANAGE");
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
# Add Record-Route for in-dialog NOTIFY as per RFC
6665.
record_route();
}
route(RELAY);
exit;
}
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
route(RELAY);
exit;
} else {
# ACK without matching transaction. Try to route
anyway - being optimistic
# since it has at least a To Tag
route(RELAY);
exit;
}
}
sl_send_reply("404","Not here");
xlog("L_INFO","Finishing WITHINDLG");
exit;
}
# Routing to foreign domains
route[SIPOUT] {
xlog("L_INFO","Entering SIPOUT");
if (uri==myself)
{
xlog("L_INFO","URI is MySelf!");
return;
}
append_hf("P-hint: outbound\r\n");
xlog("L_INFO","Finishing SIPOUT");
route(RELAY);
exit;
}
# Wrapper for relaying requests
route[RELAY] {
xlog("L_INFO","******** RELAY *******");
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) {
xlog("L_INFO","branch_route NOT SET!");
t_on_branch("MANAGE_BRANCH");
}
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) {
xlog("L_INFO","onreply_route NOT SET!");
t_on_reply("MANAGE_REPLY");
}
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) {
xlog("L_INFO","failure_route NOT SET!");
t_on_failure("MANAGE_FAILURE");
}
}
if (!t_relay()) {
xlog("L_INFO","t_relay returns FALSE");
sl_reply_error();
}
exit;
}
# URI update for dialog requests
route[DLGURI] {
xlog("L_INFO","Entering DLGURI");
#!ifdef WITH_NAT
if(!isdsturiset()) {
xlog("L_INFO","Handle ruri ALIAS");
handle_ruri_alias();
}
#!endif
return;
}
# RTPProxy control and singaling updates for NAT traversal
route[NATMANAGE] {
xlog("L_INFO","Entering NATMANAGE");
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
xlog("L_INFO","nat=yes --- Setting FLB_NATB");
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) ))
{
xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE");
return;
}
xlog("L_INFO","Starting RTPEngine");
rtpengine_manage();
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
xlog("L_INFO","adding nat=yes");
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
if(is_first_hop())
{
xlog("L_INFO","Set Contact Alias");
set_contact_alias();
}
}
}
#!endif
return;
}
# Handle SIP registrations
route[REGISTRAR] {
if(!is_method("REGISTER"))
return;
add_path_received();
route(DISPATCH);
}
# Presence server route
route[PRESENCE] {
if(!is_method("PUBLISH|SUBSCRIBE"))
return;
sl_send_reply("404", "Not here");
exit;
}
# Dispatch requests
route[DISPATCH] {
xlog("L_INFO","Entering DISPATCH");
# hash over callerid dispatching on gateways group '1'
if(!ds_select_dst("1", "10","4"))
{
xlog("L_INFO","no destination selected from dispatcher list!");
send_reply("404", "No destination");
exit;
}
xlog("L_INFO","going to <$ru> via <$du>\n");
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
# Sample failure route
failure_route[RTF_DISPATCH] {
if (t_is_canceled()) {
exit;
}
xlog("L_INFO", "Media server $du failed to answer, selecting other one!");
# next DST - only for 500 or local timeout
if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) )
{
#we mark the destination Inactive and Probing
ds_mark_dst("ip");
if(ds_next_dst())
{
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
}
}
route[CDRS]{
sql_query("ca","call kamailio_cdrs();","rb");
sql_query("ca","call kamailio_rating('default');","rb");
}
Cheers,
*Sérgio *
Hi all,
I've the following infrastructure
62.128.128.68 (Telco SBC Inbound)
|
|
10.19.139.66 (Kamailio + RTPEngine)
|
|
10.19.139.69 (Asterisk 13)
|
|
62.128.128.68 (Telco SBC Outbound)
Test scenario is:
- a mobile calls a destination; the call comes in from Telco SBC Inbound,
goes through Kamailio+RTPengine and then to Asterisk which redirects the
call to Telco SBC Outbound.
- Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it
stays in the path
when a call comes in SBC Inbound, it is correctly redireted to SBC
Outbound. The call is answered, and after 5 or so seconds, SBC Inbound
sends a new INVITE to force codecs, which Asterisk replies correctly.
the problem is that between Kamailio and SBC Inbund, when Asterisk replies
to the second invite, the media port changes, resulting in audio loss:
during the call, audio from destination to originator works OK but
originator to destination is mute.
According to the telco's guys,this is due to the fact that Kamailio is
changing during call the media port from original media port to a new one,
on reply to the second Invite, and this is wrong and reason why sound from
the originator to destination is lost (but not from destination to
originator)
I'm lost too here.....
Any clue? Can anyone help? Resuming I need to make sure that, if during a
call some INVITES are sent from SBC to Asterisk, that the reply keeps the
same IP_ADDR:port from beginning to the end of the call. As it is today,
the second invite changes te RTP Media port and no audi is heard....
If needed I can send PCap files.
please find below my kamailio.cfg file
#!define WITH_NAT
#!define WITH_PSTN
/* enables Accounting Log functions */
#!define FLT_ACC 1
/* enable Accounting of missed or failed calls */
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
/* sets the dispatcher list file path */
#!define DS_LIST "/etc/kamailio/dispatcher.list"
/* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP
Proxy */
#!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000"
// was udp:localhost:60000
/* defines DB connection string */
#!ifndef DBURL
#!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio"
#!endif
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
####### Global Parameters #########
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif
#!define FLT_DISPATCH_SETID 1
#!define FLT_FS 10
#!define FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
fork=yes
children=4
/* comment the next line to enable TCP */
disable_tcp=no
/* uncomment the next line to disable the auto discovery of local aliases
based on revers DNS on IPs (default on) */
auto_aliases=no
/* add local domain aliases */
# alias="mysipserver.com"
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:10.19.139.66:5060
listen=udp:eth1:5060
sip_warning=no;
####### Modules Section ########
#set module path
mpath="/usr/lib64/kamailio/modules/"
loadmodule "db_mysql.so"
loadmodule "jsonrpcs.so"
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
/**************/
loadmodule "usrloc.so"
/**************/
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "acc.so"
loadmodule "path.so"
loadmodule "dispatcher.so"
loadmodule "rtpengine.so"
loadmodule "nathelper.so"
loadmodule "rtimer.so"
loadmodule "sqlops.so"
# ----------------- setting module-specific parameters ---------------
# ----- jsonrpcs params -----
modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo")
modparam("jsonrpcs", "pretty_format", 1)
# ----- 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)
# ----- acc params -----
modparam("acc", "failed_transaction_flag", 3)
modparam("acc",
"log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si")
# ----- acc params -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc",
"log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc",
"db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
//;calltype=$avp(calltype)")
# ----- usrloc params -----
/* enable DB persistency for location entries */
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
# ----- tm params -----
# ----- the TM module enables stateful processing of SIP requests
modparam("tm", "fr_timer", 2000)
modparam("tm", "fr_inv_timer", 40000)
# ----- dispatcher params -----
modparam("dispatcher", "list_file", DS_LIST)
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)")
modparam("dispatcher", "ds_hash_size", 9)
modparam("dispatcher", "dstid_avp", "$avp(dsdstid)")
modparam("path", "use_received", 1)
# ----- rtpproxy params -----
modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR)
modparam("nathelper", "received_avp", "$avp(s:rcv)")
/************************/
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:ping@kamailio.org")
# params needed for NAT traversal in other modules
modparam("usrloc", "nat_bflag", FLB_NATB)
/***********************/
modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;")
modparam("rtimer", "exec", "timer=cdr;route=CDRS")
modparam("sqlops", "sqlcon",
"ca=>mysql://kamailio:**********@*************/kamailio")
####### Routing Logic ########
# main request routing logic
route {
xlog("L_INFO","$rm from $si");
xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} ");
# OPTIONS requests without a username in the Request-URI but one
# of our domains or IPs are addressed to the proxy itself and
# can be answered statelessly.
if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" &&
$fU=="ping")// && strempty((a)ruri.user) && (uri == myself))
{
sl_send_reply("200","OK");
exit;
}
# per request initial checks
route(REQINIT);
rtpengine_manage();
# NAT detection
route(NATDETECT);
# handle requests within SIP dialogs
### only initial requests (no To tag)
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans()){
route(RELAY);
}
exit;
}
# handle retransmissions
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}
route(WITHINDLG);
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
xlog("L_INFO","Removing Headers");
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")){
xlog("L_INFO","Recording Route");
record_route();
if (is_method("INVITE")) {
if (has_body("application/sdp")) {
if (rtpengine_offer())
t_on_reply("1");
} else {
t_on_reply("2");
}
}
}
if (is_method("ACK") && has_body("application/sdp"))
rtpengine_answer();
# account only INVITEs
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
}
//route(SIPOUT);
xlog("L_INFO","Setting PRESENCE");
# handle presence related requests
route(PRESENCE);
# handle registrations
xlog("L_INFO","Handling REGISTRAR");
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# dispatch destinations to PSTN
//route(PSTN);
# dispatch destinations
route(DISPATCH);
xlog("L_INFO","**********END*************");
}
onreply_route[1]
{
if (has_body("application/sdp"))
rtpengine_answer();
}
onreply_route[2]
{
if (has_body("application/sdp"))
rtpengine_offer();
}
# Per SIP request initial checks
route[REQINIT] {
xlog("L_INFO","REQINIT Starting");
if (!mf_process_maxfwd_header("10")) {
xlog("L_INFO","483 - Too Many Hops");
sl_send_reply("483","Too Many Hops");
exit;
}
if(!sanity_check("1511", "7"))
{
xlog("Sanity Check -> Malformed SIP message from
$si:$sp\n");
exit;
}
xlog("L_INFO","REQINIT Finishing");
}
/***********************/
# Caller NAT detection
route[NATDETECT] {
xlog("L_INFO","Entering NATDTECT");
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
xlog("L_INFO","Fix Nated Register");
fix_nated_register();
} else {
if(is_first_hop())
{
xlog("L_INFO","Set Contact Alias");
set_contact_alias();
}
}
xlog("L_INFO","Set FLT_NATS" + FLT_NATS);
setflag(FLT_NATS);
}
#!endif
xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS);
xlog("L_INFO","Finishing NATDETECT");
return;
}
/***********************/
# Handle requests within SIP dialogs
route[WITHINDLG] {
xlog("L_INFO","Entering WITHDLG");
if (!has_totag()) return;
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the
transaction fails
}
else if ( is_method("ACK") ) {
# ACK is forwarded statelessy
xlog("L_INFO","Going to NATMANAGE");
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
# Add Record-Route for in-dialog NOTIFY as per RFC
6665.
record_route();
}
route(RELAY);
exit;
}
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
route(RELAY);
exit;
} else {
# ACK without matching transaction. Try to route
anyway - being optimistic
# since it has at least a To Tag
route(RELAY);
exit;
}
}
sl_send_reply("404","Not here");
xlog("L_INFO","Finishing WITHINDLG");
exit;
}
# Routing to foreign domains
route[SIPOUT] {
xlog("L_INFO","Entering SIPOUT");
if (uri==myself)
{
xlog("L_INFO","URI is MySelf!");
return;
}
append_hf("P-hint: outbound\r\n");
xlog("L_INFO","Finishing SIPOUT");
route(RELAY);
exit;
}
# Wrapper for relaying requests
route[RELAY] {
xlog("L_INFO","******** RELAY *******");
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) {
xlog("L_INFO","branch_route NOT SET!");
t_on_branch("MANAGE_BRANCH");
}
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) {
xlog("L_INFO","onreply_route NOT SET!");
t_on_reply("MANAGE_REPLY");
}
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) {
xlog("L_INFO","failure_route NOT SET!");
t_on_failure("MANAGE_FAILURE");
}
}
if (!t_relay()) {
xlog("L_INFO","t_relay returns FALSE");
sl_reply_error();
}
exit;
}
# URI update for dialog requests
route[DLGURI] {
xlog("L_INFO","Entering DLGURI");
#!ifdef WITH_NAT
if(!isdsturiset()) {
xlog("L_INFO","Handle ruri ALIAS");
handle_ruri_alias();
}
#!endif
return;
}
# RTPProxy control and singaling updates for NAT traversal
route[NATMANAGE] {
xlog("L_INFO","Entering NATMANAGE");
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
xlog("L_INFO","nat=yes --- Setting FLB_NATB");
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) ))
{
xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE");
return;
}
xlog("L_INFO","Starting RTPEngine");
rtpengine_manage();
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
xlog("L_INFO","adding nat=yes");
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
if(is_first_hop())
{
xlog("L_INFO","Set Contact Alias");
set_contact_alias();
}
}
}
#!endif
return;
}
# Handle SIP registrations
route[REGISTRAR] {
if(!is_method("REGISTER"))
return;
add_path_received();
route(DISPATCH);
}
# Presence server route
route[PRESENCE] {
if(!is_method("PUBLISH|SUBSCRIBE"))
return;
sl_send_reply("404", "Not here");
exit;
}
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
xlog("L_INFO","Entering PSTN");
# route to PSTN dialed numbers starting with '+' or '0000'
# (international format)
# - update the condition to match your dialing rules for PSTN routing
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
{
xlog("L_INFO","$rU is not outbound call to PSTN");
return;
}
if (!starts_with("$si","10.19.139"))
{
xlog("L_INFO","$rU is not comming from any Media GW");
return;
}
# only local users allowed to call
if(from_uri!=myself && !ds_is_from_list("1") )
{
xlog("L_INFO", "from_uri $fu != myself, so this is Not
Allowed!!");
sl_send_reply("403", "Not Allowed");
exit;
}
xlog("L_INFO","$si request");
xlog("L_INFO","$rU is an outbound call to PSTN");
$ru = "sip:" + $rU + "@62.28.173.10"; // + $sel(cfg_get.pstn.gw_ip);
route(RELAY);
/*
if(!ds_select_dst("20", "4"))
{
xlog("L_INFO","no destination selected from dispatcher list!");
send_reply("404", "No destination");
exit;
}
xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
t_on_failure("RTF_DISPATCH");
route(RELAY);
*/
exit;
#!endif
xlog("L_INFO"," NO PSTN SET!!");
return;
}
# Dispatch requests
route[DISPATCH] {
xlog("L_INFO","Entering DISPATCH");
# hash over callerid dispatching on gateways group '1'
if(!ds_select_dst("1", "10","4"))
{
xlog("L_INFO","no destination selected from dispatcher list!");
send_reply("404", "No destination");
exit;
}
xlog("L_INFO","going to <$ru> via <$du>\n");
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
# Dispatch requests
route[DISPATCH_TO_PSTN] {
# round robin dispatching on gateways group '20' - PSTN
if(!ds_select_dst("20", "10"))
{
send_reply("404", "No destination");
exit;
}
xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
# Sample failure route
failure_route[RTF_DISPATCH] {
if (t_is_canceled()) {
exit;
}
xlog("L_INFO", "Media server $du failed to answer, selecting other one!");
# next DST - only for 500 or local timeout
if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) )
{
#we mark the destination Inactive and Probing
ds_mark_dst("ip");
if(ds_next_dst())
{
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
}
}
route[CDRS]{
sql_query("ca","call kamailio_cdrs();","rb");
sql_query("ca","call kamailio_rating('default');","rb");
}
Cheers,
*Sérgio *
Hi ppl,
Say we want to allow 1 contact per AOR.
modparam("registrar", "max_contacts", 1)
Say some UA looses connection before de-registering, then comes back
shortly after (before Contact expires) and tries to register with a
different Contact, but gets rejected with error "test_max_contacts(): too
many contacts for AOR".
What's the approach to handle this situation? I'd like the registrar to
accept and save that AoR based on the latest registration. Similar to how
Asterisk's single-contact stack `chan_sip` works - whoever was the latest
to register successfully gets to be the winner.
Thanks,
--Sergiu
Hello,
I just activated snmpstats module on Kamailio 5.0.7 and added the
following line to route(INITIAL) at the bottom of the route like this
(For noc group to monitor Kamailio dialogs):
# Per SIP request initial checks
route[INITIAL] {
...
dlg_manage();
}
Now I started to gets this error few times in an hour:
/usr/sbin/kamailio[22194]: CRITICAL: dialog [dlg_timer.c:200]:
update_dlg_timer(): Trying to update a bogus dlg tl=0x7f798ca51b98
tl->next=(nil) tl->prev=(nil) Apr 15 09:54:05
/usr/sbin/kamailio[22194]: ERROR: dialog [dlg_handlers.c:1439]:
dlg_onroute(): failed to update dialog lifetime
I see no call failures because of this but still interested what is
happening here?
--Olli Attila
Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote:
> HI i have install Kamailio 5.1.2 on Ubuntu 18.4
> there is a audio issue , when we call call from one extension to another ,,
> in bigning call goes good , after 9-10 sec it miss audio for one side (one
> side can hear other side can't hear )
Hi, mailing list sr-dev is for Kamailio development related topics,
please subscribe to sr-users for more general queries.
Before posting, make sure you have described in detail your environment
and also troubleshooting steps you have done so far:
- what modifications you made to default configuration
- what happens during SIP transactions in fail scenario
- are you using RTP relay or not (rtpproxy/rtpengine)
- what have you observed from packet captures
Good luck
--
Mikko
Hi everybody;
I'm trying to use db_cluster to connect my kamailio installation to a mariadb galera cluster,
and it's driving me crazy.
In my lab it works without issues, but when I try the setup in the production node, I get
some strange errors:
Apr 11 14:21:31 ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): driver
error on query: Commands out of sync; you can't run this command now (2014)
Apr 11 14:21:31 ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while
submitting query
This is my configuration:
modparam("db_cluster", "connection","con1=>mysql://openser:PASSWD@X.X.X.30/
openser")
modparam("db_cluster", "connection","con2=>mysql://openser:PASSWD@X.X.X.31/
openser")
modparam("db_cluster", "connection","con3=>mysql://openser:PASSWD@X.X.X.32/
openser")
modparam("db_cluster", "cluster", "cls1=>con1=9r8r;con2=9r8r;con3=9r8r")
#!define DBURL "cluster://cls1"
#!define SQLOPSURL "database=>cluster://cls1"
modparam("dispatcher", "db_url", DBURL)
modparam("acc", "db_url", DBURL)
modparam("usrloc", "db_url", DBURL)
modparam("auth_db", "db_url", DBURL)
modparam("permissions", "db_url", DBURL)
modparam("alias_db", "db_url", DBURL)
modparam( "sqlops", "sqlcon", SQLOPSURL )
I'm using version 5.1.4 of kamailio, freshly compiled from source on a debian 9.
Any ideas?
*Simone Lazzaris*
*Qcom S.p.A.*
simone.lazzaris(a)qcom.it[1] | www.qcom.it[2]
* LinkedIn[3]* | *Facebook[4]*
[5]
--------
[1] mailto:simone.lazzaris@qcom.it
[2] https://www.qcom.it
[3] https://www.linkedin.com/company/qcom-spa
[4] http://www.facebook.com/qcomspa
[5] https://www.qcom.it/includes/email-banner.gif
hi, i try to follow the how to in the wiki
https://kb.asipto.com/kamailio:presence:k43-blf and configure my two
only phones..
but for any states only one led its blinking, the busy state.. no
other state work, (if i connect directly to asterisk all work
perfeclty, i using asterisk 13 with kamailio 4.4.4)
also read the archived
https://lists.kamailio.org/pipermail/sr-users/2013-May/078234.html but
no luck
wht happened with that how to? it's different for asterisk if are
integrated? do i need a pjsip asterisk event the oldes chan?
see that in presense module parameters, the server address said that
value of Contact header filed for 200 OK replies to SUBSCRIBE and
PUBLISH and in NOTIFY messages
umm if i have integrated and working asterisk (relatime mode) with
kamailio, that address muswt be asterisk? taking in cosideration that
i have a route rule said that send auth to asterisk and trust any
incomming from it?
--
Lenz McKAY Gerardo (PICCORO)
http://qgqlochekone.blogspot.com
Hey guys,
Question.
I have this block of code at the beginning of route block:
if(is_ip($td))
{
xlog("L_INFO", "[$ci][$mi][CSeq $cs] IP detected in To
domain. Resolving. \n");
exec_avp("host $td | awk '{print substr($$5, 1,
length($$5)-1)}'", "$avp(s:newtd)");
xlog("L_INFO", "[$ci][$mi][CSeq $cs] new newtd =
$avp(s:newtd)");
$td = $avp(s:newtd);
xlog("L_INFO", "[$ci][$mi][CSeq $cs] new td = $td");
}
but $td never updates.
Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO:
<script>: [8ROWdUOINx][1]*[CSeq* 20] IP detected in To domain. Resolving.
Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO:
<script>: [8ROWdUOINx][1]*[CSeq* 20] new newtd =
sjomaintpsg50.fuzemeeting.com
Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO:
<script>: [8ROWdUOINx][1]*[CSeq* 20] new td = 206.81.191.30
Is there a reason why?
Thanks
--
Andy Chen
Sr. Telephony Lead Engineer
achen@ <achen(a)thinkingphones.com>fuze.com
--
*Confidentiality Notice: The information contained in this e-mail and any
attachments may be confidential. If you are not an intended recipient, you
are hereby notified that any dissemination, distribution or copying of this
e-mail is strictly prohibited. If you have received this e-mail in error,
please notify the sender and permanently delete the e-mail and any
attachments immediately. You should not retain, copy or use this e-mail or
any attachment for any purpose, nor disclose all or any part of the
contents to any other person. Thank you.*
Hello,
Kamailio SIP Server v5.1.8 stable release is out.
This is a maintenance release of the previous stable branch, 5.1, that
includes fixes since the release of v5.1.7. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.1.x. Deployments running previous v5.1.x
versions are strongly recommended to be upgraded to v5.1.8 or to v5.2.x
series.
For more details about version 5.1.8 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Note: the branch 5.1 is the previous stable branch. The latest stable
branch is 5.2, at this time with v5.2.2 being released out of it.
Many thanks to all contributing and using Kamailio!
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com
