18 Jul
2020
18 Jul
'20
6:11 p.m.
Hello,
I'm setting up a Kamailio MSFT Teams SBC that is connected to our SIP trunk
provider. This server is directly connected to the Internet and not behind
a NAT routing.
It is successfully processing invites from both sides (MSFT Teams and SIP
trunk). Unfortunately the ACKs and CANCEL messages are not relaying to
Teams. In the sipdump i can't find these messages.
Does anyone see what I am doing wrong? Below you can find my config
#!KAMAILIO
####### Defined Values #########
#!define MULTIDOMAIN 0
# - flags
# FLT_ - per transaction (message) flags
# FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
#!define FROM_TEAMS 11
#!define FROM_PBX 12
######## Define Modules ###########
#!define WITH_RTPENGINE
#!define WITH_TLS
#!define WITH_SIPDUMP
#!define WITH_DISPATCH
####### Global Parameters #########
### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR
debug=2
log_stderror=no
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "
/* number of SIP routing processes */
children=2
/* uncomment the next line to disable TCP (default on) */
# disable_tcp=yes
alias=SBC_FQDN
/* listen addresses */
listen=udp:PUBLIC_IP:5060 advertise SBC_FQDN:5060
#!ifdef WITH_TLS
listen=tls:PUBLIC_IP:5061 advertise SBC_FQDN:5061
#!endif
server_header= "Server: ABC SBC"
user_agent_header= "User-Agent: ABC SBC"
###### TLS Enable ######
#!ifdef WITH_TLS
enable_tls=yes
#tcp_connect_timeout=1000
tcp_accept_no_cl=yes
tcp_async = yes
tcp_connection_lifetime=600
/* upper limit for TLS connections */
tls_max_connections=2048
#!endif
####### Custom Parameters #########
/* These parameters can be modified runtime via RPC interface
* - see the documentation of 'cfg_rpc' module.
*
* Format: group.id = value 'desc' description
* Access: $sel(cfg_get.group.id) or @cfg_get.group.id */
####### Modules Section ########
/* set paths to location of modules */
loadmodule "jsonrpcs.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.so"
#!ifdef WITH_RTPENGINE
loadmodule "rtpengine.so"
#!endif
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
#!ifdef WITH_SIPDUMP
loadmodule "sipdump.so"
#!endif
#!ifdef WITH_DISPATCH
loadmodule "dispatcher.so"
#!endif
# ----------------- setting module-specific parameters ---------------
#!ifdef WITH_RTPENGINE
# ----- rtpengine params -----+
modparam("rtpengine", "rtpengine_sock",
"udp:127.0.0.1:12222")
#!endif
#!ifdef WITH_TLS
# ----- tls params -----+
modparam("tls", "xavp_cfg", "tls")
modparam("tls", "config", "/etc/kamailio/tls.cfg")
modparam("tls", "connection_timeout", 10)
modparam("tls", "ssl_release_buffers", 1)
modparam("tls", "send_close_notify", 1)
modparam("tls", "session_cache", 0)
#!endif
#!ifdef WITH_SIPDUMP
modparam("sipdump", "enable", 1)
#!endif
#!ifdef WITH_DISPATCH
#---------- dispatch
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_ping_interval", 300)
#!endif
# ----- jsonrpcs params -----
modparam("jsonrpcs", "pretty_format", 1)
/* set the path to RPC fifo control file */
# modparam("jsonrpcs", "fifo_name",
"/run/kamailio/kamailio_rpc.fifo")
/* set the path to RPC unix socket control file */
# modparam("jsonrpcs", "dgram_socket",
"/run/kamailio/kamailio_rpc.sock")
# ----- ctl params -----
/* set the path to RPC unix socket control file */
# modparam("ctl", "binrpc",
"unix:/run/kamailio/kamailio_ctl")
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
# ----- rr params -----
# set next param to 1 to add value to ;lr param (helps with some UAs)
modparam("rr", "enable_full_lr", 0)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)
# ----- 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 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", "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=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
# per request initial checks
route(REQINIT);
#check who is the sender
route(INITCHECK);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle retransmissions
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")) {
record_route();
}
# account only INVITEs
if (is_method("INVITE")) {
setflag(FLT_ACC); # do accounting
}
if ($rU==$null) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# update $du to set the destination address for proxying
#$du = "sip:" + $rd + ":9";
route(RELAY);
exit;
}
route[INITCHECK] {
if(from_uri =~ ".*microsoft.com")
{
setflag(FROM_TEAMS);
$du = "sip:" + "PBX_IP";
route(HANDLE_RTP_FROM_TEAMS);
} else if(from_uri =~ ".*" + "PBX_IP")
{
setflag(FROM_PBX);
$du="sip:sip.pstnhub.microsoft.com;transport=tls";
route(HANDLE_RTP_FROM_PBX);
} else {
exit;
}
}
#Manage RTP & transcoding comming from Teams to PBX
route[HANDLE_RTP_FROM_TEAMS] {
if (has_body("application/sdp"))
{
t_on_reply("PBX_REPLY_TO_TEAMS");
rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA
replace-origin replace-session-connection ICE=remove");
record_route();
t_relay_to_udp("PBX_IP","5060");
}
}
#Manage RTP & transcoding comming from PBX to Teams
route[HANDLE_RTP_FROM_PBX] {
if (has_body("application/sdp"))
{
t_on_reply("TEAMS_REPLY_TO_PBX");
rtpengine_manage("SRTP codec-mask=all ICE=force codec-transcode=PCMA
replace-origin replace-session-connection");
record_route_preset("SBC_FQDN:5061;transport=tls");
add_rr_param(";r2=on");
$rd = "sip.pstnhub.microsoft.com";
$td = "SBC_FQDN";
$fd = "SBC_FQDN";
#Set TLS SNI (server name & server id)
$xavp(tls=>server_name) = "SBC_FQDN";
$xavp(tls=>server_id) = "SBC_FQDN";
t_relay();
}
}
# Wrapper for relaying requests
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Per SIP request initial checks
route[REQINIT] {
if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
# silent drop for scanners - uncomment next line if want to reply
# sl_send_reply("200", "OK");
exit;
}
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if(is_method("OPTIONS")) {
sl_send_reply("200","Keepalive");
exit;
}
if(!sanity_check("1511", "7")) {
xlog("Malformed SIP message from $si:$sp\n");
exit;
}
}
# Handle requests within SIP dialogs
route[WITHINDLG] {
if (!has_totag()) return;
#Teams reINVITEs
if(isflagset(FROM_TEAMS)) {
t_relay_to_udp("PBX_IP","5060");
exit;
}
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
#set coresponding cert on transactions
if($fd == "SBC_FQDN") {
$xavp(tls=>server_name) = "SBC_FQDN";
$xavp(tls=>server_id) = "SBC_FQDN";
}
} else if ( is_method("NOTIFY") ) {
# Add Record-Route for in-dialog NOTIFY as per RFC 6665.
record_route();
}
route(RELAY);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
route(RELAY);
exit;
} else {
# ACK without matching transaction ... ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
exit;
}
# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
}
# Manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
}
#PBX On Reply
onreply_route[PBX_REPLY_TO_TEAMS]
{
if (has_body("application/sdp"))
{
rtpengine_manage("SRTP codec-mask=all codec-transcode=PCMA
replace-origin replace-session-connection media-address=PUBLIC_IP");
}
}
#From Teams On Reply
onreply_route[TEAMS_REPLY_TO_PBX]
{
if (has_body("application/sdp"))
{
rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA
replace-origin replace-session-connection media-address=PUBLIC_IP");
}
}
# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
if (t_is_canceled()) exit;
}
event_route[tm:local-request] {
if(is_method("OPTIONS") && $ru =~
"pstnhub.microsoft.com") {
append_hf("Contact:
<sip:SBC_FQDN:5061;transport=tls>\r\n");
}
xlog("L_INFO", "Sent out tm request: $mb\n");
}
20 Jul
20 Jul
3:44 p.m.
Could be anything, but most likely you aren't receiving those in Kam
because the PBX sends them somewhere else, check where the PBX sends those,
check the Contact URI, that's the address where ACKs will be sent out.
The SBC for Teams article suggests a double header record_route_preset()
for transport conversion, not sure why your config is different.
Also, at the very least, the config seems to invoke
*record_route[_preset]()* twice per INVITE, it shouldn't be like that.
Good luck.
On Sun, Jul 19, 2020 at 3:55 AM Daniel van der Tang <tangd122(a)gmail.com>
wrote:
Hello,
I'm setting up a Kamailio MSFT Teams SBC that is connected to our SIP
trunk provider. This server is directly connected to the Internet and not
behind a NAT routing.
It is successfully processing invites from both sides (MSFT Teams and SIP
trunk). Unfortunately the ACKs and CANCEL messages are not relaying to
Teams. In the sipdump i can't find these messages.
Does anyone see what I am doing wrong? Below you can find my config
#!KAMAILIO
####### Defined Values #########
#!define MULTIDOMAIN 0
# - flags
# FLT_ - per transaction (message) flags
# FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
#!define FROM_TEAMS 11
#!define FROM_PBX 12
######## Define Modules ###########
#!define WITH_RTPENGINE
#!define WITH_TLS
#!define WITH_SIPDUMP
#!define WITH_DISPATCH
####### Global Parameters #########
### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR
debug=2
log_stderror=no
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "
/* number of SIP routing processes */
children=2
/* uncomment the next line to disable TCP (default on) */
# disable_tcp=yes
alias=SBC_FQDN
/* listen addresses */
listen=udp:PUBLIC_IP:5060 advertise SBC_FQDN:5060
#!ifdef WITH_TLS
listen=tls:PUBLIC_IP:5061 advertise SBC_FQDN:5061
#!endif
server_header= "Server: ABC SBC"
user_agent_header= "User-Agent: ABC SBC"
###### TLS Enable ######
#!ifdef WITH_TLS
enable_tls=yes
#tcp_connect_timeout=1000
tcp_accept_no_cl=yes
tcp_async = yes
tcp_connection_lifetime=600
/* upper limit for TLS connections */
tls_max_connections=2048
#!endif
####### Custom Parameters #########
/* These parameters can be modified runtime via RPC interface
* - see the documentation of 'cfg_rpc' module.
*
* Format: group.id = value 'desc' description
* Access: $sel(cfg_get.group.id) or @cfg_get.group.id */
####### Modules Section ########
/* set paths to location of modules */
loadmodule "jsonrpcs.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.so"
#!ifdef WITH_RTPENGINE
loadmodule "rtpengine.so"
#!endif
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
#!ifdef WITH_SIPDUMP
loadmodule "sipdump.so"
#!endif
#!ifdef WITH_DISPATCH
loadmodule "dispatcher.so"
#!endif
# ----------------- setting module-specific parameters ---------------
#!ifdef WITH_RTPENGINE
# ----- rtpengine params -----+
modparam("rtpengine", "rtpengine_sock",
"udp:127.0.0.1:12222")
#!endif
#!ifdef WITH_TLS
# ----- tls params -----+
modparam("tls", "xavp_cfg", "tls")
modparam("tls", "config", "/etc/kamailio/tls.cfg")
modparam("tls", "connection_timeout", 10)
modparam("tls", "ssl_release_buffers", 1)
modparam("tls", "send_close_notify", 1)
modparam("tls", "session_cache", 0)
#!endif
#!ifdef WITH_SIPDUMP
modparam("sipdump", "enable", 1)
#!endif
#!ifdef WITH_DISPATCH
#---------- dispatch
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_ping_interval", 300)
#!endif
# ----- jsonrpcs params -----
modparam("jsonrpcs", "pretty_format", 1)
/* set the path to RPC fifo control file */
# modparam("jsonrpcs", "fifo_name",
"/run/kamailio/kamailio_rpc.fifo")
/* set the path to RPC unix socket control file */
# modparam("jsonrpcs", "dgram_socket",
"/run/kamailio/kamailio_rpc.sock")
# ----- ctl params -----
/* set the path to RPC unix socket control file */
# modparam("ctl", "binrpc",
"unix:/run/kamailio/kamailio_ctl")
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
# ----- rr params -----
# set next param to 1 to add value to ;lr param (helps with some UAs)
modparam("rr", "enable_full_lr", 0)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)
# ----- 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 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", "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=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
# per request initial checks
route(REQINIT);
#check who is the sender
route(INITCHECK);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle retransmissions
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")) {
record_route();
}
# account only INVITEs
if (is_method("INVITE")) {
setflag(FLT_ACC); # do accounting
}
if ($rU==$null) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# update $du to set the destination address for proxying
#$du = "sip:" + $rd + ":9";
route(RELAY);
exit;
}
route[INITCHECK] {
if(from_uri =~ ".*microsoft.com")
{
setflag(FROM_TEAMS);
$du = "sip:" + "PBX_IP";
route(HANDLE_RTP_FROM_TEAMS);
} else if(from_uri =~ ".*" + "PBX_IP")
{
setflag(FROM_PBX);
$du="sip:sip.pstnhub.microsoft.com;transport=tls";
route(HANDLE_RTP_FROM_PBX);
} else {
exit;
}
}
#Manage RTP & transcoding comming from Teams to PBX
route[HANDLE_RTP_FROM_TEAMS] {
if (has_body("application/sdp"))
{
t_on_reply("PBX_REPLY_TO_TEAMS");
rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA
replace-origin replace-session-connection ICE=remove");
record_route();
t_relay_to_udp("PBX_IP","5060");
}
}
#Manage RTP & transcoding comming from PBX to Teams
route[HANDLE_RTP_FROM_PBX] {
if (has_body("application/sdp"))
{
t_on_reply("TEAMS_REPLY_TO_PBX");
rtpengine_manage("SRTP codec-mask=all ICE=force codec-transcode=PCMA
replace-origin replace-session-connection");
record_route_preset("SBC_FQDN:5061;transport=tls");
add_rr_param(";r2=on");
$rd = "sip.pstnhub.microsoft.com";
$td = "SBC_FQDN";
$fd = "SBC_FQDN";
#Set TLS SNI (server name & server id)
$xavp(tls=>server_name) = "SBC_FQDN";
$xavp(tls=>server_id) = "SBC_FQDN";
t_relay();
}
}
# Wrapper for relaying requests
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Per SIP request initial checks
route[REQINIT] {
if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
# silent drop for scanners - uncomment next line if want to reply
# sl_send_reply("200", "OK");
exit;
}
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if(is_method("OPTIONS")) {
sl_send_reply("200","Keepalive");
exit;
}
if(!sanity_check("1511", "7")) {
xlog("Malformed SIP message from $si:$sp\n");
exit;
}
}
# Handle requests within SIP dialogs
route[WITHINDLG] {
if (!has_totag()) return;
#Teams reINVITEs
if(isflagset(FROM_TEAMS)) {
t_relay_to_udp("PBX_IP","5060");
exit;
}
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
#set coresponding cert on transactions
if($fd == "SBC_FQDN") {
$xavp(tls=>server_name) = "SBC_FQDN";
$xavp(tls=>server_id) = "SBC_FQDN";
}
} else if ( is_method("NOTIFY") ) {
# Add Record-Route for in-dialog NOTIFY as per RFC 6665.
record_route();
}
route(RELAY);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
route(RELAY);
exit;
} else {
# ACK without matching transaction ... ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
exit;
}
# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
}
# Manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
}
#PBX On Reply
onreply_route[PBX_REPLY_TO_TEAMS]
{
if (has_body("application/sdp"))
{
rtpengine_manage("SRTP codec-mask=all codec-transcode=PCMA
replace-origin replace-session-connection media-address=PUBLIC_IP");
}
}
#From Teams On Reply
onreply_route[TEAMS_REPLY_TO_PBX]
{
if (has_body("application/sdp"))
{
rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA
replace-origin replace-session-connection media-address=PUBLIC_IP");
}
}
# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
if (t_is_canceled()) exit;
}
event_route[tm:local-request] {
if(is_method("OPTIONS") && $ru =~
"pstnhub.microsoft.com") {
append_hf("Contact:
<sip:SBC_FQDN:5061;transport=tls>\r\n");
}
xlog("L_INFO", "Sent out tm request: $mb\n");
}
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4:07 p.m.
Likely there is some misrouting configuration with contact addresses,
like Sergiu suggested, but as an extra hint, when kamailio doesn't
report any log message in syslog about processing sip messages or
nothing printed by sipdump, run ngrep/sngrep on port 5060 to see if
something is received over udp. If something shows up there, check your
firewall or kernel/OS policies. s/ngrep get the traffic at the network
layer and it can be discarded before getting to application layer. In
some rare conditions I faced kernel module conntrack discarding
(randomly) some packets, including the ACK after 200ok.
Cheers,
Daniel
On 20.07.20 15:44, Sergiu Pojoga wrote:
Could be anything, but most likely you aren't
receiving those in Kam
because the PBX sends them somewhere else, check where the PBX sends
those, check the Contact URI, that's the address where ACKs will be
sent out.
The SBC for Teams article suggests a double header
record_route_preset() for transport conversion, not sure why your
config is different.
Also, at the very least, the config seems to invoke
/record_route[_preset]()/ twice per INVITE, it shouldn't be like that.
Good luck.
On Sun, Jul 19, 2020 at 3:55 AM Daniel van der Tang
<tangd122(a)gmail.com <mailto:tangd122@gmail.com>> wrote:
Hello,
I'm setting up a Kamailio MSFT Teams SBC that is connected to our
SIP trunk provider. This server is directly connected to the
Internet and not behind a NAT routing.
It is successfully processing invites from both sides (MSFT Teams
and SIP trunk). Unfortunately the ACKs and CANCEL messages are not
relaying to Teams. In the sipdump i can't find these messages.
Does anyone see what I am doing wrong? Below you can find my config
#!KAMAILIO
####### Defined Values #########
#!define MULTIDOMAIN 0
# - flags
# FLT_ - per transaction (message) flags
# FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
#!define FROM_TEAMS 11
#!define FROM_PBX 12
######## Define Modules ###########
#!define WITH_RTPENGINE
#!define WITH_TLS
#!define WITH_SIPDUMP
#!define WITH_DISPATCH
####### Global Parameters #########
### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR
debug=2
log_stderror=no
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "
/* number of SIP routing processes */
children=2
/* uncomment the next line to disable TCP (default on) */
# disable_tcp=yes
alias=SBC_FQDN
/* listen addresses */
listen=udp:PUBLIC_IP:5060 advertise SBC_FQDN:5060
#!ifdef WITH_TLS
listen=tls:PUBLIC_IP:5061 advertise SBC_FQDN:5061
#!endif
server_header= "Server: ABC SBC"
user_agent_header= "User-Agent: ABC SBC"
###### TLS Enable ######
#!ifdef WITH_TLS
enable_tls=yes
#tcp_connect_timeout=1000
tcp_accept_no_cl=yes
tcp_async = yes
tcp_connection_lifetime=600
/* upper limit for TLS connections */
tls_max_connections=2048
#!endif
####### Custom Parameters #########
/* These parameters can be modified runtime via RPC interface
* - see the documentation of 'cfg_rpc' module.
*
* Format: group.id <http://group.id/> = value 'desc' description
* Access: $sel(cfg_get.group.id <http://cfg_get.group.id/>) or
@cfg_get.group.id <http://cfg_get.group.id/> */
####### Modules Section ########
/* set paths to location of modules */
loadmodule "jsonrpcs.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.so"
#!ifdef WITH_RTPENGINE
loadmodule "rtpengine.so"
#!endif
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
#!ifdef WITH_SIPDUMP
loadmodule "sipdump.so"
#!endif
#!ifdef WITH_DISPATCH
loadmodule "dispatcher.so"
#!endif
# ----------------- setting module-specific parameters ---------------
#!ifdef WITH_RTPENGINE
# ----- rtpengine params -----+
modparam("rtpengine", "rtpengine_sock",
"udp:127.0.0.1:12222
<http://127.0.0.1:12222/>")
#!endif
#!ifdef WITH_TLS
# ----- tls params -----+
modparam("tls", "xavp_cfg", "tls")
modparam("tls", "config", "/etc/kamailio/tls.cfg")
modparam("tls", "connection_timeout", 10)
modparam("tls", "ssl_release_buffers", 1)
modparam("tls", "send_close_notify", 1)
modparam("tls", "session_cache", 0)
#!endif
#!ifdef WITH_SIPDUMP
modparam("sipdump", "enable", 1)
#!endif
#!ifdef WITH_DISPATCH
#---------- dispatch
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_ping_interval", 300)
#!endif
# ----- jsonrpcs params -----
modparam("jsonrpcs", "pretty_format", 1)
/* set the path to RPC fifo control file */
# modparam("jsonrpcs", "fifo_name",
"/run/kamailio/kamailio_rpc.fifo")
/* set the path to RPC unix socket control file */
# modparam("jsonrpcs", "dgram_socket",
"/run/kamailio/kamailio_rpc.sock")
# ----- ctl params -----
/* set the path to RPC unix socket control file */
# modparam("ctl", "binrpc",
"unix:/run/kamailio/kamailio_ctl")
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
# ----- rr params -----
# set next param to 1 to add value to ;lr param (helps with some UAs)
modparam("rr", "enable_full_lr", 0)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)
# ----- 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 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", "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=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
####### Routing Logic ########
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
# per request initial checks
route(REQINIT);
#check who is the sender
route(INITCHECK);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle retransmissions
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# record routing for dialog forming requests (in case they are
routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")) {
record_route();
}
# account only INVITEs
if (is_method("INVITE")) {
setflag(FLT_ACC); # do accounting
}
if ($rU==$null) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# update $du to set the destination address for proxying
#$du = "sip:" + $rd + ":9";
route(RELAY);
exit;
}
route[INITCHECK] {
if(from_uri =~ ".*microsoft.com <http://microsoft.com/>")
{
setflag(FROM_TEAMS);
$du = "sip:" + "PBX_IP";
route(HANDLE_RTP_FROM_TEAMS);
} else if(from_uri =~ ".*" + "PBX_IP")
{
setflag(FROM_PBX);
$du="sip:sip.pstnhub.microsoft.com
<http://sip.pstnhub.microsoft.com/>;transport=tls";
route(HANDLE_RTP_FROM_PBX);
} else {
exit;
}
}
#Manage RTP & transcoding comming from Teams to PBX
route[HANDLE_RTP_FROM_TEAMS] {
if (has_body("application/sdp"))
{
t_on_reply("PBX_REPLY_TO_TEAMS");
rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA
replace-origin replace-session-connection ICE=remove");
record_route();
t_relay_to_udp("PBX_IP","5060");
}
}
#Manage RTP & transcoding comming from PBX to Teams
route[HANDLE_RTP_FROM_PBX] {
if (has_body("application/sdp"))
{
t_on_reply("TEAMS_REPLY_TO_PBX");
rtpengine_manage("SRTP codec-mask=all ICE=force
codec-transcode=PCMA replace-origin replace-session-connection");
record_route_preset("SBC_FQDN:5061;transport=tls");
add_rr_param(";r2=on");
$rd = "sip.pstnhub.microsoft.com
<http://sip.pstnhub.microsoft.com/>";
$td = "SBC_FQDN";
$fd = "SBC_FQDN";
#Set TLS SNI (server name & server id)
$xavp(tls=>server_name) = "SBC_FQDN";
$xavp(tls=>server_id) = "SBC_FQDN";
t_relay();
}
}
# Wrapper for relaying requests
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route"))
t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Per SIP request initial checks
route[REQINIT] {
if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
# silent drop for scanners - uncomment next line if want to reply
# sl_send_reply("200", "OK");
exit;
}
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if(is_method("OPTIONS")) {
sl_send_reply("200","Keepalive");
exit;
}
if(!sanity_check("1511", "7")) {
xlog("Malformed SIP message from $si:$sp\n");
exit;
}
}
# Handle requests within SIP dialogs
route[WITHINDLG] {
if (!has_totag()) return;
#Teams reINVITEs
if(isflagset(FROM_TEAMS)) {
t_relay_to_udp("PBX_IP","5060");
exit;
}
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
#set coresponding cert on transactions
if($fd == "SBC_FQDN") {
$xavp(tls=>server_name) = "SBC_FQDN";
$xavp(tls=>server_id) = "SBC_FQDN";
}
} else if ( is_method("NOTIFY") ) {
# Add Record-Route for in-dialog NOTIFY as per RFC 6665.
record_route();
}
route(RELAY);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
route(RELAY);
exit;
} else {
# ACK without matching transaction ... ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
exit;
}
# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
}
# Manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
}
#PBX On Reply
onreply_route[PBX_REPLY_TO_TEAMS]
{
if (has_body("application/sdp"))
{
rtpengine_manage("SRTP codec-mask=all codec-transcode=PCMA
replace-origin replace-session-connection media-address=PUBLIC_IP");
}
}
#From Teams On Reply
onreply_route[TEAMS_REPLY_TO_PBX]
{
if (has_body("application/sdp"))
{
rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA
replace-origin replace-session-connection media-address=PUBLIC_IP");
}
}
# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
if (t_is_canceled()) exit;
}
event_route[tm:local-request] {
if(is_method("OPTIONS") && $ru =~
"pstnhub.microsoft.com
<http://pstnhub.microsoft.com/>") {
append_hf("Contact:
<sip:SBC_FQDN:5061;transport=tls>\r\n");
}
xlog("L_INFO", "Sent out tm request: $mb\n");
}
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--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla
1 Aug
1 Aug
7:39 p.m.
Thank you for your replies! Outbound calling is now completely working.
For incoming calls from SIP trunk to Teams there's still no ACK and CANCEL
messages. Which causes disconnect in 20 seconds and no audio.
I think there's something wrong with the routing or via headers in the 200
OK but don't know what.
The complete sip dump: https://paste.projectdev.org/imiluwecaf
Unfortunately i can't see the sip messages from the SIP provider side.
--
Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
2 Aug
2 Aug
2:13 p.m.
Hi,
use siptrace to see what Kamailio sees on the tls side or use the message
buffer pv $mb
Like this (please check Syntax at your own, copy pasted that with my
smartphone so no guarantees)
It shows the raw message (unencrypted) before tls module it handles or
after its decrypted AFAIK.
event_route[network:msg] {
if (is_incoming()) {
xlog("L_INFO", "Received message '$mb' \n");
} else {
xlog("L_INFO", "Sending message '$mb' \n");
}
}
Kind regards
Karsten Horsmann
tangd122 <tangd122(a)gmail.com> schrieb am Sa., 1. Aug. 2020, 19:40:
Thank you for your replies! Outbound calling is now
completely working.
For incoming calls from SIP trunk to Teams there's still no ACK and CANCEL
messages. Which causes disconnect in 20 seconds and no audio.
I think there's something wrong with the routing or via headers in the 200
OK but don't know what.
The complete sip dump: https://paste.projectdev.org/imiluwecaf
Unfortunately i can't see the sip messages from the SIP provider side.
--
Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
_______________________________________________
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https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
7:52 p.m.
On latest versions I’m trying the sipdump module, the one benefit I’m
liking over its alternatives is that you can enable/disable dumping to file
via RPC..
So in normal operation you are not dumping all tls traffic, and then you
can enable it only when you need to troubleshoot something..
Just throwing this out there in case it’s useful :)
Cheers,
Joel.
On Sun, Aug 2, 2020 at 05:15 Karsten Horsmann <khorsmann(a)gmail.com> wrote:
Hi,
use siptrace to see what Kamailio sees on the tls side or use the message
buffer pv $mb
Like this (please check Syntax at your own, copy pasted that with my
smartphone so no guarantees)
It shows the raw message (unencrypted) before tls module it handles or
after its decrypted AFAIK.
event_route[network:msg] {
if (is_incoming()) {
xlog("L_INFO", "Received message '$mb' \n");
} else {
xlog("L_INFO", "Sending message '$mb' \n");
}
}
Kind regards
Karsten Horsmann
tangd122 <tangd122(a)gmail.com> schrieb am Sa., 1. Aug. 2020, 19:40:
Thank you for your replies! Outbound calling is
now completely working.
For incoming calls from SIP trunk to Teams there's still no ACK and CANCEL
messages. Which causes disconnect in 20 seconds and no audio.
I think there's something wrong with the routing or via headers in the 200
OK but don't know what.
The complete sip dump: https://paste.projectdev.org/imiluwecaf
Unfortunately i can't see the sip messages from the SIP provider side.
--
Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
_______________________________________________
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https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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1369
days inactive
1384
days old
5 comments
6 participants
participants (6)
-
Daniel van der Tang -
Daniel-Constantin Mierla -
Joel Serrano -
Karsten Horsmann -
Sergiu Pojoga -
tangd122