[OpenSIPS-Users] Opensips with RTPEngine
HS
bullehs at gmail.com
Wed Nov 18 07:33:56 EST 2020
Hi all.
I just tried to replace RTPProxy with RTPEngine. I did the change as I was
getting one-way audio previously. However, one connection seems
inconsistent (One-way audio) sometimes and the other connection does not
actually even ring a bell. I have tried with both devices on the same Wi-Fi
and one on Wi-Fi and the other on a cellular network. There's no error in
opensips or rtpengine logs, so it seems that the routing is incorrect. I am
using Opensips 3.0. Please let me know what is incorrect or if further
information is needed. Following is my opensips.cfg file:
Thanks for the help in advance.
#
# OpenSIPS residential configuration script
# by OpenSIPS Solutions <team at opensips-solutions.com>
#
# This script was generated via "make menuconfig", from
# the "Residential" scenario.
# You can enable / disable more features / functionalities by
# re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
# https://opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#
####### Global Parameters #########
log_level=3
log_stderror=no
#log_facility=LOG_LOCAL0
log_facility=LOG_LOCAL7
udp_workers=8
/* uncomment the following lines to enable debugging */
#debug_mode=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
/* comment the next line to enable the auto discovery of local aliases
based on reverse DNS on IPs */
auto_aliases=no
advertised_address=<Public_IP>
listen=<udp:Pvt-IP:5060> # CUSTOMIZE ME
listen=<tcp:Pvt-IP:5060> # CUSTOMIZE ME
# Set up listeners
#listen=ws:<Pvt-IP:8080>
#listen=wss:<Pvt-IP:443>
#listen=tls:<Pvt-IP:5061>
####### Modules Section ########
#set module path
mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"
#### SIGNALING module
loadmodule "signaling.so"
#### StateLess module
loadmodule "sl.so"
#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)
#### MAX ForWarD module
loadmodule "maxfwd.so"
#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"
#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)
#### MYSQL module
loadmodule "db_mysql.so"
#### HTTPD module
loadmodule "httpd.so"
modparam("httpd", "port", 8888)
#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
modparam("registrar", "received_avp", "$avp(received_nh)")/* uncomment the
next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", 0)
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")
#### ALIAS module
loadmodule "alias_db.so"
modparam("alias_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### DOMAIN module
loadmodule "domain.so"
modparam("domain", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("domain", "db_mode", 0) # Use caching
modparam("auth_db|usrloc", "use_domain", 1)
#### PRESENCE modules
loadmodule "xcap.so"
loadmodule "presence.so"
loadmodule "presence_xml.so"
modparam("xcap|presence", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("presence_xml", "force_active", 1)
modparam("presence", "fallback2db", 0)
#### DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600) # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### NAT modules
loadmodule "nathelper.so"
modparam("registrar|nathelper", "received_avp", "$avp(rcv)")
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG")
modparam("nathelper", "sipping_from", "sip:pinger at 127.0.0.1") #CUSTOMIZE ME
modparam("nathelper", "received_avp", "$avp(received_nh)")
##loadmodule "rtpproxy.so"
#modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221") # CUSTOMIZE ME
loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:<Pvt-IP:60000">) # CUSTOMIZE ME
#modparam("rtpengine", "rtpengine_sock", "udp:localhost:60000") # CUSTOMIZE
ME
#### DIALPLAN module
loadmodule "dialplan.so"
modparam("dialplan", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### DYNAMMIC ROUTING module
loadmodule "drouting.so"
modparam("drouting", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
#### MI_HTTP module
loadmodule "mi_http.so"
loadmodule "proto_udp.so"
loadmodule "proto_tcp.so"
#loadmodule "proto_tls.so"
#### WebSocket and WebSocketSecure protocol
#loadmodule "proto_wss.so"
#loadmodule "proto_ws.so"
# Certificate management
# loadmodule "tls_mgm.so" - W
# modparam("tls_mgm", "tls_method", "[default]SSLv23")
# modparam("tls_mgm","verify_cert", "1")
# modparam("tls_mgm","require_cert", "0")
# modparam("tls_mgm","tls_method", "[Mydomain:5061]TLSv1")
# modparam("tls_mgm", "server_domain", "Mydomain:5061")
# modparam("tls_mgm", "certificate",
"[Mydomain]/etc/opensips/tls/rootCA/cacert.pem")
# modparam("tls_mgm", "private_key",
"[Mydomain]/etc/opensips/tls/rootCA/private/cakey.pem")
####### Routing Logic ########
# main request routing logic
route{
# initial NAT handling; detect if the request comes from behind a NAT
# and apply contact fixing
force_rport();
if (nat_uac_test(23)) {
if (is_method("REGISTER")) {
fix_nated_register();
setbflag(NAT);
} else {
fix_nated_contact();
setflag(NAT);
}
}
if (!mf_process_maxfwd_header(10)) {
send_reply(483,"Too Many Hops");
exit;
}
if (has_totag()) {
# handle hop-by-hop ACK (no routing required)
if ( is_method("ACK") && t_check_trans() ) {
t_relay();
exit;
}
# sequential request within a dialog should
# take the path determined by record-routing
if ( !loose_route() ) {
if (is_method("SUBSCRIBE") && is_myself("$rd")) {
# in-dialog subscribe requests
route(handle_presence);
exit;
}
# we do record-routing for all our traffic, so we should not
# receive any sequential requests without Route hdr.
send_reply(404,"Not here");
exit;
}
# validate the sequential request against dialog
if ( $DLG_status!=NULL && !validate_dialog() ) {
xlog("In-Dialog $rm from $si (callid=$ci) is not valid according to
dialog\n");
## exit;
}
if (is_method("BYE")) {
# do accounting even if the transaction fails
do_accounting("db","failed");
}
if (check_route_param("nat=yes"))
setflag(NAT);
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(relay);
exit;
}
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans())
t_relay();
exit;
}
# absorb retransmissions, but do not create transaction
t_check_trans();
if ( !(is_method("REGISTER") || is_from_gw() ) ) {
if (is_from_local()) {
# authenticate if from local subscriber
# authenticate all initial non-REGISTER request that pretend to be
# generated by local subscriber (domain from FROM URI is local)
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", 0);
exit;
}
if ($au!=$fU) {
send_reply(403,"Forbidden auth ID");
exit;
}
consume_credentials();
# caller authenticated
} else {
# if caller is not local, then called number must be local
if (!is_uri_host_local()) {
send_reply(403,"Relay Forbidden");
exit;
}
}
}
# preloaded route checking
if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
send_reply(403,"Preload Route denied");
exit;
}
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
# create dialog with timeout
if ( !create_dialog("B") ) {
send_reply(500,"Internal Server Error");
exit;
}
do_accounting("db");
}
if (!is_uri_host_local()) {
append_hf("P-hint: outbound\r\n");
route(relay);
}
# requests for my domain
if( is_method("PUBLISH|SUBSCRIBE"))
route(handle_presence);
# check if the clients are using WebSockets or WebSocketSecure
# if ($proto == "ws" || $proto == "wss")
# setflag(SRC_WS);
# else
# setflag(SRC_SIP);
# }
if (is_method("REGISTER")) {
# authenticate the REGISTER requests
# if (!www_authorize("", "subscriber")) {
# www_challenge("", 0);
# exit;
# }
$var(auth_code) = www_authorize("", "subscriber");
if ( $var(auth_code) == -1 || $var(auth_code) == -2 ) {
xlog("L_NOTICE","Auth error for $fU@$fd from $si cause $var(auth_code)");
}
if ( $var(auth_code) < 0 ) {
www_challenge("", 0);
exit;
}
if ($au!=$tU) {
send_reply(403,"Forbidden auth ID");
exit;
}
if ($proto == "tcp")
setflag(TCP_PERSISTENT);
if (isflagset(NAT)) {
setbflag(SIP_PING_FLAG);
# check if the clients are using WebSockets or WebSocketSecure
if ($proto == "ws" || $proto == "wss")
setflag(SRC_WS);
else
setflag(SRC_SIP);
}
if (isflagset(SRC_WS))
setbflag(DST_WS);
if (!save("location"))
sl_reply_error();
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
send_reply(484,"Address Incomplete");
exit;
}
# apply DB based aliases
alias_db_lookup("dbaliases");
# apply transformations from dialplan table
dp_translate( 0, "$rU", $rU);
if ($rU=~"^\+[1-9][0-9]+$") {
if (!do_routing(0)) {
send_reply(500,"No PSTN Route found");
exit;
}
route(relay);
exit;
}
# do lookup with method filtering
if (!lookup("location","m")) {
if (!db_does_uri_exist("$ru","subscriber")) {
send_reply(420,"Bad Extension");
exit;
}
# redirect to a different VM system
$du = "sip:127.0.0.2:5060"; # CUSTOMIZE ME
route(relay);
}
if (isbflagset(NAT)) setflag(NAT);
# when routing via usrloc, log the missed calls also
do_accounting("db","missed");
route(relay);
}
route[relay] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
# if (isflagset(NAT)) {
# if has_body("application/sdp")) {
# rtpengine_offer("trust-address");
# }
t_on_branch("handle_nat");
t_on_reply("handle_nat");
t_on_failure("missed_call");
}
else if (is_method("BYE|CANCEL")) {
rtpengine_delete();
}
# 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();
# }
if (isflagset(NAT)) {
add_rr_param(";nat=yes");
}
if (!t_relay()) {
send_reply(500,"Internal Error");
}
exit;
}
# Presence route
route[handle_presence]
{
if (!t_newtran()) {
sl_reply_error();
exit;
}
if(is_method("PUBLISH")) {
handle_publish();
} else
if( is_method("SUBSCRIBE")) {
handle_subscribe();
}
exit;
}
branch_route[per_branch_ops] {
xlog("new branch at $ru\n");
}
branch_route[handle_nat] {
if (!is_method("INVITE") || !has_body("application/sdp"))
return;
if (isflagset(SRC_WS) && isbflagset(DST_WS))
$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
else if (isflagset(SRC_WS) && !isbflagset(DST_WS))
$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
else if (!isflagset(SRC_WS) && isbflagset(DST_WS))
$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
else if (!isflagset(SRC_WS) && !isbflagset(DST_WS))
$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
rtpengine_offer("$var(rtpengine_flags)");
}
#onreply_route[handle_nat] {
# if (nat_uac_test(1))
# fix_nated_contact();
# if ( isflagset(NAT) )
# rtpproxy_answer("ro");
#onreply_route[handle_nat] {
# if (nat_uac_test(1))
# fix_nated_contact();
# if ( isflagset(NAT) )
# rtpengine_answer("trust-address");
# if (has_body("application/sdp"))
# rtpengine_answer();
# xlog("incoming reply\n");
#}
onreply_route[handle_nat] {
fix_nated_contact();
if (!has_body("application/sdp"))
return;
if (isflagset(SRC_WS) && isbflagset(DST_WS))
$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
else if (isflagset(SRC_WS) && !isbflagset(DST_WS))
$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
else if (!isflagset(SRC_WS) && isbflagset(DST_WS))
$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
else if (!isflagset(SRC_WS) && !isbflagset(DST_WS))
$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
rtpengine_answer("$var(rtpengine_flags)");
}
failure_route[missed_call] {
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;
##}
# redirect the failed to a different VM system
if (t_check_status("486|408")) {
$du = "sip:127.0.0.2:5060"; # CUSTOMIZE ME
# do not set the missed call flag again
route(relay);
}
}
local_route {
if (is_method("BYE") && $DLG_dir=="UPSTREAM") {
acc_db_request("200 Dialog Timeout", "acc");
}
}
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