Kamailio：調(diào)用set_contact_alias()之后，在原有的contact的后面增加參數(shù)，具體地說，就是網(wǎng)絡(luò)地址，網(wǎng)絡(luò)端口和transport，好處是收到后續(xù)請求之時可以恢復原有contact的內(nèi)容（當然也有壞處，就是增加參數(shù)之后導致包大，可能超過MTU）

OpenSIPS：調(diào)用fix_nated_contact()之后，用網(wǎng)絡(luò)地址和網(wǎng)絡(luò)端口直接覆蓋了在原有的contact，那么當收到后續(xù)請求時不太可能恢復原有contact的內(nèi)容。竊以為，一定要慎重

比如，下圖所示，OpenSIPS就不要修改uac的contact，顯然如此

UAC--->第三方Proxy--->OpenSIPS--->FreeSWITCH

參考鏈接：

https://kb.smartvox.co.uk/opensips/nat-contact-and-via-fixing-in-sip-part-3/

當然，我們可以模仿set_contact_alias函數(shù)

route[set_contact_alias] {if (is_present_hf("contact")) {$var(sut) = $si + "~" +  $sp + "~" + $socket_in(proto); # 遠端網(wǎng)絡(luò)地址$var(ct) = $(ct{re.subst,/^(.*)<(.*)>/\2/}); # 取contact <> 里面的內(nèi)容# 更換成新的contact$var(new_ct) = "<" + $var(ct) + ";my_alias=" + $var(sut) + ">";remove_hf("contact");insert_hf("Contact: $var(new_ct)\r\n", "Call-ID");}
}route[restore_contact_by_alias] {if (!has_totag()) { # 對話內(nèi)請求才能調(diào)用這個路由return;}# xlog("***ru = $ru, du = $du\n");if ($du != NULL) { # 如果已經(jīng)設(shè)置了$du，那么不能再調(diào)用這個路由return;}$var(my_alias) = $(ru{nameaddr.param,my_alias});# xlog("***my_alias = $var(my_alias)\n");if ($var(my_alias) != NULL) {$var(ip) = $(var(my_alias){s.select,0,~});$var(port) = $(var(my_alias){s.select,1,~});$var(transport) = $(var(my_alias){s.select,2,~});$var(url) = "sip:" + $var(ip) + ":" + $var(port) + ";transport=" + $var(transport);# xlog("url = $var(url)\n");setdsturi($var(url));$ru  = $(ru{re.subst,/^(sip:.*);my_alias=.*/\1/}); # 刪除ru里面的my_alias以及內(nèi)容		}
}

完整的路由腳本如下：

####### Global Parameters #########/* uncomment the following lines to enable debugging */
#debug_mode=yeslog_level=3
xlog_level=3
stderror_enabled=no
syslog_enabled=yes
syslog_facility=LOG_LOCAL0udp_workers=4/* uncomment the next line to enable the auto temporary blacklisting ofnot available destinations (default disabled) */
#disable_dns_blacklist=no/* uncomment the next line to enable IPv6 lookup after IPv4 dnslookup failures (default disabled) */
#dns_try_ipv6=yessocket=udp:172.20.10.6:5060   # CUSTOMIZE ME
socket=tcp:172.20.10.6:5060   # CUSTOMIZE ME####### 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", "working_mode_preset", "single-instance-sql-write-back")
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", "$avp(received_nh)")/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.if you enable this parameter, be sure to enable "append_fromtag"in "rr" module */
modparam("acc", "detect_direction", 0)####  NAT modules
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG")
modparam("nathelper", "sipping_from", "sip:pinger@127.0.0.1") #CUSTOMIZE ME
modparam("nathelper", "received_avp", "$avp(received_nh)")# loadmodule "rtpproxy.so"
# modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:12221") # CUSTOMIZE ME####  MI_HTTP module
loadmodule "mi_http.so"loadmodule "proto_udp.so"
loadmodule "proto_tcp.so"
####### Routing Logic ######### main request routing logicroute{xlog("$rm|$fU|$tU|$ci from $si:$sp\n");# initial NAT handling; detect if the request comes from behind a NAT# and apply contact fixingforce_rport();#if (nat_uac_test(23)) {if (true) {if (is_method("REGISTER")) {fix_nated_register();setbflag("NAT");} else {# fix_nated_contact();route(set_contact_alias);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-routingif ( !loose_route() ) {# 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;}route(restore_contact_by_alias);if (is_method("BYE")) {# do accounting even if the transaction failsdo_accounting("log","failed");}if (check_route_param("nat=yes"))setflag("NAT");# route it out to whatever destination was set by loose_route()# in $du (destination URI).route(relay);exit;}# CANCEL processingif (is_method("CANCEL")) {if (t_check_trans())t_relay();exit;}# absorb retransmissions, but do not create transactiont_check_trans();if ( !(is_method("REGISTER")  ) ) {if (is_myself("$fd")) {} else {# if caller is not local, then called number must be localif (!is_myself("$rd")) {send_reply(403,"Relay Forbidden");exit;}}}# preloaded route checkingif (loose_route()) {xlog("L_ERR","Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");if (!is_method("ACK"))send_reply(403,"Preload Route denied");exit;}# record routingif (!is_method("REGISTER|MESSAGE"))record_route();# account only INVITEsif (is_method("INVITE")) {do_accounting("log");}if (!is_myself("$rd")) {append_hf("P-hint: outbound\r\n");route(relay);}# requests for my domainif (is_method("PUBLISH|SUBSCRIBE")) {send_reply(503, "Service Unavailable");exit;}if (is_method("REGISTER")) {if ($socket_in(proto) == "tcp")setflag("TCP_PERSISTENT");if (isflagset("NAT")) {setbflag("SIP_PING_FLAG");}# store the registration and generate a SIP replyif (!save("location"))xlog("failed to register AoR $tu\n");exit;}if ($rU==NULL) {# request with no Username in RURIsend_reply(484,"Address Incomplete");exit;}# do lookup with method filteringif (!lookup("location", "method-filtering")) {t_reply(404, "Not Found");exit;}if (isbflagset("NAT")) setflag("NAT");# when routing via usrloc, log the missed calls alsodo_accounting("log","missed");route(relay);
}route[relay] {# for INVITEs enable some additional helper routesif (is_method("INVITE")) {# if (isflagset("NAT") && has_body("application/sdp")) {#	rtpproxy_offer("ro");# }t_on_branch("per_branch_ops");t_on_reply("handle_nat");t_on_failure("missed_call");}if (isflagset("NAT")) {add_rr_param(";nat=yes");}if (!t_relay()) {send_reply(500,"Internal Error");}exit;
}branch_route[per_branch_ops] {xlog("new branch at $ru\n");
}onreply_route[handle_nat] {#if (nat_uac_test(1))#	fix_nated_contact();route(set_contact_alias);# if ( isflagset("NAT") && has_body("application/sdp") )#	rtpproxy_answer("ro");xlog("incoming reply\n");
}failure_route[missed_call] {if (t_was_cancelled()) {exit;}# uncomment the following lines if you want to block client# redirect based on 3xx replies.##if (t_check_status("3[0-9][0-9]")) {##t_reply(404,"Not found");##	exit;##}
}route[set_contact_alias] {if (is_present_hf("contact")) {$var(sut) = $si + "~" +  $sp + "~" + $socket_in(proto); # 遠端網(wǎng)絡(luò)地址$var(ct) = $(ct{re.subst,/^(.*)<(.*)>/\2/}); # 取contact <> 里面的內(nèi)容# 更換成新的contact$var(new_ct) = "<" + $var(ct) + ";my_alias=" + $var(sut) + ">";remove_hf("contact");insert_hf("Contact: $var(new_ct)\r\n", "Call-ID");}
}route[restore_contact_by_alias] {if (!has_totag()) { # 對話內(nèi)請求才能調(diào)用這個路由return;}# xlog("***ru = $ru, du = $du\n");if ($du != NULL) { # 如果已經(jīng)設(shè)置了$du，那么不能再調(diào)用這個路由return;}$var(my_alias) = $(ru{nameaddr.param,my_alias});# xlog("***my_alias = $var(my_alias)\n");if ($var(my_alias) != NULL) {$var(ip) = $(var(my_alias){s.select,0,~});$var(port) = $(var(my_alias){s.select,1,~});$var(transport) = $(var(my_alias){s.select,2,~});$var(url) = "sip:" + $var(ip) + ":" + $var(port) + ";transport=" + $var(transport);# xlog("url = $var(url)\n");setdsturi($var(url));$ru  = $(ru{re.subst,/^(sip:.*);my_alias=.*/\1/}); # 刪除ru里面的my_alias以及內(nèi)容		}
}

以上腳本在OpenSIPS3.4.3上調(diào)試通過，但沒測試其他版本

于2024年2月8日（次日即除夕）

全文完