[SR-Users] Setting up MSFT Teams SBC <> SIP trunk but forwarding ACK and CANCEL messages
Sergiu Pojoga
pojogas at gmail.com
Mon Jul 20 15:44:11 CEST 2020
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 at 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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> sr-users at lists.kamailio.org
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>
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