[OpenSIPS-Users] Multiple branches for a single AOR
Gerwin van de Steeg
gerwin.van.de.steeg at vadacom.com
Tue Jul 17 07:35:38 EDT 2018
Folks,
I'm trying to narrow down a 482 Merged Request problem on calls from one
SIP device to another via OpenSIPS 2.4.1. Yealink T41P SIP device
(A-party), calls via OpenSIPS, to another AOR owned by a Zoiper5 device
(B-party).
The intent is to ensure that when the B-party rejects the call with a 486
Busy Here, that the response code gets sent through to A-party. However
what I'm seeing is the 486 gets sent to OpenSIPS which ACK's it, but
doesn't go anywhere from there, and then something causes a second invite
to be sent from OpenSIPS to the B-party which then responds of course with
482 Merged Request.
The call as it is progressing through the call flow seems to be starting a
second branch to the AOR (only one SIP device registered using UDP per AOR).
What would be causing that second call so that I can eliminate it and get
to the behaviour I'm expecting. Just using a slightly modified residential
default config template with websocket support (the problem was noticed
using SIP.JS but exists also in generic SIP device to SIP device calls).
Image containing sngrep of call: https://imgur.com/RCZXkO6
Subscribers are in the form of <username>@<domain>
With an alias setup for an extension number.
ie.
alfred.anderson at ... = 552
alice.bell at ... = 553
excerpt from opensips.cfg
if ($rU==NULL) {
# request with no Username in RURI
send_reply("484","Address Incomplete");
exit;
}
$acc_extra(src_ip) = $si; # source IP of the request
$acc_leg(caller) = $fu;
$acc_leg(callee) = $ru;
# apply DB based aliases
if (alias_db_lookup("dbaliases")) {
xlog("Alias lookup success [$fu/$tu/$ru/$ci]");
}
else {
xlog("Alias lookup failure [$fu/$tu/$ru/$ci]");
}
# do blind callforward lookup
if (avp_db_load("rU", "$avp(callfwd)")) {
t_reply("181", "Call Is Being Forwarded");
$ru = $avp(callfwd);
xlog("forwarded call to: $avp(callfwd)");
route(relay);
exit;
}
# apply transformations from dialplan table
dp_translate("0", "$rU/$rU");
# check if the call needs to be routed to freeswitch
route(freeswitch);
# here we would set the redirect URI if it had one
route(lookup);
}
route[lookup] {
script_trace(1, "$rm from $si, rur=$ru", "me");
xlog("route:lookup");
# do lookup with method filtering
if (!lookup("location","m")) {
xlog("lookup failure");
t_newtran();
if (!db_does_uri_exist()) {
xlog("$cfg_line: URI doesn't exist");
send_reply("420", "Bad Extension");
exit;
}
t_reply("404", "Not Found");
exit;
}
# when routing via usrloc, log the missed calls also
do_accounting("db","missed");
route(relay);
}
route[freeswitch] {
xlog("route:freeswitch");
if (!is_method("INVITE")) {
return;
}
# if the called number begins with the right dialplan redirect it to
freeswitch
# here we take everythign prefixed with a *, strip it, and send it
to freeswitch
if ($rU=~"^\*") {
strip(1);
$du = "sip:10.23.4.192:50600";
route(relay);
}
}
route[relay] {
xlog("route:relay: Relaying: method=$rm");
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("per_branch_ops");
t_on_reply("handle_nat");
t_on_failure("missed_call");
}
else if (is_method("BYE|CANCEL")) {
# cancel the rtpengine transcoding
rtpengine_delete();
}
if (!t_relay()) {
send_reply("500","Internal Error");
}
exit;
}
branch_route[per_branch_ops] {
script_trace(1, "$rm from $si, rur=$ru", "me");
xlog("[$ci/$T_branch_idx] branch_route:per_branch_ops: new branch at
$ru\n");
# WebSocket specific handling with NORMAL SDP negotiation
# assumes SDP offer in the INVITE from the UAC, and SDP
# answer is in 200 OK from the UAS
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))
$var(rtpengine_flags) = "RTP/AVP replace-session-connection
replace-origin ICE=remove";
# only enable transcoding if websocket call for now
if (isflagset(SRC_WS) || isbflagset(DST_WS)) {
rtpengine_offer("$var(rtpengine_flags)");
}
}
onreply_route[handle_nat] {
script_trace(1, "$rm from $si, rur=$ru", "me");
xlog("[$ci/$T_branch_idx] onreply_route:handle_nat: $ru\n");
# WebSocket specific handling with NORMAL SDP negotiation
# assumes SDP offer in the INVITE from the UAC, and SDP
# answer is in 200 OK from the UAS
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))
$var(rtpengine_flags) = "RTP/AVP replace-session-connection
replace-origin ICE=remove";
# only enable transcoding if websocket call for now
if (isflagset(SRC_WS) || isbflagset(DST_WS)) {
rtpengine_answer("$var(rtpengine_flags)");
}
}
failure_route[missed_call] {
script_trace(1, "$rm from $si, rur=$ru", "me");
xlog("[$ci/$T_branch_idx] failure_route:missed_call: incoming
failure response to $rm <- $T_reply_code/$T_ruri");
if (t_was_cancelled()) {
xlog("[$ci/$T_branch_idx] was cancelled");
exit;
}
do_accounting("db", "missed");
if (!t_relay()) {
send_reply("500","Internal Error");
}
else {
xlog("[$ci/$T_branch_idx] Relay success $rm/$T_reply_code");
}
}
Cheers,
Gerwin
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