[OpenSIPS-Users] Multiple branches for a single AOR
Liviu Chircu
liviu at opensips.org
Tue Jul 17 09:02:50 EDT 2018
Hi Ben,
Just for the sake of completeness, if we do decide to retry a request
within a failure route, we will not do a "retransmission", but actually
create a brand new transaction (with a unique ";branch=" param)on the
SIP proxy's UAC side (the outgoing side).
Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 17.07.2018 15:54, Ben Newlin wrote:
>
> Gerwin,
>
> Specifically, it is the t_relay() call within your failure_route.
> t_relay() is used to send requests, not responses. The automatic
> action of failure_route if no retransmission is attempted is to send
> the reply back upstream [1]. So you don’t have to do anything if that
> is the functionality you desire.
>
> [1] http://www.opensips.org/Documentation/Script-Routes-2-4#toc3
> <http://www.opensips.org/Documentation/Script-Routes-2-4#toc3>
>
> Ben Newlin
>
> *From: *Users <users-bounces at lists.opensips.org> on behalf of Liviu
> Chircu <liviu at opensips.org>
> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date: *Tuesday, July 17, 2018 at 8:50 AM
> *To: *"users at lists.opensips.org" <users at lists.opensips.org>
> *Subject: *Re: [OpenSIPS-Users] Multiple branches for a single AOR
>
> Hi Gerwin,
>
> Inside your failure route, you are always attempting a retry of any
> failed request. This logic conflicts with your initial statement that
> "The intent is to ensure that the response code gets sent through to
> A-party".
>
> Best regards,
>
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
>
> On 17.07.2018 14:35, Gerwin van de Steeg wrote:
>
> 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
> <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);
> $du = "sip:10.23.4.192:50600
> <http://10.23.4.192:50600>";
> route(relay);
> }
> }
>
>
> route[relay] {
> xlog("route:relay: Relaying: method=$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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