[OpenSIPS-Users] Multiple branches for a single AOR

Gerwin van de Steeg gerwin.van.de.steeg at vadacom.com
Tue Jul 17 09:27:05 EDT 2018


Perfect, thanks that fixed it.

On 18 July 2018 at 01:02, Liviu Chircu <liviu at opensips.org> wrote:

> 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 Developerhttp://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
>
>
>
> Ben Newlin
>
>
>
> *From: *Users <users-bounces at lists.opensips.org>
> <users-bounces at lists.opensips.org> on behalf of Liviu Chircu
> <liviu at opensips.org> <liviu at opensips.org>
> *Reply-To: *OpenSIPS users mailling list <users at lists.opensips.org>
> <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>
> <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
>
>
>
> 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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