[OpenSIPS-Users] 3.1 - Mid_Registrar - AOR throttling with WebRTC failing

[Mark Allen]

I don't know if anyone has had a chance to look at my problem but I wonder
if at least I could get an opinion on the following:

1 - Should I be seeing the path saved in the appropriate column in the
"location" table?
2 - Am I using mid_registrar_save() and mid_registrar_lookup() with path
support correctly in my script?
3 - have I correctly understood how to combine WebRTC with mid-registrar
module, path, and AOR throttling so that it should work for calls
originating from the main registrar?

I'm stuck on how to move forward with this

Cheers,

Mark

Relevant code snippets...

loadmodule "mid_registrar.so"
modparam("mid_registrar", "mode", 2) /* 0 = mirror / 1 = ct / 2 = AoR */
modparam("mid_registrar", "outgoing_expires", 3600)

add_path_received();
$avp(returncode) = mid_registrar_save("location","p0v");
switch ($avp(returncode)) {
    case 1:
        route(resolve_registrar);
        $ru = "sip:" + $avp(main_registrar) + ":5060";
        t_on_failure("1");
        t_relay();
        break;
    case 2:
        break;
    default:
}

if (!mid_registrar_lookup("location")) {
    t_reply(404, "Not Found");
    exit;
}


NB - route(resolve_registrar) sets the variable $avp(main_registrar) to the
IP address of the Asterisk server

On Thu, 30 Jul 2020 at 09:16, Mark Allen <mark at allenclan.co.uk> wrote:

> We are working on a test setup, hoping to move to a production system in
> mid-August. We want to use mid-registrar AOR throttling. Users will connect
> through OpenSIPS using a combination of SIP and WebRTC endpoints,
> registering to an extension on an Asterisk main-registrar...
>
>               +----------+
> <SIP>    ---> |          |      +----------+
> <SIP>    ---> | OpenSIPS | ---> | Asterisk |
> <WebRTC> ---> |          |      +----------+
>               +----------+
>
> Multiple SIP phones (hardware or softphones) registering via an OpenSIPS
> 3.1 mid_registration AOR is working fine. A call to the extension number on
> Asterisk results in all mid-registered SIP extensions ringing and when one
> answers, the other devices register a missed call. So far, so good.
>
> With 3.0 - we had a problem with WebRTC "phones" (even when just using
> mid_registrar in "mirroring" mode). Webphone could register and call other
> phones without a problem. However, calls to the WebPhone failed - there was
> a problem with the WebSocket addressing giving "476 Unresolvable
> destination" when the call originates from the main registrar - e.g. one
> extension calling another. The /var/log/syslog entry said...
>
>   ERROR:core:sip_resolvehost: forced proto 6 not matching sips uri
>   CRITICAL:core:mk_proxy: could not resolve hostname:
> "4xp44jxl0qq0.invalid"
>   ERROR:tm:uri2proxy: bad host name in URI <sips:11001 at
> 4xp44jxl0qq0.invalid;rtcweb-breaker=yes;transport=wss>
>   ERROR:tm:t_forward_nonack: failure to add branches
>
> Stas Kobar gave me a way to resolve this -
> http://lists.opensips.org/pipermail/users/2020-July/043443.html  As we
> were using 3.0, I used the "path" module and  "add_path_received()" to
> handle this for WebRTC. This worked for a single device registered to an
> address. However, as far as I could see, using "path" effectively bypassed
> the "contact" address held in the OpenSIPS "location" table so it didn't
> work for AOR throttling.
>
> I was hoping that, with mid_registrar on 3.1 baking in path support, I
> could just use "mid_registrar_save('location','p0v')" to store the WebRTC
> destination path in the "location" table. Then, with a call to the WebRTC
> endpoint from the main registrar, "mid_registrar_lookup('location')" would
> use the stored path from the "location" table to send traffic on to the
> WebRTC phone and it would work fine with AOR throttling. However, that's
> not happening, and looking at the "location" table, no path seems to
> be being stored.
>
> If I register a WebRTC "phone" first, the path is included on the
> registration SIP message sent from OpenSIPS to Asterisk. If I then register
> additional SIP phones on OpenSIPS, AOR throttling works, because, when the
> call originates from Asterisk it includes the "route" HF that points to the
> WebRTC destination. However, if a SIP phone registers first, Asterisk
> doesn't get the WebRTC path, so calls fail to reach the WebRTC destination
> because it tries to use the first registered SIP phone's path.
>
> So - 2 questions really...
>
> 1 - Can I use AOR throttling with WebRTC (I can't guarantee that the
> WebRTC endpoint will be the first to register or that there will only be
> one WebRTC endpoint)
>
> 2 - If the answer to 1 is yes, what am I doing wrong?
>
> Relevant code snippets...
>
> loadmodule "mid_registrar.so"
> modparam("mid_registrar", "mode", 2) /* 0 = mirror / 1 = ct / 2 = AoR */
> modparam("mid_registrar", "outgoing_expires", 3600)
>
> add_path_received();
> $avp(returncode) = mid_registrar_save("location","p0v");
> switch ($avp(returncode)) {
>     case 1:
>         route(resolve_registrar);
>         $ru = "sip:" + $avp(main_registrar) + ":5060";
>         t_on_failure("1");
>         t_relay();
>         break;
>     case 2:
>         break;
>     default:
> }
>
> if (!mid_registrar_lookup("location")) {
>     t_reply(404, "Not Found");
>     exit;
> }
>
>
> NB - route(resolve_registrar) sets the variable $avp(main_registrar) to
> the IP address of the Asterisk server
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20200803/859e94dc/attachment.html>