[OpenSIPS-Users] WebRTC and mid_registrar issue

[Terry Walters]

I have a working OpenSIPS deployment running as a PSTN proxy and am trying to add in support for WebRTC which will forward requests to an internal SIP PBX. Currently the registrations appear to be passing through the mid_registrar correctly and the initial invite from my PBX is correctly routed to the websocket client.

The PBX receives messages up to the 180 ringing correctly, but on the 200 OK from the websocket client the r.uri contains a d.id rather than the ctid, this then results in the calling timing out and the ACK from the PBX not routing to the websocket client. I believe this is the script area that is falling over once the has_totag function is checked.

if (loose_route())
  {
   if (is_method("INVITE"))
   {
    # even if in most of the cases is useless, do RR for
    # re-INVITEs alos, as some buggy clients do change route set
    # during the dialog.
    record_route();
   }
   else if (is_method("ACK"))
   {
    if (has_body("application/sdp"))
    {
     # check if destination is WS
     if ($du != NULL)
      $var(proto) = $dP;
     else
      $var(proto) = $rP;
     if ($var(proto) == "WS" || $var(proto) == "WSS")
      setbflag(DST_WS);

     route(rtpengine_answer);
    }
   }

   # route it out to whatever destination was set by loose_route()
   # in $du (destination URI).
   route(websocket);
  }
  else
  {
   if ( is_method("ACK") )
   {
    if ( t_check_trans() )
    {
     # non loose-route, but stateful ACK; must be an ACK after
     # a 487 or e.g. 404 from upstream server
     t_relay();
     exit;
    }
    else
    {
     # ACK without matching transaction ->
     # ignore and discard
     exit;
    }
   }
   sl_send_reply("404","Not here");

The route[websocket] is using force send socket to remove the NAT address that cannot be accessed internally:

route[webscoket]
{
xlog("L_INFO","Entered websocket route");
# for each branch we will call the function below
t_on_branch("per_branch_ops");

# for each reply we will call the function below
t_on_reply("handle_nat");
# initial invites from the main registrar - need to look them up!
if (is_method("INVITE"))
 if($Ri == "1.1.1.1" and $Rp == 5070)
 {
  xlog("L_INFO","Call received for a websocket client");
  if (!mid_registrar_lookup("location"))
  {
   t_reply("404", "Not Found");
   exit;
  }

 }
 else  if($Ri=="1.1.1.1" && $Rp == "8080")
 {
  xlog("L_INFO","Inbound call received on port 8080");
  route("ToInternal");
  t_on_failure("int_invites");
 }

if((is_method("REGISTER"))&&($Ri=="1.1.1.1")&&($Rp =="8080"))
{
 xlog("L_INFO","Register received - mid-registrar actions");
 fix_nated_register();
 mid_registrar_save("location");
 switch ($retcode)
 {
 case 1:
  xlog("L_INFO", "forwarding REGISTER to main registrar...\n");
  $ru = "sip:2.2.2.2:5060";
  force_send_socket(UDP:1.1.1.1:5070);
 if (!t_relay()) {
  send_reply("500", "Server Internal Error 1");
 }
  t_on_failure("int_invites");
  break;
 case 2:
  xlog("L_INFO", "REGISTER has been absorbed!\n");
  break;
 default:
  xlog("L_ERR", "mid-registrar error!\n");
  send_reply("500", "Server Internal Error 2");
 }

 exit;
}

# removing the rtpproxy session
if(is_method("CANCEL|BYE"))
{
 rtpengine_delete();
}

    # try to send the request on its way, if it fails send back a
    # stateless error to the requestor
    if (t_relay())
    {
        xlog("L_INFO", "$ci|pass|$rd:$rp");
    }
    else
    {
        xlog("L_ERR", "$ci|end|unable to relay message");

        sl_reply_error();
    }


}
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