[OpenSIPS-Users] WebRTC and mid_registrar issue

[Liviu Chircu]

Hi Terry,


Currently, mid-registrar is incompatible with topology hiding, as they 
both attempt to edit the "Contact" header field.

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 09.04.2019 14:01, Terry Walters wrote:
> 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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