[sr-dev] [kamailio/kamailio] Topos not working correctly with loos_route() on a leg (Issue #2905)

Anis Bedhiafi notifications at github.com
Wed Oct 27 15:56:48 CEST 2021


### Description

I noticed an issue regarding loose_route() when using Topos module.

My architecture is the following:

```
Client (behind NAT or not) ------>   Kamailio (Access SBC) -----------> Kamailio (Interconnect SBC) -----------> Carrier

```

Kamailio Access SBC is listening on 2 interfaces:

```
Public interface : udp:A.A.A.A:5060
Core interface   : udp:B.B.B.B:5060

```

Kamailio Interconnect SBC is listening on 2 interfaces:

```
Core interface           : udp:C.C.C.C:5060
Carrier-side interface   : udp:D.D.D.D:5060

```

TOPOS is configured only on second kamailio box (Interconnect SBC).
Record routing is used on both kamailios.


TOPOS module is configured as follows:

```
modparam("ndb_redis", "server", "name=srv1;addr=127.0.0.1;port=6379")
modparam("topos_redis", "serverid", "srv1")

modparam("topos", "storage", "redis")
modparam("topos", "contact_mode", 1)
modparam("topos", "cparam_name", "id")
modparam("topos", "rr_update", 1)

```

RR module is configured as follows:

```
modparam("rr", "enable_full_lr", 1)
modparam("rr", "append_fromtag", 1)
modparam("rr", "enable_double_rr", 1)

```


For an incomoing INVITE (direction: client to carrier) all is fine and topos can strip headers on bleg and insert the cookie on bleg contact as a param.

When Callee sends a BYE or re-INVITE, topos inserts route headers in aleg as shown in the next trace: (BYE as received from the carrier on the carrier-side interface D.D.D.D)

```
BYE sip:D.D.D.D:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKrdhsvc001omkvssqvfk0sd0185mt1.1
From: <sip:+33123456789 at my-carrier-domain;transport=UDP;user=phone>;tag=SD69ao497-1472568650-1635329820043
To: <sip:+33987654321 at my-own-domain;transport=UDP;user=phone>;tag=3ab10616
Call-ID: NjZjMDgyMGMxNzdkN2FhYmJlYzQyYjA5YWIzZThmZmI.
CSeq: 547381883 BYE
Max-Forwards: 64
Content-Length: 0
Route: <sip:D.D.D.D;r2=on;lr=on;ftag=3ab10616>,<sip:C.C.C.C;r2=on;lr=on;ftag=3ab10616>
Route: <sip:B.B.B.B;r2=on;lr=on;ftag=3ab10616;dlg_id=426.26c2>,<sip:A.A.A.A;r2=on;lr=on;ftag=3ab10616;dlg_id=426.26c2>
P-SR-XUID: atpsh-617925e5-82a5-1

```

Topos on the kamailio interconnect handles the BYE/re-INVITE and forward it to the second interface C.C.C.C as shown in the following trace:

```
BYE sip:A.A.A.A;r2=on;lr=on;ftag=3ab10616;dlg_id=426.26c2 SIP/2.0
Via: SIP/2.0/UDP 185.164.213.110;branch=z9hG4bKbd7.ab692eeb60369aa3a5e07f6b63cc9dd7.0
From: <sip:+33123456789 at my-carrier-domain;transport=UDP;user=phone>;tag=SD69ao497-1472568650-1635329820043
To: <sip:+33987654321 at my-own-domain;transport=UDP;user=phone>;tag=3ab10616
Call-ID: NjZjMDgyMGMxNzdkN2FhYmJlYzQyYjA5YWIzZThmZmI.
CSeq: 547381883 BYE
Max-Forwards: 63
Content-Length: 0
Route: <sip:C.C.C.C;r2=on;lr=on;ftag=3ab10616>
Route: <sip:B.B.B.B;r2=on;lr=on;ftag=3ab10616;dlg_id=426.26c2>
Contact: <sip:+33123456789 at 10.3.60.45;id=atpsh-617925e5-82a5-1>

```

When the BYE/re-INVITE request reaches the C.C.C.C interface, it is then forwarded directly to the ruri sip:A.A.A.A.

Or what I expect to receive in C.C.C.C interface is the following :

```
BYE <HERE SHOULD BE THE CALLER CONTACT URI> SIP/2.0
Via: SIP/2.0/UDP 80.10.231.173:5060;branch=z9hG4bKrdhsvc001omkvssqvfk0sd0185mt1.1
From: <sip:+33123456789 at my-carrier-domain;transport=UDP;user=phone>;tag=SD69ao497-1472568650-1635329820043
To: <sip:+33987654321 at my-own-domain;transport=UDP;user=phone>;tag=3ab10616
Call-ID: NjZjMDgyMGMxNzdkN2FhYmJlYzQyYjA5YWIzZThmZmI.
CSeq: 547381883 BYE
Max-Forwards: 64
Content-Length: 0
Route: <sip:D.D.D.D;r2=on;lr=on;ftag=3ab10616>
Route: <sip:C.C.C.C;r2=on;lr=on;ftag=3ab10616>
Route: <sip:B.B.B.B;r2=on;lr=on;ftag=3ab10616;dlg_id=426.26c2>
Route: <sip:A.A.A.A;r2=on;lr=on;ftag=3ab10616;dlg_id=426.26c2>
Contact: <sip:+33123456789 at 10.3.60.45;id=atpsh-617925e5-82a5-1>

```

I could work around this issue by doing the following :

1. When receiving the BYE/re-INVITE on interface D.D.D.D, I explode the Route headers, delete old route headers and create new ones in order from the result of explode
2. I update the ru with the contact uri which I stored in a htable when the initial INVITE was received


```
# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
	xdbg("new branch [$T_branch_idx] to $ru\n");

	# TOPOS FIX
	if (is_method("INVITE")) {
		$sht(tpsindlg=>$ci) = $ci;
		$sht(tpsindlg=>$ci:ft) = $ft;
		$sht(tpsindlg=>$ci:ct) = $ct;
	}
}


route[WITHINDLG] {
	if (!has_totag()) return;

	# Fix bad Route Headers caused by TOPOS module
	if ($Ri == "D.D.D.D") {
		$var(rh0) = $(hdr(Route)[0]{s.select,0,,});
		$var(rh1) = $(hdr(Route)[0]{s.select,1,,});
		$var(rh2) = $(hdr(Route)[1]{s.select,0,,});
		$var(rh3) = $(hdr(Route)[1]{s.select,1,,});

		remove_hf_re("^Route$");

		append_hf("Route: $var(rh0)\r\n");
		append_hf("Route: $var(rh1)\r\n");
		append_hf("Route: $var(rh2)\r\n");
		append_hf("Route: $var(rh3)\r\n");
	}

	# sequential request within a dialog should
	# take the path determined by record-routing
	if (loose_route()) {

	  # Fix R-URI caused by TOPOS module
		if ($Ri == "D.D.D.D") {
			if ($ci == $sht(tpsindlg=>$ci) && $tt == $sht(tpsindlg=>$ci:ft)) {
				$ru = $(sht(tpsindlg=>$ci:ct){s.strip,1}{s.striptail,1});

				if (is_method("BYE")) {
					$sht(tpsindlg=>$ci) = $null;
					$sht(tpsindlg=>$ci:ft) = $null;
					$sht(tpsindlg=>$ci:ct) = $null;
				}
			}
		}

		route(RELAY);
		exit;
	}

	if (is_method("ACK")) {
		if ( t_check_trans() ) {
			# no loose-route, but stateful ACK;
			# must be an ACK after a 487
			# or e.g. 404 from upstream server
			route(RELAY);
			exit;
		} else {
			# ACK without matching transaction ... ignore and discard
			exit;
		}
	}

	sl_send_reply("404","Not here");
	exit;
}

```

Could you please tell if this is a normal behavior by TOPOS module or is it a bug? 

Thank you.

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