fix(sip): preserve Record-Route in INVITE 2xx

crates/asterisk-sip/src/session/mod.rs

@@ -380,12 +380,27 @@ impl SipSession {
 
     /// Build a 200 OK response (for UAS).
     pub fn build_200_ok(&self) -> Option<SipMessage> {
+        self.build_200_ok_with_signaling_hostport(&self.signaling_hostport())
+    }
+
+    fn build_200_ok_with_signaling_hostport(&self, signaling_hostport: &str) -> Option<SipMessage> {
         let invite = self.invite.as_ref()?;
         let mut response = invite.create_response(200, "OK").ok()?;
 
+        // RFC 3261 section 12.1.1: a response that establishes a dialog MUST
+        // copy every Record-Route value from the request, preserving order and
+        // all parameters. Without this the UAC builds an empty route set and
+        // its ACK may bypass the proxy path that delivered the INVITE.
+        for record_route in invite.get_headers(header_names::RECORD_ROUTE) {
+            response.headers.push(SipHeader {
+                name: header_names::RECORD_ROUTE.to_string(),
+                value: record_route.to_string(),
+            });
+        }
+
         // Add Contact (NAT-scoped toward the peer: external addr/port for a
         // peer outside local_net, internal otherwise — New-3).
-        let contact = format!("<sip:asterisk@{}>", self.signaling_hostport());
+        let contact = format!("<sip:asterisk@{signaling_hostport}>");
         response.headers.push(SipHeader {
             name: header_names::CONTACT.to_string(),
             value: contact,
@@ -925,6 +940,120 @@ mod cp1_tests {
         assert_eq!(s.in_dialog_next_hop(), Some(addr("10.0.0.9:5070")));
     }
 
+    #[test]
+    fn inbound_twoxx_copies_production_shaped_route_set_and_public_contact() {
+        let invite = SipMessage::parse(
+            b"INVITE sip:service@pbx.example.invalid SIP/2.0\r\n\
+              Record-Route: <sip:edge.example.invalid:5060;r2=on;lr;ftag=fixture;did=fixture;nat=yes>\r\n\
+              Record-Route: <sip:edge.example.invalid;transport=tcp;r2=on;lr;ftag=fixture;did=fixture;nat=yes>\r\n\
+              Via: SIP/2.0/UDP edge.example.invalid:5060;branch=z9hG4bKedge\r\n\
+              Via: SIP/2.0/TCP 10.0.0.9;branch=z9hG4bKuac\r\n\
+              From: <sip:caller@10.0.0.9>;tag=fixture\r\n\
+              To: <sip:service@pbx.example.invalid>\r\n\
+              Call-ID: fixture-call\r\n\
+              CSeq: 100000001 INVITE\r\n\
+              Contact: <sip:caller@10.0.0.9>\r\n\
+              Content-Length: 0\r\n\r\n",
+        )
+        .unwrap();
+        let session = SipSession::new_inbound(
+            &invite,
+            addr("192.0.2.10:45070"),
+            addr("198.51.100.7:5060"),
+        )
+        .unwrap();
+
+        let ok = session
+            .build_200_ok_with_signaling_hostport("203.0.113.99:45070")
+            .unwrap();
+        assert_eq!(
+            ok.get_header(header_names::CONTACT),
+            Some("<sip:asterisk@203.0.113.99:45070>"),
+            "the dialog Contact must advertise the public signaling target",
+        );
+        assert_eq!(
+            ok.get_headers(header_names::RECORD_ROUTE),
+            invite.get_headers(header_names::RECORD_ROUTE),
+            "the 2xx must preserve every Record-Route value, parameter, and order",
+        );
+    }
+
+    #[test]
+    fn preserved_route_set_sends_uac_ack_to_proxy_before_public_contact() {
+        let mut uac = SipSession::new_outbound(
+            addr("192.0.2.20:5060"),
+            addr("198.51.100.10:5060"),
+        );
+        let mut routed_invite =
+            uac.build_invite("sip:service@198.51.100.10:5060");
+        routed_invite.add_header(
+            header_names::RECORD_ROUTE,
+            "<sip:198.51.100.20:5060;r2=on;lr;nat=yes>",
+        );
+        routed_invite.add_header(
+            header_names::RECORD_ROUTE,
+            "<sip:198.51.100.21:5060;transport=tcp;r2=on;lr;nat=yes>",
+        );
+        let uas = SipSession::new_inbound(
+            &routed_invite,
+            addr("10.0.0.10:45070"),
+            addr("198.51.100.20:5060"),
+        )
+        .unwrap();
+        let ok = uas
+            .build_200_ok_with_signaling_hostport("203.0.113.99:45070")
+            .unwrap();
+
+        uac.on_response(&ok);
+        let ack = uac.build_ack().unwrap();
+        assert_eq!(
+            request_uri(&ack),
+            "sip:asterisk@203.0.113.99:45070",
+            "the public Contact is the ACK's remote target",
+        );
+        assert_eq!(
+            ack.get_headers(header_names::ROUTE),
+            vec![
+                "<sip:198.51.100.21:5060;transport=tcp;r2=on;lr;nat=yes>",
+                "<sip:198.51.100.20:5060;r2=on;lr;nat=yes>",
+            ],
+            "the UAC must reverse the copied Record-Route set for its ACK",
+        );
+        assert_eq!(
+            uac.in_dialog_next_hop(),
+            Some(addr("198.51.100.21:5060")),
+            "the ACK must reach the route-set proxy rather than going directly to Contact",
+        );
+    }
+
+    #[test]
+    fn inbound_twoxx_does_not_invent_record_route() {
+        let invite = SipMessage::parse(
+            b"INVITE sip:service@192.0.2.10 SIP/2.0\r\n\
+              Via: SIP/2.0/UDP 198.51.100.7;branch=z9hG4bKdirect\r\n\
+              From: <sip:caller@198.51.100.7>;tag=direct\r\n\
+              To: <sip:service@192.0.2.10>\r\n\
+              Call-ID: direct-call\r\n\
+              CSeq: 1 INVITE\r\n\
+              Contact: <sip:caller@198.51.100.7>\r\n\
+              Content-Length: 0\r\n\r\n",
+        )
+        .unwrap();
+        let session = SipSession::new_inbound(
+            &invite,
+            addr("192.0.2.10:45070"),
+            addr("198.51.100.7:5060"),
+        )
+        .unwrap();
+        let ok = session
+            .build_200_ok_with_signaling_hostport("203.0.113.99:45070")
+            .unwrap();
+        assert!(
+            ok.get_headers(header_names::RECORD_ROUTE).is_empty(),
+            "a direct dialog must not gain a proxy route set",
+        );
+    }
+
     #[test]
     fn bye_follows_route_set_to_target_with_incremented_cseq() {
         let mut s = established_after_challenge();

tests/e2e-trunk/chime_caller.py

@@ -207,6 +207,35 @@ def parse_sdp_pts(body):
     return pcmu, tev, rip, rport
 
 
+def parse_contact_target(message):
+    """Return the 200 Contact URI and its IPv4 UDP target.
+
+    The hermetic UAC deliberately derives its ACK target from the response
+    instead of sending to the configured INVITE destination by coincidence.
+    """
+    contact = next(
+        (line.split(":", 1)[1].strip() for line in message.split("\r\n")
+         if line.lower().startswith("contact:")),
+        None,
+    )
+    if contact is None:
+        return None, None
+    match = re.search(
+        r"(?i)<\s*(sip:[^@>]+@(?P<host>\[[^]]+\]|[^:;>]+)(?::(?P<port>\d+))?[^>]*)>",
+        contact,
+    )
+    if match is None:
+        return None, None
+    uri = match.group(1)
+    host = match.group("host").strip("[]")
+    port = int(match.group("port") or 5060)
+    try:
+        target = socket.getaddrinfo(host, port, socket.AF_INET, socket.SOCK_DGRAM)[0][4]
+    except OSError:
+        return uri, None
+    return uri, target
+
+
 def main():
     ap = argparse.ArgumentParser()
     ap.add_argument("--dst-ip", required=True)
@@ -274,6 +303,7 @@ def main():
 
     totag = None
     ok_body = None
+    ok_message = None
     deadline = time.time() + 8
     while time.time() < deadline:
         try:
@@ -303,6 +333,7 @@ def main():
                 if l.lower().startswith("to:") and "tag=" in l.lower():
                     totag = l.split("tag=", 1)[1].strip()
             ok_body = msg.split("\r\n\r\n", 1)[1] if "\r\n\r\n" in msg else ""
+            ok_message = msg
             break
         # 100/180/183 -> keep waiting
     if ok_body is None:
@@ -314,6 +345,27 @@ def main():
         )
         return 2
 
+    contact_uri, contact_target = parse_contact_target(ok_message)
+    if contact_target is None:
+        log("200 OK has no resolvable SIP Contact; ACK cannot reach the dialog target")
+        _write_result(args, {"error": "unreachable_contact", "voice_rx": 0}, exit_code=2)
+        return 2
+    if args.invite_fixture:
+        expected_routes = [
+            line.split(":", 1)[1].strip()
+            for line in invite.split("\r\n")
+            if line.lower().startswith("record-route:")
+        ]
+        actual_routes = [
+            line.split(":", 1)[1].strip()
+            for line in ok_message.split("\r\n")
+            if line.lower().startswith("record-route:")
+        ]
+        if actual_routes != expected_routes:
+            log("200 OK did not preserve the INVITE Record-Route set")
+            _write_result(args, {"error": "record_route_mismatch", "voice_rx": 0}, exit_code=2)
+            return 2
+
     pcmu_pt, tev_pt, rip, rport = parse_sdp_pts(ok_body)
     if not rip or rip == "0.0.0.0":
         rip = args.dst_ip
@@ -329,12 +381,14 @@ def main():
             "transport=udp;lr;nat=yes>\r\n"
         )
     ack = (
-        f"ACK {ruri} SIP/2.0\r\n"
+        f"ACK {contact_uri} SIP/2.0\r\n"
         f"Via: SIP/2.0/UDP {args.src_ip}:{args.sip_port};branch=z9hG4bK{random.randint(0, 1 << 32):08x};rport\r\n"
         f"Max-Forwards: 70\r\n{ack_route}From: <sip:chime@{args.src_ip}>;tag={fromtag}\r\n"
         f"To: <{ruri}>;tag={totag}\r\nCall-ID: {callid}\r\nCSeq: {cseq} ACK\r\nContent-Length: 0\r\n\r\n"
     )
-    sipsock.sendto(ack.encode(), (args.dst_ip, args.dst_port))
+    # This harness represents the last proxy hop: Route remains on the wire,
+    # while the datagram is delivered to the UAS's advertised Contact target.
+    sipsock.sendto(ack.encode(), contact_target)
 
     # ---- media threads ----
     stop = threading.Event()
@@ -418,12 +472,12 @@ def send_dtmf(digit):
     stop.set()
     time.sleep(0.3)
     bye = (
-        f"BYE {ruri} SIP/2.0\r\n"
+        f"BYE {contact_uri} SIP/2.0\r\n"
         f"Via: SIP/2.0/UDP {args.src_ip}:{args.sip_port};branch=z9hG4bK{random.randint(0, 1 << 32):08x};rport\r\n"
         f"Max-Forwards: 70\r\nFrom: <sip:chime@{args.src_ip}>;tag={fromtag}\r\n"
         f"To: <{ruri}>;tag={totag}\r\nCall-ID: {callid}\r\nCSeq: {cseq + 1} BYE\r\nContent-Length: 0\r\n\r\n"
     )
-    sipsock.sendto(bye.encode(), (args.dst_ip, args.dst_port))
+    sipsock.sendto(bye.encode(), contact_target)
 
     # write RX wav (8k mono s16) from the measurement window
     with wave.open(args.rx_wav, "wb") as w:

tests/e2e-trunk/run.sh

@@ -155,17 +155,19 @@ read = all
 write = system
 EOF
 
-    # NOTE: no external_media_address / external_signaling_address — for a
-    # loopback test rustisk then advertises its own bind IP ($RK_IP) as the
-    # media address (avoids the external-address blackhole documented in the
-    # rustisk SDP path). Each endpoint is identified by a DISTINCT /32 so an
+    # Exercise the external-signaling Contact path with a reachable loopback
+    # address/port. external_media_address remains unset so the media answer
+    # uses its routed loopback address (avoids an artificial media blackhole).
+    # Each endpoint is identified by a DISTINCT /32 so an
     # inbound INVITE from the caller maps to chime-trunk and the mock's
     # REGISTER/traffic maps to qa-bridge with no ambiguity.
     cat >"$CONFIG_DIR/pjsip.conf" <<EOF
 [transport-udp]
 type = transport
 protocol = udp
 bind = $RK_IP:$RK_SIP
+external_signaling_address = $RK_IP
+external_signaling_port = $RK_SIP
 
 [chime-trunk]
 type = endpoint