forked from ttcn3/osmo-ttcn3-hacks
738 lines
26 KiB
Plaintext
738 lines
26 KiB
Plaintext
module SIP_Tests {
|
|
|
|
/* osmo-sip-connector test suite in TTCN-3
|
|
* (C) 2018-2019 Harald Welte <laforge@gnumonks.org>
|
|
* All rights reserved.
|
|
*
|
|
* Released under the terms of GNU General Public License, Version 2 or
|
|
* (at your option) any later version.
|
|
*
|
|
* SPDX-License-Identifier: GPL-2.0-or-later
|
|
*/
|
|
|
|
import from General_Types all;
|
|
import from Osmocom_Types all;
|
|
import from Native_Functions all;
|
|
import from Misc_Helpers all;
|
|
|
|
import from Osmocom_CTRL_Functions all;
|
|
import from Osmocom_CTRL_Types all;
|
|
import from Osmocom_CTRL_Adapter all;
|
|
|
|
import from TELNETasp_PortType all;
|
|
import from Osmocom_VTY_Functions all;
|
|
|
|
import from MNCC_Emulation all;
|
|
import from MNCC_Types all;
|
|
|
|
import from SDP_Types all;
|
|
import from SDP_Templates all;
|
|
|
|
import from SIP_Emulation all;
|
|
import from SIPmsg_Types all;
|
|
import from SIP_Templates all;
|
|
|
|
modulepar {
|
|
charstring mp_local_host := "127.0.0.2";
|
|
charstring mp_osmosip_host := "127.0.0.1";
|
|
integer mp_osmosip_port_ctrl := -1; /* RFU */
|
|
charstring mp_mncc := "/tmp/mncc";
|
|
}
|
|
|
|
type component test_CT extends CTRL_Adapter_CT {
|
|
var MNCC_Emulation_CT vc_MNCC;
|
|
var SIP_Emulation_CT vc_SIP;
|
|
|
|
port TELNETasp_PT SIPVTY;
|
|
}
|
|
|
|
type component ConnHdlr extends SIP_ConnHdlr, MNCC_ConnHdlr {
|
|
var ConnHdlrPars g_pars;
|
|
timer g_Tguard;
|
|
}
|
|
|
|
type record ConnHdlrPars {
|
|
float t_guard,
|
|
CallPars g_cp optional
|
|
}
|
|
|
|
type record CallPars {
|
|
boolean is_mo,
|
|
charstring calling,
|
|
charstring called,
|
|
|
|
uint32_t mncc_call_id optional,
|
|
CallParsComputed comp optional,
|
|
|
|
charstring sip_rtp_addr,
|
|
uint16_t sip_rtp_port,
|
|
charstring cn_rtp_addr,
|
|
uint16_t cn_rtp_port,
|
|
|
|
/* Send SDP to MNCC, and expect to receive SDP from MNCC. mncc_with_sdp := false tests legacy compatibility to
|
|
* the time when we did not include SDP in MNCC messages. mncc_with_sdp := true expects SDP to pass through the
|
|
* SUT osmo-sip-connector unchanged. */
|
|
boolean mncc_with_sdp
|
|
}
|
|
|
|
type record CallParsComputed {
|
|
CallidString sip_call_id,
|
|
SipAddr sip_url_ext,
|
|
SipAddr sip_url_gsm,
|
|
charstring sip_body,
|
|
integer sip_seq_nr
|
|
}
|
|
|
|
private template (value) CallPars t_CallPars(boolean is_mo, boolean mncc_with_sdp := true) := {
|
|
is_mo := is_mo,
|
|
calling := "12345",
|
|
called := "98766",
|
|
|
|
mncc_call_id := omit,
|
|
comp := omit,
|
|
sip_rtp_addr := "1.2.3.4",
|
|
sip_rtp_port := 1234,
|
|
cn_rtp_addr := "5.6.7.8",
|
|
cn_rtp_port := 5678,
|
|
mncc_with_sdp := mncc_with_sdp
|
|
}
|
|
|
|
private function f_CallPars_compute(inout CallPars cp) {
|
|
if (cp.is_mo) {
|
|
cp.comp.sip_url_ext := valueof(ts_SipAddr(ts_HostPort(mp_local_host, 5060),
|
|
ts_UserInfo(cp.called)));
|
|
cp.comp.sip_url_gsm := valueof(ts_SipAddr(ts_HostPort(mp_osmosip_host, 5060),
|
|
ts_UserInfo(cp.calling)));
|
|
cp.mncc_call_id := f_sip_rand_seq_nr();
|
|
} else {
|
|
cp.comp.sip_url_ext := valueof(ts_SipAddr(ts_HostPort(mp_local_host, 5060),
|
|
ts_UserInfo(cp.calling)));
|
|
cp.comp.sip_url_gsm := valueof(ts_SipAddr(ts_HostPort(mp_osmosip_host, 5060),
|
|
ts_UserInfo(cp.called)));
|
|
cp.comp.sip_call_id := hex2str(f_rnd_hexstring(15));
|
|
}
|
|
cp.comp.sip_seq_nr := f_sip_rand_seq_nr();
|
|
cp.comp.sip_body := "";
|
|
}
|
|
|
|
function f_init_mncc(charstring id) runs on test_CT {
|
|
id := id & "-MNCC";
|
|
var MnccOps ops := {
|
|
create_cb := refers(MNCC_Emulation.ExpectedCreateCallback),
|
|
unitdata_cb := refers(MNCC_Emulation.DummyUnitdataCallback)
|
|
};
|
|
|
|
vc_MNCC := MNCC_Emulation_CT.create(id);
|
|
map(vc_MNCC:MNCC, system:MNCC_CODEC_PT);
|
|
vc_MNCC.start(MNCC_Emulation.main(ops, id, mp_mncc, true));
|
|
}
|
|
|
|
function f_init() runs on test_CT {
|
|
//f_ipa_ctrl_start_client(mp_osmosip_host, mp_osmosip_port_ctrl);
|
|
f_init_mncc("SIP_Test");
|
|
log("end of f_init_mncc");
|
|
f_init_sip(vc_SIP, "SIP_Test");
|
|
log("end of f_init_sip");
|
|
|
|
map(self:SIPVTY, system:SIPVTY);
|
|
f_vty_set_prompts(SIPVTY);
|
|
f_vty_transceive(SIPVTY, "enable");
|
|
log("end of f_init");
|
|
}
|
|
|
|
type function void_fn(charstring id) runs on ConnHdlr;
|
|
|
|
function f_start_handler(void_fn fn, ConnHdlrPars pars)
|
|
runs on test_CT return ConnHdlr {
|
|
var ConnHdlr vc_conn;
|
|
var charstring id := testcasename();
|
|
|
|
vc_conn := ConnHdlr.create(id);
|
|
|
|
connect(vc_conn:SIP, vc_SIP:CLIENT);
|
|
connect(vc_conn:SIP_PROC, vc_SIP:CLIENT_PROC);
|
|
|
|
connect(vc_conn:MNCC, vc_MNCC:MNCC_CLIENT);
|
|
connect(vc_conn:MNCC_PROC, vc_MNCC:MNCC_PROC);
|
|
|
|
vc_conn.start(f_handler_init(fn, id, pars));
|
|
return vc_conn;
|
|
}
|
|
|
|
private altstep as_Tguard() runs on ConnHdlr {
|
|
[] g_Tguard.timeout {
|
|
setverdict(fail, "Tguard timeout");
|
|
mtc.stop;
|
|
}
|
|
}
|
|
|
|
private function f_handler_init(void_fn fn, charstring id, ConnHdlrPars pars)
|
|
runs on ConnHdlr {
|
|
g_pars := pars;
|
|
g_Tguard.start(pars.t_guard);
|
|
activate(as_Tguard());
|
|
|
|
/* call the user-supied test case function */
|
|
fn.apply(id);
|
|
}
|
|
|
|
|
|
template (value) ConnHdlrPars t_Pars := {
|
|
t_guard := 30.0,
|
|
g_cp := omit
|
|
}
|
|
|
|
altstep as_SIP_expect_resp(template PDU_SIP_Response sip_expect) runs on ConnHdlr
|
|
{
|
|
[] SIP.receive(sip_expect);
|
|
[] SIP.receive {
|
|
log("FAIL: expected SIP message ", sip_expect);
|
|
Misc_Helpers.f_shutdown(__BFILE__, __LINE__, fail, "Received unexpected SIP message");
|
|
}
|
|
}
|
|
|
|
function f_SIP_expect_req(template PDU_SIP_Request sip_expect) runs on ConnHdlr return PDU_SIP_Request
|
|
{
|
|
var PDU_SIP_Request rx;
|
|
alt {
|
|
[] SIP.receive(sip_expect) -> value rx;
|
|
[] SIP.receive {
|
|
log("FAIL: expected SIP message ", sip_expect);
|
|
Misc_Helpers.f_shutdown(__BFILE__, __LINE__, fail, "Received unexpected SIP message");
|
|
}
|
|
}
|
|
return rx;
|
|
}
|
|
|
|
/* Update 'last_sdp', and match with expectation of what the current SDP should be.
|
|
* Useful to ensure that MNCC or SIP send and possibly resend only the expected SDP.
|
|
* last_sdp keeps the last non-empty rx_sdp, across multiple check_sdp() invocations.
|
|
* rx_sdp is the SDP charstring just received. If it is nonempty, update last_sdp to rx_sdp.
|
|
* After updating last_sdp as appropriate, match last_sdp with expect_sdp. */
|
|
private function check_sdp(inout charstring last_sdp,
|
|
charstring rx_sdp,
|
|
template charstring expect_sdp)
|
|
{
|
|
/* If there is new SDP, store it. */
|
|
if (lengthof(rx_sdp) > 0) {
|
|
if (last_sdp != rx_sdp) {
|
|
log("SDP update from ", last_sdp, " to ", rx_sdp);
|
|
}
|
|
|
|
/* If MNCC sent SDP data, remember it as the last valid SDP */
|
|
last_sdp := rx_sdp;
|
|
}
|
|
/* Validate expectations of the SDP data */
|
|
if (not match(last_sdp, expect_sdp)) {
|
|
log("FAIL: expected SDP ", expect_sdp, " but got ", last_sdp);
|
|
Misc_Helpers.f_shutdown(__BFILE__, __LINE__, fail, "unexpected SDP");
|
|
}
|
|
}
|
|
|
|
/* Establish a mobile terminated call described in 'cp' */
|
|
function f_establish_mt(inout CallPars cp) runs on ConnHdlr {
|
|
var template SipAddr sip_addr_gsm := tr_SipAddr_from_val(cp.comp.sip_url_gsm);
|
|
var template SipAddr sip_addr_ext := tr_SipAddr_from_val(cp.comp.sip_url_ext);
|
|
var MNCC_PDU mncc;
|
|
|
|
/* The last SDP that the MSC received via MNCC from osmo-sip-connector */
|
|
var charstring sdp_to_msc := "";
|
|
/* At first, allow any empty and nonempty SDP. As the test progresses, this may expect specific SDP instead. */
|
|
var template charstring expect_sdp_to_msc := *;
|
|
|
|
/* If cp.mncc_with_sdp == true, expect SDP forwarding like this:
|
|
*
|
|
* SDP1: SIP agent's RTP and codec info
|
|
* SDP2: osmo-msc's RTP and codec info
|
|
*
|
|
* MNCC osmo-sip-connector SIP
|
|
* |<--SDP1----- SIP Invite
|
|
* |-----------> SIP (Invite) Trying
|
|
* <--SDP1-------| MNCC SETUP req
|
|
* ------------->| MNCC CALL CONF ind
|
|
* <-------------| MNCC RTP CREATE (SDP optional, still unchanged from SDP1)
|
|
* -------SDP2-->| MNCC RTP CREATE
|
|
* ------------->| MNCC ALERT ind
|
|
* |--------------> SIP (Invite) Ringing
|
|
* (MT picks up) |
|
|
* ------------->| MNCC SETUP CNF
|
|
* <-------------| MNCC RTP CONNECT (SDP optional, still unchanged from SDP1)
|
|
* |--------SDP2--> SIP (Invite) OK
|
|
* |<-------------- SIP ACK
|
|
* <-------------| MNCC SETUP COMPL (SDP optional, still unchanged from SDP1)
|
|
*/
|
|
|
|
/* Ask MNCC_Emulation to "expect" a call to the given called number */
|
|
f_create_mncc_expect(cp.called);
|
|
|
|
/* OSC <- SIP: A party sends SIP invite for a MT-call into OSC */
|
|
SIP.send(ts_SIP_INVITE(cp.comp.sip_call_id, cp.comp.sip_url_ext, cp.comp.sip_url_gsm,
|
|
cp.comp.sip_seq_nr, cp.comp.sip_body));
|
|
if (cp.mncc_with_sdp) {
|
|
/* We just sent SDP via SIP, now expect the same SDP in MNCC to the MSC */
|
|
expect_sdp_to_msc := cp.comp.sip_body;
|
|
}
|
|
|
|
/* OSC -> SIP */
|
|
as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm,
|
|
tr_Via_from(tr_HostPort(sip_addr_ext.addr.nameAddr.addrSpec.hostPort)),
|
|
*,
|
|
"INVITE", 100, ?, "Trying", *));
|
|
|
|
alt {
|
|
/* MSC <- OSC: OSC generates MNCC_SETUP_REQ from INVITE */
|
|
[] MNCC.receive(tr_MNCC_SETUP_req) -> value mncc {
|
|
cp.mncc_call_id := mncc.u.signal.callref;
|
|
/* Expect the SDP sent via SIP to arrive in MNCC */
|
|
check_sdp(sdp_to_msc, mncc.u.signal.sdp, expect_sdp_to_msc);
|
|
}
|
|
[] SIP.receive {
|
|
setverdict(fail, "Received unexpected SIP response");
|
|
SIP.send(ts_SIP_ACK(cp.comp.sip_call_id, cp.comp.sip_url_ext, cp.comp.sip_url_gsm,
|
|
cp.comp.sip_seq_nr, omit));
|
|
mtc.stop;
|
|
}
|
|
}
|
|
|
|
/* MSC -> OSC: After MS sends CALL CONF in response to SETUP */
|
|
MNCC.send(ts_MNCC_CALL_CONF_ind(cp.mncc_call_id));
|
|
/* MSC <- OSC: OSC asks MSC to create RTP socket */
|
|
MNCC.receive(tr_MNCC_RTP_CREATE(cp.mncc_call_id)) -> value mncc {
|
|
check_sdp(sdp_to_msc, mncc.u.rtp.sdp, expect_sdp_to_msc);
|
|
}
|
|
|
|
/* MSC -> OSC: SDP that the MSC will send via MNCC */
|
|
var charstring cn_sdp := "v=0\r\no=Osmocom 0 0 IN IP4 1.1.1.1\r\ns=GSM Call\r\nc=IN " &
|
|
f_sdp_addr2addrtype(cp.cn_rtp_addr) & " " & cp.cn_rtp_addr &
|
|
"\r\nt=0 0\r\nm=audio " & int2str(cp.cn_rtp_port) &
|
|
" RTP/AVP 0\r\na=rtpmap:0 GSM/8000\r\n";
|
|
/* OSC -> SIP: what SDP to expect in SIP from osmo-sip-connector */
|
|
var template charstring expect_sdp_to_sip := pattern "*" & cp.cn_rtp_addr & "*";
|
|
|
|
mncc := valueof(ts_MNCC_RTP_CREATE(cp.mncc_call_id));
|
|
mncc.u.rtp.is_ipv6 := f_addr_is_ipv6(cp.cn_rtp_addr);
|
|
mncc.u.rtp.ip := f_addrstr2addr(cp.cn_rtp_addr);
|
|
mncc.u.rtp.rtp_port := cp.cn_rtp_port;
|
|
if (cp.mncc_with_sdp) {
|
|
/* MSC -> OSC: tell OSC our RTP info in SDP form */
|
|
mncc.u.rtp.sdp := cn_sdp;
|
|
/* OSC -> SIP: and expect it unchanged on SIP later, but allow osmo-sip-connector to append an
|
|
* "a=sendrecv;" */
|
|
expect_sdp_to_sip := pattern cn_sdp & "*";
|
|
}
|
|
MNCC.send(mncc);
|
|
|
|
/* MSC -> OSC: After MS is ringing and sent CC ALERTING */
|
|
MNCC.send(ts_MNCC_ALERT_ind(cp.mncc_call_id));
|
|
|
|
/* Now expect SIP response "Ringing" back to MO, containing the same SDP information as in the MNCC RTP CREATE
|
|
* sent to OSC above */
|
|
SIP.clear;
|
|
|
|
/* 180 Ringing should not contain any SDP. */
|
|
as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm,
|
|
tr_Via_from(tr_HostPort(sip_addr_ext.addr.nameAddr.addrSpec.hostPort)),
|
|
*,
|
|
"INVITE", 180, ?, "Ringing", omit));
|
|
|
|
/* MSC -> OSC: After MT user has picked up and sent CC CONNECT */
|
|
MNCC.send(ts_MNCC_SETUP_CNF(cp.mncc_call_id));
|
|
|
|
SIP.clear;
|
|
/* MSC <- OSC: OSC asks MSC to connect its RTP stream to remote end */
|
|
MNCC.receive(tr_MNCC_RTP_CONNECT(cp.mncc_call_id, f_addrstr2addr(cp.sip_rtp_addr), cp.sip_rtp_port))
|
|
-> value mncc {
|
|
check_sdp(sdp_to_msc, mncc.u.rtp.sdp, expect_sdp_to_msc);
|
|
}
|
|
|
|
/* OSC -> SIP: OSC confirms call establishment to SIP side */
|
|
as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm,
|
|
tr_Via_from(tr_HostPort(sip_addr_ext.addr.nameAddr.addrSpec.hostPort)),
|
|
contact := ?,
|
|
method := "INVITE", status_code := 200,
|
|
seq_nr := ?, reason := "OK",
|
|
body := expect_sdp_to_sip));
|
|
|
|
/* OSC <- SIP: SIP world acknowledges "200 OK" */
|
|
SIP.send(ts_SIP_ACK(cp.comp.sip_call_id, cp.comp.sip_url_ext, cp.comp.sip_url_gsm,
|
|
cp.comp.sip_seq_nr, omit));
|
|
/* MSC <- OSC: OSC sends SETUP COMPL to MNCC (which triggers CC CONNECT ACK */
|
|
MNCC.receive(tr_MNCC_SETUP_COMPL_req(cp.mncc_call_id)) -> value mncc {
|
|
check_sdp(sdp_to_msc, mncc.u.signal.sdp, expect_sdp_to_msc);
|
|
}
|
|
}
|
|
|
|
/* Establish a mobile originated call described in 'cp' */
|
|
function f_establish_mo(inout CallPars cp) runs on ConnHdlr {
|
|
var MNCC_number dst := valueof(ts_MNCC_number(cp.called, GSM48_TON_UNKNOWN));
|
|
var MNCC_number src := valueof(ts_MNCC_number(cp.calling, GSM48_TON_UNKNOWN));
|
|
var template SipAddr sip_addr_gsm := tr_SipAddr_from_val(cp.comp.sip_url_gsm);
|
|
var template SipAddr sip_addr_ext := tr_SipAddr_from_val(cp.comp.sip_url_ext);
|
|
var PDU_SIP_Request sip_req;
|
|
var integer seq_nr;
|
|
var MNCC_PDU mncc;
|
|
|
|
/* The last SDP that the MSC received via MNCC from osmo-sip-connector */
|
|
var charstring sdp_to_msc := "";
|
|
/* At first, allow any empty and nonempty SDP. As the test progresses, this may expect specific SDP instead. */
|
|
var template charstring expect_sdp_to_msc := *;
|
|
|
|
/* If cp.mncc_with_sdp == true, expect SDP forwarding like this:
|
|
*
|
|
* SDP1: osmo-msc's RTP and codec info
|
|
* SDP2: SIP agent's RTP and codec info
|
|
*
|
|
* MNCC osmo-sip-connector SIP
|
|
* -------SDP1-->| MNCC SETUP ind
|
|
* <-------------| MNCC RTP CREATE (?)
|
|
* |-----SDP1--> SIP Invite
|
|
* |<----------- SIP (Invite) Trying
|
|
* <-------------| MNCC CALL PROC req
|
|
* |<----------- SIP (Invite) Ringing
|
|
* <-------------| MNCC ALERT req
|
|
* | (MT picks up)
|
|
* |<--SDP2----- SIP (Invite) OK
|
|
* <--SDP2-------| MNCC RTP CONNECT (SDP optional, still unchanged from SDP2)
|
|
* <-------------| MNCC SETUP rsp (SDP optional, still unchanged from SDP2)
|
|
* ------------->| MNCC SETUP COMPL ind (SDP optional, still unchanged from SDP1)
|
|
* |------------> SIP ACK
|
|
*/
|
|
|
|
var charstring cn_sdp := "v=0\r\no=Osmocom 0 0 IN IP4 1.1.1.1\r\ns=GSM Call\r\nc=IN " &
|
|
f_sdp_addr2addrtype(cp.cn_rtp_addr) & " " & cp.cn_rtp_addr &
|
|
"\r\nt=0 0\r\nm=audio " & int2str(cp.cn_rtp_port) &
|
|
" RTP/AVP 0\r\na=rtpmap:0 GSM/8000\r\n";
|
|
|
|
f_create_sip_expect(cp.comp.sip_url_ext.addr.nameAddr.addrSpec);
|
|
|
|
/* MSC -> OSC: MSC sends SETUP.ind after CC SETUP was received from MS */
|
|
mncc := valueof(ts_MNCC_SETUP_ind(cp.mncc_call_id, dst, src, "262420123456789"));
|
|
if (cp.mncc_with_sdp) {
|
|
mncc.u.signal.sdp := cn_sdp;
|
|
}
|
|
MNCC.send(mncc);
|
|
|
|
/* MSC <- OSC: Create GSM side RTP socket */
|
|
MNCC.receive(tr_MNCC_RTP_CREATE(cp.mncc_call_id)) {
|
|
mncc := valueof(ts_MNCC_RTP_CREATE(cp.mncc_call_id));
|
|
mncc.u.rtp.payload_msg_type := oct2int('0300'O);
|
|
/* FIXME: makes no sense to send cp.cn_rtp_addr back to the cn. */
|
|
mncc.u.rtp.is_ipv6 := f_addr_is_ipv6(cp.cn_rtp_addr);
|
|
mncc.u.rtp.ip := f_addrstr2addr(cp.cn_rtp_addr);
|
|
mncc.u.rtp.rtp_port := cp.cn_rtp_port;
|
|
MNCC.send(mncc);
|
|
}
|
|
|
|
/* OSC -> SIP: Send INVITE with GSM side IP/Port in SDP */
|
|
var template charstring expect_sdp_to_sip := ?;
|
|
if (cp.mncc_with_sdp) {
|
|
/* Expect the same SDP as sent to osmo-sip-connector in MNCC, and allow osmo-sip-connector to append an
|
|
* "a=sendrecv;" */
|
|
expect_sdp_to_sip := pattern cn_sdp & "*";
|
|
}
|
|
sip_req := f_SIP_expect_req(tr_SIP_INVITE(?, sip_addr_gsm, sip_addr_ext, ?, expect_sdp_to_sip));
|
|
cp.comp.sip_url_gsm.params := sip_req.msgHeader.fromField.fromParams;
|
|
cp.comp.sip_call_id := sip_req.msgHeader.callId.callid;
|
|
seq_nr := sip_req.msgHeader.cSeq.seqNumber;
|
|
|
|
/* OSC <- SIP: Notify call is proceeding */
|
|
SIP.send(ts_SIP_Response(cp.comp.sip_call_id, cp.comp.sip_url_gsm, cp.comp.sip_url_ext,
|
|
"INVITE", 100, seq_nr, "Trying", sip_req.msgHeader.via));
|
|
/* MSC <- OSC: "100 Trying" translated to MNCC_CALL_PROC_REQ */
|
|
MNCC.receive(tr_MNCC_CALL_PROC_req(cp.mncc_call_id)) -> value mncc {
|
|
check_sdp(sdp_to_msc, mncc.u.signal.sdp, "");
|
|
}
|
|
|
|
/* OSC <- SIP: SIP-terminated user is ringing now. 180 Ringing should not contain any SDP. */
|
|
SIP.send(ts_SIP_Response(cp.comp.sip_call_id, cp.comp.sip_url_gsm, cp.comp.sip_url_ext,
|
|
"INVITE", 180, seq_nr, "Ringing", sip_req.msgHeader.via, omit));
|
|
|
|
/* MSC <- OSC: "180 Ringing" translated to MNCC_ALERT_REQ */
|
|
MNCC.receive(tr_MNCC_ALERT_req(cp.mncc_call_id)) -> value mncc {
|
|
check_sdp(sdp_to_msc, mncc.u.signal.sdp, expect_sdp_to_msc);
|
|
}
|
|
|
|
/* OSC <- SIP: SIP-terminated user has accepted the call */
|
|
SIP.send(ts_SIP_Response(cp.comp.sip_call_id, cp.comp.sip_url_gsm, cp.comp.sip_url_ext,
|
|
"INVITE", 200, seq_nr, "OK", sip_req.msgHeader.via,
|
|
cp.comp.sip_body));
|
|
|
|
if (cp.mncc_with_sdp) {
|
|
/* If we expect SDP forwarding, from now on expect MNCC to reflect the SDP that we just sent on SIP. */
|
|
expect_sdp_to_msc := cp.comp.sip_body;
|
|
}
|
|
/* If we don't expect SDP forwarding, just keep expect_sdp_to_msc := *. */
|
|
|
|
MNCC.receive(tr_MNCC_RTP_CONNECT(cp.mncc_call_id)) -> value mncc {
|
|
check_sdp(sdp_to_msc, mncc.u.rtp.sdp, expect_sdp_to_msc);
|
|
}
|
|
/* MSC <- OSC: "200 OK" translated to MNCC_SETUP_RSP */
|
|
MNCC.receive(tr_MNCC_SETUP_rsp(cp.mncc_call_id)) -> value mncc {
|
|
check_sdp(sdp_to_msc, mncc.u.signal.sdp, expect_sdp_to_msc);
|
|
}
|
|
|
|
/* MSC -> OSC: CC CONNECT ACK was received from MS */
|
|
MNCC.send(ts_MNCC_SETUP_COMPL_ind(cp.mncc_call_id));
|
|
/* OSC -> SIP: Acknowledge the call */
|
|
SIP.receive(tr_SIP_ACK(cp.comp.sip_call_id, sip_addr_gsm, sip_addr_ext, ?, omit));
|
|
}
|
|
|
|
/* Release call from the mobile side */
|
|
function f_release_mobile(inout CallPars cp) runs on ConnHdlr {
|
|
var template SipAddr sip_addr_gsm := tr_SipAddr_from_val(cp.comp.sip_url_gsm);
|
|
var template SipAddr sip_addr_ext := tr_SipAddr_from_val(cp.comp.sip_url_ext);
|
|
var PDU_SIP_Request sip_req;
|
|
SIP.clear;
|
|
/* MSC -> OSC: Simulate a CC DISCONNET from the MT user */
|
|
MNCC.send(ts_MNCC_DISC_ind(cp.mncc_call_id, ts_MNCC_cause(0)));
|
|
|
|
/* OSC -> SIP: Expect BYE from OSC to SIP side */
|
|
sip_req := f_SIP_expect_req(tr_SIP_BYE(cp.comp.sip_call_id, sip_addr_gsm, sip_addr_ext, ?, *));
|
|
cp.comp.sip_url_gsm.params := sip_req.msgHeader.fromField.fromParams;
|
|
|
|
/* OSC <- SIP: Acknowledge the BYE */
|
|
SIP.send(ts_SIP_Response(cp.comp.sip_call_id, cp.comp.sip_url_gsm, cp.comp.sip_url_ext,
|
|
"BYE", 200, sip_req.msgHeader.cSeq.seqNumber, "OK",
|
|
sip_req.msgHeader.via));
|
|
/* MSC <- OSC: Send REL_REQ to MSC, triggers CC RELEASE REQ to MS */
|
|
MNCC.receive(tr_MNCC_REL_req(cp.mncc_call_id)); // CAUSE?
|
|
/* MSC -> OSC: MS has responded with CC CLEAR COMPL, triggers MNCC_REL_CNF */
|
|
MNCC.send(ts_MNCC_REL_cnf(cp.mncc_call_id, ts_MNCC_cause(0)));
|
|
}
|
|
|
|
/* Release call from the SIP side */
|
|
function f_release_sip(inout CallPars cp) runs on ConnHdlr {
|
|
var template SipAddr sip_addr_gsm := tr_SipAddr_from_val(cp.comp.sip_url_gsm);
|
|
var template SipAddr sip_addr_ext := tr_SipAddr_from_val(cp.comp.sip_url_ext);
|
|
/* OSC <- SIP: SIP-side sends a BYE to OSC */
|
|
SIP.send(ts_SIP_BYE(cp.comp.sip_call_id, cp.comp.sip_url_ext, cp.comp.sip_url_gsm,
|
|
cp.comp.sip_seq_nr, omit));
|
|
/* MSC <- OSC: Expect OSC to cause MNCC Disconnect Request */
|
|
MNCC.receive(tr_MNCC_DISC_req(cp.mncc_call_id));
|
|
/* MSC -> OSC: Indicate GSM side release */
|
|
MNCC.send(ts_MNCC_REL_ind(cp.mncc_call_id, ts_MNCC_cause(0)));
|
|
/* OSC -> SIP: Confirmation to SIP side */
|
|
as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm,
|
|
tr_Via_from(tr_HostPort(sip_addr_ext.addr.nameAddr.addrSpec.hostPort)),
|
|
*,
|
|
"BYE", 200, cp.comp.sip_seq_nr, "OK", omit));
|
|
}
|
|
|
|
/* Successful MT Call, which is subsequently released by GSM side */
|
|
private function f_TC_mt_success_rel_gsm(charstring id) runs on ConnHdlr {
|
|
var CallPars cp := g_pars.g_cp;
|
|
f_CallPars_compute(cp);
|
|
cp.comp.sip_body := "v=0\r\no=Osmocom 0 0 IN IP4 1.1.1.1\r\ns=GSM Call\r\nc=IN " &
|
|
f_sdp_addr2addrtype(cp.sip_rtp_addr) & " " & cp.sip_rtp_addr &
|
|
"\r\nt=0 0\r\nm=audio " & int2str(cp.sip_rtp_port) &
|
|
" RTP/AVP 0\r\na=rtpmap:0 GSM/8000\r\n";
|
|
f_sleep(3.0)
|
|
|
|
f_establish_mt(cp);
|
|
/* now call is fully established */
|
|
f_sleep(2.0);
|
|
f_release_mobile(cp);
|
|
setverdict(pass);
|
|
}
|
|
testcase TC_mt_success_rel_gsm() runs on test_CT {
|
|
var ConnHdlrPars pars;
|
|
var ConnHdlr vc_conn;
|
|
f_init();
|
|
pars := valueof(t_Pars);
|
|
pars.g_cp := valueof(t_CallPars(false, false));
|
|
vc_conn := f_start_handler(refers(f_TC_mt_success_rel_gsm), pars);
|
|
vc_conn.done;
|
|
}
|
|
testcase TC_mt_success_rel_gsm_ipv6() runs on test_CT {
|
|
var ConnHdlrPars pars;
|
|
var ConnHdlr vc_conn;
|
|
f_init();
|
|
pars := valueof(t_Pars);
|
|
pars.g_cp := valueof(t_CallPars(false, false));
|
|
pars.g_cp.sip_rtp_addr := "::1";
|
|
pars.g_cp.cn_rtp_addr := "::2";
|
|
vc_conn := f_start_handler(refers(f_TC_mt_success_rel_gsm), pars);
|
|
vc_conn.done;
|
|
}
|
|
|
|
/* Successful MT Call, which is subsequently released by SIP side */
|
|
private function f_TC_mt_success_rel_sip(charstring id) runs on ConnHdlr {
|
|
var CallPars cp := g_pars.g_cp;
|
|
f_CallPars_compute(cp);
|
|
cp.comp.sip_body := "v=0\r\no=Osmocom 0 0 IN IP4 1.1.1.1\r\ns=GSM Call\r\nc=IN " &
|
|
f_sdp_addr2addrtype(cp.sip_rtp_addr) & " " & cp.sip_rtp_addr &
|
|
"\r\nt=0 0\r\nm=audio " & int2str(cp.sip_rtp_port) &
|
|
" RTP/AVP 0\r\na=rtpmap:0 GSM/8000\r\n";
|
|
f_sleep(3.0)
|
|
|
|
f_establish_mt(cp);
|
|
/* now call is fully established */
|
|
f_sleep(2.0);
|
|
f_release_sip(cp);
|
|
setverdict(pass);
|
|
}
|
|
testcase TC_mt_success_rel_sip() runs on test_CT {
|
|
var ConnHdlrPars pars;
|
|
var ConnHdlr vc_conn;
|
|
f_init();
|
|
pars := valueof(t_Pars);
|
|
pars.g_cp := valueof(t_CallPars(false, false));
|
|
vc_conn := f_start_handler(refers(f_TC_mt_success_rel_sip), pars);
|
|
vc_conn.done;
|
|
}
|
|
|
|
|
|
/* Successful MO Call, which is subsequently released by GSM side */
|
|
private function f_TC_mo_success_rel_gsm(charstring id) runs on ConnHdlr {
|
|
var CallPars cp := g_pars.g_cp;
|
|
f_CallPars_compute(cp);
|
|
cp.comp.sip_body := "v=0\r\no=Osmocom 0 0 IN IP4 1.1.1.1\r\ns=GSM Call\r\nc=IN " &
|
|
f_sdp_addr2addrtype(cp.sip_rtp_addr) & " " & cp.sip_rtp_addr &
|
|
"\r\nt=0 0\r\nm=audio " & int2str(cp.sip_rtp_port) &
|
|
" RTP/AVP 0\r\na=rtpmap:0 GSM/8000\r\n";
|
|
f_sleep(3.0)
|
|
|
|
f_establish_mo(cp);
|
|
/* now call is fully established */
|
|
f_sleep(2.0);
|
|
f_release_mobile(cp);
|
|
setverdict(pass);
|
|
}
|
|
testcase TC_mo_success_rel_gsm() runs on test_CT {
|
|
var ConnHdlrPars pars;
|
|
var ConnHdlr vc_conn;
|
|
f_init();
|
|
pars := valueof(t_Pars);
|
|
pars.g_cp := valueof(t_CallPars(true, false));
|
|
vc_conn := f_start_handler(refers(f_TC_mo_success_rel_gsm), pars);
|
|
vc_conn.done;
|
|
}
|
|
testcase TC_mo_success_rel_gsm_ipv6() runs on test_CT {
|
|
var ConnHdlrPars pars;
|
|
var ConnHdlr vc_conn;
|
|
f_init();
|
|
pars := valueof(t_Pars);
|
|
pars.g_cp := valueof(t_CallPars(true, false));
|
|
pars.g_cp.sip_rtp_addr := "::1";
|
|
pars.g_cp.cn_rtp_addr := "::2";
|
|
vc_conn := f_start_handler(refers(f_TC_mo_success_rel_gsm), pars);
|
|
vc_conn.done;
|
|
}
|
|
|
|
/* Successful MO Call, which is subsequently released by SIP side */
|
|
private function f_TC_mo_success_rel_sip(charstring id) runs on ConnHdlr {
|
|
var CallPars cp := g_pars.g_cp;
|
|
f_CallPars_compute(cp);
|
|
cp.comp.sip_body := "v=0\r\no=Osmocom 0 0 IN IP4 1.1.1.1\r\ns=GSM Call\r\nc=IN " &
|
|
f_sdp_addr2addrtype(cp.sip_rtp_addr) & " " & cp.sip_rtp_addr &
|
|
"\r\nt=0 0\r\nm=audio " & int2str(cp.sip_rtp_port) &
|
|
" RTP/AVP 0\r\na=rtpmap:0 GSM/8000\r\n";
|
|
f_sleep(3.0)
|
|
|
|
f_establish_mo(cp);
|
|
/* now call is fully established */
|
|
f_sleep(2.0);
|
|
f_release_sip(cp);
|
|
setverdict(pass);
|
|
}
|
|
testcase TC_mo_success_rel_sip() runs on test_CT {
|
|
var ConnHdlrPars pars;
|
|
var ConnHdlr vc_conn;
|
|
f_init();
|
|
pars := valueof(t_Pars);
|
|
pars.g_cp := valueof(t_CallPars(is_mo := true, mncc_with_sdp := false));
|
|
vc_conn := f_start_handler(refers(f_TC_mo_success_rel_sip), pars);
|
|
vc_conn.done;
|
|
}
|
|
|
|
/* SETUP followed by DISC results in lingering B-leg (OS#3518)*/
|
|
private function f_TC_mo_setup_disc_late_rtp(charstring id) runs on ConnHdlr {
|
|
var CallPars cp := g_pars.g_cp;
|
|
f_CallPars_compute(cp);
|
|
cp.comp.sip_body := "v=0\r\no=Osmocom 0 0 IN IP4 1.1.1.1\r\ns=GSM Call\r\nc=IN " &
|
|
f_sdp_addr2addrtype(cp.sip_rtp_addr) & " " & cp.sip_rtp_addr &
|
|
"\r\nt=0 0\r\nm=audio " & int2str(cp.sip_rtp_port) &
|
|
" RTP/AVP 0\r\na=rtpmap:0 GSM/8000\r\n";
|
|
f_sleep(3.0);
|
|
|
|
var MNCC_number dst := valueof(ts_MNCC_number(cp.called, GSM48_TON_UNKNOWN));
|
|
var MNCC_number src := valueof(ts_MNCC_number(cp.calling, GSM48_TON_UNKNOWN));
|
|
var template SipAddr sip_addr_gsm := tr_SipAddr_from_val(cp.comp.sip_url_gsm);
|
|
var template SipAddr sip_addr_ext := tr_SipAddr_from_val(cp.comp.sip_url_ext);
|
|
|
|
f_create_sip_expect(cp.comp.sip_url_ext.addr.nameAddr.addrSpec);
|
|
|
|
/* MSC -> OSC: MSC sends SETUP.ind after CC SETUP was received from MS */
|
|
MNCC.send(ts_MNCC_SETUP_ind(cp.mncc_call_id, dst, src, "262420123456789"));
|
|
|
|
/* MSC -> OSC: Simulate a CC DISCONNET from the MT user *before* responding to the RTP_CREATE */
|
|
MNCC.send(ts_MNCC_DISC_ind(cp.mncc_call_id, ts_MNCC_cause(0)));
|
|
|
|
/* MSC <- OSC: Create GSM side RTP socket (too late) */
|
|
MNCC.receive(tr_MNCC_RTP_CREATE(cp.mncc_call_id)) {
|
|
var MNCC_PDU mncc := valueof(ts_MNCC_RTP_CREATE(cp.mncc_call_id));
|
|
mncc.u.rtp.payload_msg_type := oct2int('0300'O);
|
|
mncc.u.rtp.is_ipv6 := f_addr_is_ipv6(cp.cn_rtp_addr);
|
|
mncc.u.rtp.ip := f_addrstr2addr(cp.cn_rtp_addr);
|
|
mncc.u.rtp.rtp_port := cp.cn_rtp_port;
|
|
MNCC.send(mncc);
|
|
}
|
|
|
|
/* OSC -> SIP: We should never receive INVITE */
|
|
timer T := 10.0;
|
|
T.start;
|
|
alt {
|
|
[] SIP.receive(tr_SIP_INVITE(?, sip_addr_gsm, sip_addr_ext, ?, ?)) {
|
|
setverdict(fail, "Received unexpected INVITE");
|
|
}
|
|
[] T.timeout {
|
|
setverdict(pass);
|
|
}
|
|
}
|
|
}
|
|
testcase TC_mo_setup_disc_late_rtp() runs on test_CT {
|
|
var ConnHdlrPars pars;
|
|
var ConnHdlr vc_conn;
|
|
f_init();
|
|
pars := valueof(t_Pars);
|
|
pars.g_cp := valueof(t_CallPars(is_mo := true, mncc_with_sdp := false));
|
|
vc_conn := f_start_handler(refers(f_TC_mo_setup_disc_late_rtp), pars);
|
|
vc_conn.done;
|
|
}
|
|
|
|
testcase TC_mt_with_sdp() runs on test_CT {
|
|
var ConnHdlrPars pars;
|
|
var ConnHdlr vc_conn;
|
|
f_init();
|
|
pars := valueof(t_Pars);
|
|
pars.g_cp := valueof(t_CallPars(is_mo := false, mncc_with_sdp := true));
|
|
vc_conn := f_start_handler(refers(f_TC_mt_success_rel_gsm), pars);
|
|
vc_conn.done;
|
|
}
|
|
|
|
testcase TC_mo_with_sdp() runs on test_CT {
|
|
var ConnHdlrPars pars;
|
|
var ConnHdlr vc_conn;
|
|
f_init();
|
|
pars := valueof(t_Pars);
|
|
pars.g_cp := valueof(t_CallPars(is_mo := true, mncc_with_sdp := true));
|
|
vc_conn := f_start_handler(refers(f_TC_mo_success_rel_sip), pars);
|
|
vc_conn.done;
|
|
}
|
|
|
|
control {
|
|
execute( TC_mt_success_rel_gsm() );
|
|
execute( TC_mt_success_rel_gsm_ipv6() );
|
|
execute( TC_mt_success_rel_sip() );
|
|
execute( TC_mo_success_rel_gsm() );
|
|
execute( TC_mo_success_rel_gsm_ipv6() );
|
|
execute( TC_mo_success_rel_sip() );
|
|
execute( TC_mo_setup_disc_late_rtp() );
|
|
execute( TC_mt_with_sdp() );
|
|
execute( TC_mo_with_sdp() );
|
|
}
|
|
|
|
|
|
|
|
}
|