neels has uploaded this change for review. (
https://gerrit.osmocom.org/c/osmo-ttcn3-hacks/+/34417?usp=email )
Change subject: sip: test SDP forwarding via MNCC
......................................................................
sip: test SDP forwarding via MNCC
Add CallPars.mncc_with_sdp: when true, the call establishing functions
f_establish_{mo,mt} now send valid SDP via MNCC, and validate that the
SDP received on MNCC and SIP are as expected.
Keep all current tests unchanged with mncc_with_sdp := false: they will
continue to test the case without SDP (for legacy compatibility). These
tests will still pass on the 'latest' builds.
Add two new tests for mncc_with_sdp := true: TC_mt_with_sdp and
TC_mo_with_sdp. These new tests will fail on our 'latest' builds until
the SDP forwarding feature in osmo-sip-connector is released.
Related: osmo-sip-connector I3df5d06f38ee2d122706a9ebffde7db4f2bd6bae
Change-Id: Ib2ae8449e673f5027f01d428d3718c006f76d93e
---
M sip/SIP_Tests.ttcn
1 file changed, 210 insertions(+), 16 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-ttcn3-hacks
refs/changes/17/34417/1
diff --git a/sip/SIP_Tests.ttcn b/sip/SIP_Tests.ttcn
index fe6600b..d2e1785 100644
--- a/sip/SIP_Tests.ttcn
+++ b/sip/SIP_Tests.ttcn
@@ -66,7 +66,12 @@
charstring sip_rtp_addr,
uint16_t sip_rtp_port,
charstring cn_rtp_addr,
- uint16_t cn_rtp_port
+ 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 {
@@ -77,7 +82,7 @@
integer sip_seq_nr
}
-private template (value) CallPars t_CallPars(boolean is_mo) := {
+private template (value) CallPars t_CallPars(boolean is_mo, boolean
mncc_with_sdp := false) := {
is_mo := is_mo,
calling := "12345",
called := "98766",
@@ -87,7 +92,8 @@
sip_rtp_addr := "1.2.3.4",
sip_rtp_port := 1234,
cn_rtp_addr := "5.6.7.8",
- cn_rtp_port := 5678
+ cn_rtp_port := 5678,
+ mncc_with_sdp := mncc_with_sdp
}
private function f_CallPars_compute(inout CallPars cp) {
@@ -212,18 +218,76 @@
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);
+ setverdict(fail, "unexpected SDP");
+ mtc.stop;
+ }
+}
+
/* 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
+ * |--------SDP2--> 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, *,
"INVITE", 100, ?, "Trying", *));
@@ -231,7 +295,9 @@
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;
+ 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");
@@ -244,26 +310,49 @@
/* 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));
+ 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_mgcp_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 on SIP */
+ var template charstring expect_sdp_to_sip := *;
+
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 */
+ expect_sdp_to_sip := 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;
as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext,
sip_addr_gsm, *,
- "INVITE", 180, ?, "Ringing", *));
+ "INVITE", 180, ?, "Ringing",
expect_sdp_to_sip);
/* 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));
+ 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, contact_addr := ?,
@@ -275,7 +364,9 @@
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));
+ 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' */
@@ -286,15 +377,53 @@
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
+ * |<--SDP2----- 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_mgcp_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.send(ts_MNCC_SETUP_ind(cp.mncc_call_id, dst, src,
"262420123456789"));
+ 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)) {
- var MNCC_PDU mncc :=
valueof(ts_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;
@@ -302,7 +431,13 @@
}
/* OSC -> SIP: Send INVITE with GSM side IP/Port in SDP */
- sip_req := f_SIP_expect_req(tr_SIP_INVITE(?, sip_addr_gsm,
sip_addr_ext, ?, ?));
+ 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;
@@ -311,22 +446,37 @@
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));
+ 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 */
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));
+ "INVITE", 180, seq_nr, "Ringing",
sip_req.msgHeader.via, cp.comp.sip_body));
/* MSC <- OSC: "180 Ringing" translated to MNCC_ALERT_REQ */
- MNCC.receive(tr_MNCC_ALERT_req(cp.mncc_call_id)) {}
+ 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));
- MNCC.receive(tr_MNCC_RTP_CONNECT(cp.mncc_call_id));
+
+ 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));
+ 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));
@@ -555,6 +705,26 @@
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() );
@@ -563,6 +733,8 @@
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() );
}
--
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Gerrit-Project: osmo-ttcn3-hacks
Gerrit-Branch: master
Gerrit-Change-Id: Ib2ae8449e673f5027f01d428d3718c006f76d93e
Gerrit-Change-Number: 34417
Gerrit-PatchSet: 1
Gerrit-Owner: neels <nhofm...@sysmocom.de>
Gerrit-MessageType: newchange
