pespin has submitted this change. (
https://gerrit.osmocom.org/c/osmo-ttcn3-hacks/+/37489?usp=email )
Change subject: asterisk: MO internal call: Expect direct rtp re-INVITE
......................................................................
asterisk: MO internal call: Expect direct rtp re-INVITE
Change-Id: I4ac602865cf0c18036b411acff4562322b32933b
---
M asterisk/Asterisk_Tests.ttcn
M asterisk/SIP_ConnectionHandler.ttcn
2 files changed, 70 insertions(+), 6 deletions(-)
Approvals:
pespin: Looks good to me, approved
Jenkins Builder: Verified
diff --git a/asterisk/Asterisk_Tests.ttcn b/asterisk/Asterisk_Tests.ttcn
index 9b5bb11..71d3576 100644
--- a/asterisk/Asterisk_Tests.ttcn
+++ b/asterisk/Asterisk_Tests.ttcn
@@ -213,7 +213,7 @@
COORD.send(COORD_CMD_REGISTERED);
COORD.receive(COORD_CMD_START);
- f_SIP_mo_call_setup();
+ f_SIP_mo_call_setup(exp_update_to_direct_rtp :=
g_pars.cp.exp_update_to_direct_rtp);
COORD.send(COORD_CMD_CALL_ESTABLISHED);
}
private function f_TC_internal_hangup_call_mo_unregister() runs on SIPConnHdlr
{
@@ -271,7 +271,7 @@
if (g_pars.cp.exp_update_to_direct_rtp) {
/* Once MO hangs up, Asterisk updates us to point RTP to it: */
- as_SIP_exp_call_update(g_pars.cp.sip_seq_nr + 1);
+ as_SIP_exp_mt_call_update(g_pars.cp.sip_seq_nr + 1);
}
as_SIP_exp_call_hangup(g_pars.cp.sip_seq_nr + 1);
@@ -716,6 +716,7 @@
private function f_ims_call_mo_configure(inout SIPConnHdlrPars sip_pars, inout
IMS_ConnHdlrPars ims_pars,
charstring tgt_ext_msisdn := "90829")
runs on test_CT {
+ sip_pars.cp.exp_update_to_direct_rtp := false;
sip_pars.cp.calling := sip_pars.registrar_sip_record;
sip_pars.cp.called :=
valueof(ts_SipAddr(ts_HostPort(sip_pars.remote_sip_host),
ts_UserInfo(tgt_ext_msisdn)));
diff --git a/asterisk/SIP_ConnectionHandler.ttcn
b/asterisk/SIP_ConnectionHandler.ttcn
index 49fa04c..c3d955a 100644
--- a/asterisk/SIP_ConnectionHandler.ttcn
+++ b/asterisk/SIP_ConnectionHandler.ttcn
@@ -467,7 +467,8 @@
}
function f_SIP_mo_call_setup(template (present) integer exp_status_code := 200,
- template (present) charstring exp_reason := "OK")
runs on SIPConnHdlr
+ template (present) charstring exp_reason := "OK",
+ boolean exp_update_to_direct_rtp := false) runs on
SIPConnHdlr
{
var template (value) PDU_SIP_Request req;
var template (present) PDU_SIP_Response exp;
@@ -545,9 +546,62 @@
g_pars.cp.sip_seq_nr,
omit);
SIP.send(req);
+
+ if (exp_update_to_direct_rtp) {
+ /* Asterisk will now update the session to connect us to MT
directly: */
+ /* Via is not kept since anyway "branch" will change upon
following INVITE. */
+ as_SIP_exp_mo_call_update(?);
+ }
+
g_pars.cp.sip_seq_nr := g_pars.cp.sip_seq_nr + 1;
}
+/* New (re-)INVITE arrives after MO call is established. Accept it: */
+altstep as_SIP_exp_mo_call_update(template (present) integer exp_seq_nr := ?,
boolean fail_others := true) runs on SIPConnHdlr
+{
+ var template (present) PDU_SIP_Request exp_req :=
+ tr_SIP_INVITE(f_tr_SipUrl_opt_defport(g_pars.local_sip_url_ext),
+ ?,
+
tr_From(tr_Addr_Union_from_val(g_pars.cp.called.addr), *),
+
tr_To(tr_Addr_Union_from_val(g_pars.cp.calling.addr), *),
+
tr_Via_from(f_tr_HostPort(g_pars.remote_sip_host, g_pars.remote_sip_port)),
+ exp_seq_nr,
+ body := ?);
+ var charstring sip_expect_str := log2str(exp_req);
+
+ [] SIP.receive(exp_req) -> value g_rx_sip_req {
+ var template (value) PDU_SIP_Response tx_resp;
+ var charstring tx_sdp;
+
+ f_SDP_decodeMessage(g_rx_sip_req.messageBody,
g_pars.cp.peer_sdp);
+ log("Rx Update MO INVITE decoded SDP: ", g_pars.cp.peer_sdp);
+
+ /* Tx 200 OK */
+ tx_sdp := f_gen_sdp();
+ tx_resp := ts_SIP_Response(g_rx_sip_req.msgHeader.callId.callid,
+ g_rx_sip_req.msgHeader.fromField,
+ g_rx_sip_req.msgHeader.toField,
+ "INVITE", 200,
+
g_rx_sip_req.msgHeader.cSeq.seqNumber,
+ "OK",
+ g_rx_sip_req.msgHeader.via,
+ body := tx_sdp);
+ SIP.send(tx_resp);
+
+ /* Wait for ACK */
+ exp_req :=
tr_SIP_ACK(f_tr_SipUrl_opt_defport(g_pars.local_sip_url_ext),
+ g_rx_sip_req.msgHeader.callId.callid,
+ g_rx_sip_req.msgHeader.fromField,
+ g_rx_sip_req.msgHeader.toField,
+
f_tr_Via_response(g_rx_sip_req.msgHeader.via),
+ g_rx_sip_req.msgHeader.cSeq.seqNumber,
+ *);
+ as_SIP_expect_req(exp_req);
+ }
+ [fail_others] as_SIP_fail_resp(sip_expect_str);
+ [fail_others] as_SIP_fail_req(sip_expect_str);
+}
+
private function f_ConnHdlr_parse_initial_SIP_INVITE(PDU_SIP_Request
rx_sip_req) runs on SIPConnHdlr
{
f_SDP_decodeMessage(rx_sip_req.messageBody, g_pars.cp.peer_sdp);
@@ -621,7 +675,7 @@
if (exp_update_to_direct_rtp) {
/* Asterisk will now update the session to connect us
to MO directly: */
/* Via is not kept since anyway "branch" will change
upon following INVITE. */
- as_SIP_exp_call_update(g_pars.cp.sip_seq_nr + 1);
+ as_SIP_exp_mt_call_update(g_pars.cp.sip_seq_nr + 1);
}
}
[fail_others] as_SIP_fail_resp(sip_expect_str);
@@ -694,7 +748,7 @@
}
/* New INVITE arrives after MT call is established. Accept it: */
-altstep as_SIP_exp_call_update(template (present) integer exp_seq_nr := ?,
boolean fail_others := true) runs on SIPConnHdlr
+altstep as_SIP_exp_mt_call_update(template (present) integer exp_seq_nr := ?,
boolean fail_others := true) runs on SIPConnHdlr
{
var template (present) PDU_SIP_Request exp_req :=
tr_SIP_INVITE(f_tr_SipUrl_opt_defport(g_pars.local_sip_url_ext),
@@ -760,7 +814,7 @@
via := g_pars.local_via;
via.viaBody[0].viaParams := f_sip_param_set(via.viaBody[0].viaParams,
"branch", branch_value);
- /* Transmit ACK */
+ /* Transmit BYE */
req := ts_SIP_BYE(g_pars.cp.sip_call_id,
g_pars.cp.from_addr,
g_pars.cp.to_addr,
--
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Gerrit-Project: osmo-ttcn3-hacks
Gerrit-Branch: master
Gerrit-Change-Id: I4ac602865cf0c18036b411acff4562322b32933b
Gerrit-Change-Number: 37489
Gerrit-PatchSet: 2
Gerrit-Owner: pespin <[email protected]>
Gerrit-Reviewer: Jenkins Builder
Gerrit-Reviewer: pespin <[email protected]>
Gerrit-MessageType: merged
