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[mod_sofia] Add a unit-test for the 3pcc telephone event.

This commit is contained in:
dhruvecosmob 2021-08-27 13:44:49 +03:00 committed by Andrey Volk
parent 3e93f2423d
commit 8308233819
3 changed files with 281 additions and 9 deletions
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7
src/mod/endpoints/mod_sofia/test/conf-sipp/freeswitch.xml
View File

@ -584,6 +584,13 @@
</condition>
</extension>
<extension name="sipp_telephone_check">
<condition field="destination_number" expression="^1212121212$">
<action application="answer"/>
<action application="set" data="park_after_bridge=true"/>
<action application="park"/>
</condition>
</extension>
<extension name="sipp">
<condition>

124
src/mod/endpoints/mod_sofia/test/sipp-based-tests.c
View File

@ -37,6 +37,23 @@
int test_success = 0;
int test_sofia_debug = 1;
static const char *test_wait_for_chan_var(switch_channel_t *channel, const char *seq)
{
int loop_count = 50;
const char *var=NULL;
do {
if (!strcmp(switch_channel_get_variable(channel, "sip_cseq"),seq)){
switch_sleep(100 * 1000);
var = switch_channel_get_variable(channel, "rtp_local_sdp_str");
break;
}
switch_sleep(100 * 1000);
} while(loop_count--);
return var;
}
static switch_bool_t has_ipv6()
{
switch_stream_handle_t stream = { 0 };
@ -73,9 +90,9 @@ static void unregister_gw()
switch_safe_free(stream.data);
}
static int start_sipp_uac(const char *ip, int remote_port,const char *scenario_uac, const char *extra)
static int start_sipp_uac(const char *ip, int remote_port, const char *dialed_number, const char *scenario_uac, const char *extra)
{
char *cmd = switch_mprintf("sipp %s:%d -nr -p 5062 -m 1 -s 1001 -recv_timeout 10000 -timeout 10s -sf %s -bg %s", ip, remote_port, scenario_uac, extra);
char *cmd = switch_mprintf("sipp %s:%d -nr -p 5062 -m 1 -s %s -recv_timeout 10000 -timeout 10s -sf %s -bg %s", ip, remote_port, dialed_number, scenario_uac, extra);
int sys_ret = switch_system(cmd, SWITCH_TRUE);
printf("%s\n", cmd);
@ -199,9 +216,98 @@ FST_CORE_EX_BEGIN("./conf-sipp", SCF_VG | SCF_USE_SQL)
}
FST_TEARDOWN_END()
FST_TEST_BEGIN(uac_telephone_event_check)
{
const char *local_ip_v4 = switch_core_get_variable("local_ip_v4");
char *channel_data = NULL;
char uuid[100] = "";
int sipp_ret;
int sdp_count = 0 , loop_count =50;
switch_stream_handle_t stream = { 0 };
sipp_ret = start_sipp_uac(local_ip_v4, 5080, "1212121212", "sipp-scenarios/uac_telephone_event.xml", "");
if (sipp_ret < 0 || sipp_ret == 127) {
fst_requires(0); /* sipp not found */
}
do {
SWITCH_STANDARD_STREAM(stream);
switch_api_execute("show", "channels", NULL, &stream);
if (!strncmp((char *)stream.data, "uuid,", 5)) {
channel_data = switch_mprintf("%s", (char *)stream.data);
switch_safe_free(stream.data);
break;
}
switch_safe_free(stream.data);
switch_sleep(100 * 1000);
} while (loop_count--);
if (channel_data) {
char *temp = NULL;
int i;
if ((temp = strchr(channel_data, '\n'))) {
temp++;
for (i = 0; temp[i] != ',' && i < 99; i++){
uuid[i] = temp[i];
}
uuid[i] = '\0';
}
if (!zstr(uuid)) {
switch_core_session_t *session = switch_core_session_locate(uuid);
switch_channel_t *channel;
const char *sdp_str1 = NULL, *sdp_str2 = NULL;
fst_requires(session);
channel = switch_core_session_get_channel(session);
sdp_str1 = test_wait_for_chan_var(channel,"1");
sdp_str2 = test_wait_for_chan_var(channel,"2");
if (sdp_str1 && sdp_str2 && (strstr(sdp_str1,"telephone-event")) && (strstr(sdp_str2,"telephone-event"))){
temp = NULL;
sdp_count = 1;
if ((temp = strstr(sdp_str2,"RTP/AVP"))) {
int count = 0;
for (i = 7; temp[i] != '\n' && i < 99; i++) {
/* checking for payload-type 101.*/
if(temp[i++] == '1' && temp[i++] == '0' && temp[i++] == '1') {
count++;
}
}
if (count > 1) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Duplicate entry of payload in SDP.\n");
sdp_count = 0;
}
}
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Telephone-event missing in SDP.\n");
}
switch_core_session_rwunlock(session);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Uuid not found in Channel Data.\n");
}
free(channel_data);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Unable to find Channel Data.\n");
}
fst_check(sdp_count == 1);
/* sipp should timeout, attempt kill, just in case.*/
kill_sipp();
}
FST_TEST_END()
FST_TEST_BEGIN(uac_digest_leak_udp)
{
switch_core_session_t *session;
switch_core_session_t *session;
switch_call_cause_t cause;
switch_status_t status;
switch_channel_t *channel;
@ -211,7 +317,7 @@ FST_CORE_EX_BEGIN("./conf-sipp", SCF_VG | SCF_USE_SQL)
switch_event_bind("sofia", SWITCH_EVENT_CUSTOM, NULL, event_handler, NULL);
status = switch_ivr_originate(NULL, &session, &cause, "loopback/+15553334444", 2, NULL, NULL, NULL, NULL, NULL, SOF_NONE, NULL, NULL);
sipp_ret = start_sipp_uac(local_ip_v4, 5080, "sipp-scenarios/uac_digest_leak.xml", "");
sipp_ret = start_sipp_uac(local_ip_v4, 5080, "1001", "sipp-scenarios/uac_digest_leak.xml", "");
if (sipp_ret < 0 || sipp_ret == 127) {
fst_requires(0); /* sipp not found */
}
@ -250,7 +356,7 @@ FST_CORE_EX_BEGIN("./conf-sipp", SCF_VG | SCF_USE_SQL)
FST_TEST_BEGIN(uac_digest_leak_tcp)
{
switch_core_session_t *session;
switch_core_session_t *session;
switch_call_cause_t cause;
switch_status_t status;
switch_channel_t *channel;
@ -260,7 +366,7 @@ FST_CORE_EX_BEGIN("./conf-sipp", SCF_VG | SCF_USE_SQL)
switch_event_bind("sofia", SWITCH_EVENT_CUSTOM, NULL, event_handler, NULL);
status = switch_ivr_originate(NULL, &session, &cause, "loopback/+15553334444", 2, NULL, NULL, NULL, NULL, NULL, SOF_NONE, NULL, NULL);
sipp_ret = start_sipp_uac(local_ip_v4, 5080, "sipp-scenarios/uac_digest_leak-tcp.xml", "-t t1");
sipp_ret = start_sipp_uac(local_ip_v4, 5080, "1001", "sipp-scenarios/uac_digest_leak-tcp.xml", "-t t1");
if (sipp_ret < 0 || sipp_ret == 127) {
fst_requires(0); /* sipp not found */
}
@ -299,7 +405,7 @@ FST_CORE_EX_BEGIN("./conf-sipp", SCF_VG | SCF_USE_SQL)
FST_TEST_BEGIN(uac_digest_leak_udp_ipv6)
{
switch_core_session_t *session;
switch_core_session_t *session;
switch_call_cause_t cause;
switch_status_t status;
switch_channel_t *channel;
@ -318,9 +424,9 @@ FST_CORE_EX_BEGIN("./conf-sipp", SCF_VG | SCF_USE_SQL)
status = switch_ivr_originate(NULL, &session, &cause, "loopback/+15553334444", 2, NULL, NULL, NULL, NULL, NULL, SOF_NONE, NULL, NULL);
if (!ipv6) {
sipp_ret = start_sipp_uac(local_ip_v6, 6060, "sipp-scenarios/uac_digest_leak-ipv6.xml", "-i [::1]");
sipp_ret = start_sipp_uac(local_ip_v6, 6060, "1001", "sipp-scenarios/uac_digest_leak-ipv6.xml", "-i [::1]");
} else {
sipp_ret = start_sipp_uac(ipv6, 6060, "sipp-scenarios/uac_digest_leak-ipv6.xml", "-i [::1] -mi [::1]");
sipp_ret = start_sipp_uac(ipv6, 6060, "1001", "sipp-scenarios/uac_digest_leak-ipv6.xml", "-i [::1] -mi [::1]");
}
if (sipp_ret < 0 || sipp_ret == 127) {

159
src/mod/endpoints/mod_sofia/test/sipp-scenarios/uac_telephone_event.xml Normal file
View File

@ -0,0 +1,159 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: t_sipp <sip:t_sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:t_sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
a=sendrecv
a=ptime:20
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true" crlf="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: t_sipp <sip:t_sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:t_sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<pause milliseconds="1500"/>
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: t_sipp <sip:t_sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: sip:t_sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: [len]
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="200" rtd="true" crlf="true">
<!--<action>
<ereg regexp="m=audio.*[0-9][1-5].*101.*"
<ereg regexp="101 telephone-event"
search_in="body"
check_it="true"
assign_to="5"/>
</action>-->
</recv>
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: t_sipp <sip:t_sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:t_sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
]]>
</send>
<send retrans="500">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: t_sipp <sip:t_sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 BYE
Contact: sip:t_sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>