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[Core] Add Unit test script for Check RTP/SAVP transport protocol wit…
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…h crypto attribute.
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@dhruvecosmob @andywolk
dhruvecosmob authored and andywolk committed Oct 23, 2021
1 parent 8a2ad23 commit 047c3c7
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153 changes: 104 additions & 49 deletions src/mod/endpoints/mod_sofia/test/sipp-based-tests.c
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Original file line number Diff line number Diff line change
Expand Up @@ -37,6 +37,39 @@
int test_success = 0;
int test_sofia_debug = 1;

static void test_wait_for_uuid(char *uuid)
{
switch_stream_handle_t stream = { 0 };
int loop_count = 50;
char *channel_data=NULL;

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];
}
}
free(channel_data);
}
}

static const char *test_wait_for_chan_var(switch_channel_t *channel, const char *seq)
{
int loop_count = 50;
Expand Down Expand Up @@ -219,84 +252,106 @@ FST_CORE_EX_BEGIN("./conf-sipp", SCF_VG | SCF_USE_SQL)
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 };
int sdp_count = 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;
}
test_wait_for_uuid(uuid);
if (!zstr(uuid)) {
const char *sdp_str1 = NULL, *sdp_str2 = NULL;
switch_core_session_t *session = switch_core_session_locate(uuid);
switch_channel_t *channel = switch_core_session_get_channel(session);
fst_requires(channel);

sdp_str1 = test_wait_for_chan_var(channel,"1");
sdp_str2 = test_wait_for_chan_var(channel,"2");

switch_safe_free(stream.data);
switch_sleep(100 * 1000);
} while (loop_count--);
if (sdp_str1 && sdp_str2 && (strstr(sdp_str1,"telephone-event")) && (strstr(sdp_str2,"telephone-event"))){
char *temp = NULL;
sdp_count = 1;

if (channel_data) {
char *temp = NULL;
int i;
if ((temp = strstr(sdp_str2,"RTP/AVP"))) {
int count = 0, i;

if ((temp = strchr(channel_data, '\n'))) {
temp++;
for (i = 0; temp[i] != ',' && i < 99; i++){
uuid[i] = temp[i];
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;
}
}
uuid[i] = '\0';

} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Telephone-event missing in SDP.\n");
}
switch_core_session_rwunlock(session);

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;
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Uuid not found in Channel Data.\n");
}

fst_requires(session);
channel = switch_core_session_get_channel(session);
fst_check(sdp_count == 1);
/* sipp should timeout, attempt kill, just in case.*/
kill_sipp();
}
FST_TEST_END()

sdp_str1 = test_wait_for_chan_var(channel,"1");
sdp_str2 = test_wait_for_chan_var(channel,"2");
FST_TEST_BEGIN(uac_savp_check)
{
const char *local_ip_v4 = switch_core_get_variable("local_ip_v4");
char uuid[100] = "";
int sipp_ret;
int sdp_count = 0;

sipp_ret = start_sipp_uac(local_ip_v4, 5080, "1212121212", "sipp-scenarios/uac_savp_check.xml", "");
if (sipp_ret < 0 || sipp_ret == 127) {
fst_requires(0); /* sipp not found */
}

if (sdp_str1 && sdp_str2 && (strstr(sdp_str1,"telephone-event")) && (strstr(sdp_str2,"telephone-event"))){
temp = NULL;
sdp_count = 1;
test_wait_for_uuid(uuid);
if (!zstr(uuid)) {
const char *sdp_str1 = NULL, *sdp_str2 = NULL;
const char *temp = NULL, *temp1 = NULL;
switch_core_session_t *session = switch_core_session_locate(uuid);
switch_channel_t *channel = switch_core_session_get_channel(session);
fst_requires(channel);

if ((temp = strstr(sdp_str2,"RTP/AVP"))) {
int count = 0;
sdp_str1 = test_wait_for_chan_var(channel,"1");
sdp_str2 = test_wait_for_chan_var(channel,"2");

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 (sdp_str1 && sdp_str2 && (temp = strstr(sdp_str2,"RTP/SAVP")) && (temp1 = strstr(temp,"crypto"))) {
int i = 0;

if (count > 1) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Duplicate entry of payload in SDP.\n");
sdp_count = 1;
for (i = 0; temp1[i]; i++) {

if ((temp = strstr(temp1,"RTP/SAVP"))) {
if ((temp1 = strstr(temp,"crypto"))) {
i = 0;
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Fail due to no crypto found with SAVP.\n");
sdp_count = 0;
break;
}
}
} 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");
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "SAVP not found in SDP.\n");
}
switch_core_session_rwunlock(session);

free(channel_data);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Unable to find Channel Data.\n");
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Uuid not found in Channel Data.\n");
}

fst_check(sdp_count == 1);
Expand Down
148 changes: 148 additions & 0 deletions src/mod/endpoints/mod_sofia/test/sipp-scenarios/uac_savp_check.xml
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Original file line number Diff line number Diff line change
@@ -0,0 +1,148 @@
<?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: s_sipp <sip:s_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:s_sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=CiscoSystemsCCM-SIP 1195507 1 IN IP[local_ip_type] [local_ip]
s=SIP Call
c=IN IP[local_ip_type] [local_ip]
b=TIAS:64000
b=AS:80
t=0 0
m=audio [auto_media_port] RTP/SAVP 18 0 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mSPPjYxzAEWkICVXidkYXFdsHr/J2NhpkqQepffH
]]>
</send>


<recv response="100" optional="true">
</recv>

<recv response="180" optional="true">
</recv>
<recv response="183" 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: s_sipp <sip:s_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:s_sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>

<pause milliseconds="1000"/>

<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: s_sipp <sip:s_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:s_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">
</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: s_sipp <sip:s_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:s_sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</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: s_sipp <sip:s_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:s_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>

1 comment on commit 047c3c7

@andywolk
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https://github.com/signalwire/stack/pull/470

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