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;
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);
--- /dev/null
+<?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>
+
projects / thirdparty / freeswitch.git / commitdiff
