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 };
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);
}
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;
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 */
}
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;
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 */
}
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;
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) {
--- /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: 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>
+
projects / thirdparty / freeswitch.git / commitdiff
