mirror of
https://gitlab.linphone.org/BC/public/linphone-iphone.git
synced 2026-02-01 02:39:22 +00:00
Added test with 200OK without contact header
This commit is contained in:
Sylvain Berfini
parent
df0e8ea550
commit
5339fed4dc
2 changed files with 173 additions and 0 deletions
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@ -70,6 +70,24 @@ static FILE *sip_start(const char *senario, const char* dest_username, LinphoneA
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return NULL;
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#endif
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}
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static FILE *sip_start_recv(const char *senario) {
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#if HAVE_SIPP
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char *command;
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FILE *file;
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//until errors logs are handled correctly and stop breaks output, they will be DISABLED
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command = ms_strdup_printf(SIPP_COMMAND" -sf %s -trace_err -trace_msg -rtp_echo -m 1 -d 1000",senario);
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ms_message("Starting sipp commad [%s]",command);
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file = popen(command, "r");
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ms_free(command);
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return file;
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#else
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return NULL;
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#endif
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}
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/*static void dest_server_server_resolved(void *data, const char *name, struct addrinfo *ai_list) {
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*(struct addrinfo **)data =ai_list;
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}*/
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@ -286,12 +304,49 @@ static void call_with_multiple_video_mline_in_sdp() {
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linphone_core_manager_destroy(mgr);
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}
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static void call_invite_200ok_without_contact_header() {
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LinphoneCoreManager *mgr;
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char *identity_char;
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char *scen;
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FILE * sipp_out;
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LinphoneCall *call = NULL;
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/*currently we use direct connection because sipp do not properly set ACK request uri*/
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mgr= linphone_core_manager_new2("empty_rc", FALSE);
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mgr->identity = linphone_core_get_primary_contact_parsed(mgr->lc);
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linphone_address_set_username(mgr->identity,"marie");
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identity_char = linphone_address_as_string(mgr->identity);
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linphone_core_set_primary_contact(mgr->lc,identity_char);
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linphone_core_iterate(mgr->lc);
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scen = bc_tester_res("sipp/call_invite_200ok_without_contact_header.xml");
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sipp_out = sip_start_recv(scen);
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if (sipp_out) {
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call = linphone_core_invite(mgr->lc, "sipp@127.0.0.1");
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BC_ASSERT_PTR_NOT_NULL(call);
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BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallOutgoingInit, 1));
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BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallOutgoingProgress, 1));
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BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallOutgoingRinging, 1));
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if (call) {
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BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallStreamsRunning, 1));
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check_rtcp(call);
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linphone_core_terminate_call(mgr->lc, call);
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}
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pclose(sipp_out);
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}
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linphone_core_manager_destroy(mgr);
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}
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static test_t tests[] = {
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{ "SIP UPDATE within incoming reinvite without sdp", sip_update_within_icoming_reinvite_with_no_sdp },
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{ "Call with audio mline before video in sdp", call_with_audio_mline_before_video_in_sdp },
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{ "Call with video mline before audio in sdp", call_with_video_mline_before_audio_in_sdp },
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{ "Call with multiple audio mline in sdp", call_with_multiple_audio_mline_in_sdp },
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{ "Call with multiple video mline in sdp", call_with_multiple_video_mline_in_sdp },
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{ "Call invite 200ok without contact header", call_invite_200ok_without_contact_header },
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};
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test_suite_t complex_sip_call_test_suite = {
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118
tester/sipp/call_invite_200ok_without_contact_header.xml
Normal file
118
tester/sipp/call_invite_200ok_without_contact_header.xml
Normal file
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@ -0,0 +1,118 @@
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<?xml version="1.0" encoding="ISO-8859-1" ?>
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<!DOCTYPE scenario SYSTEM "sipp.dtd">
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<!-- This program is free software; you can redistribute it and/or -->
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<!-- modify it under the terms of the GNU General Public License as -->
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<!-- published by the Free Software Foundation; either version 2 of the -->
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<!-- License, or (at your option) any later version. -->
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<!-- -->
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<!-- This program is distributed in the hope that it will be useful, -->
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<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
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<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
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<!-- GNU General Public License for more details. -->
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<!-- -->
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<!-- You should have received a copy of the GNU General Public License -->
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<!-- along with this program; if not, write to the -->
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<!-- Free Software Foundation, Inc., -->
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<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
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<!-- -->
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<!-- Sipp default 'uas' scenario. -->
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<!-- -->
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<scenario name="Basic UAS responder">
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<!-- By adding rrs="true" (Record Route Sets), the route sets -->
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<!-- are saved and used for following messages sent. Useful to test -->
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<!-- against stateful SIP proxies/B2BUAs. -->
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<recv request="INVITE" crlf="true">
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</recv>
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<!-- The '[last_*]' keyword is replaced automatically by the -->
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<!-- specified header if it was present in the last message received -->
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<!-- (except if it was a retransmission). If the header was not -->
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<!-- present or if no message has been received, the '[last_*]' -->
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<!-- keyword is discarded, and all bytes until the end of the line -->
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<!-- are also discarded. -->
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<!-- -->
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<!-- If the specified header was present several times in the -->
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<!-- message, all occurences are concatenated (CRLF seperated) -->
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<!-- to be used in place of the '[last_*]' keyword. -->
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<send>
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<![CDATA[
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SIP/2.0 180 Ringing
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[last_Via:]
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[last_From:]
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[last_To:];tag=[call_number]
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[last_Call-ID:]
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[last_CSeq:]
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Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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Content-Length: 0
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]]>
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</send>
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<send retrans="500">
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<![CDATA[
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SIP/2.0 200 OK
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[last_Via:]
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[last_From:]
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[last_To:];tag=[call_number]
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[last_Call-ID:]
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[last_CSeq:]
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Content-Type: application/sdp
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Content-Length: [len]
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v=0
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o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
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s=-
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c=IN IP[media_ip_type] [media_ip]
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t=0 0
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m=audio [media_port] RTP/AVP 96 97 0 8 101 98
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a=rtpmap:96 speex/16000
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a=fmtp:96 vbr=on
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a=rtpmap:97 speex/8000
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a=fmtp:97 vbr=on
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a=rtpmap:101 telephone-event/16000
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a=rtpmap:98 telephone-event/8000
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]]>
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</send>
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<recv request="ACK"
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optional="true"
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rtd="true"
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crlf="true">
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</recv>
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<recv request="BYE">
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</recv>
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<send>
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<![CDATA[
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SIP/2.0 200 OK
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[last_Via:]
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[last_From:]
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[last_To:]
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[last_Call-ID:]
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[last_CSeq:]
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Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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Content-Length: 0
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]]>
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</send>
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<!-- Keep the call open for a while in case the 200 is lost to be -->
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<!-- able to retransmit it if we receive the BYE again. -->
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<pause milliseconds="1000"/>
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<!-- definition of the response time repartition table (unit is ms) -->
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<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
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<!-- definition of the call length repartition table (unit is ms) -->
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<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
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</scenario>
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