add test for late media / 3pcc invite, which should now work

test/scenarios/uac-late-media.xml

@@ -19,22 +19,21 @@
 <!--                 Sipp 'uac' scenario with pcap (rtp) play           -->
 <!--                                                                    -->
 
-<scenario name="UAC with late media">
+<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:16173333456@[remote_ip]:[remote_port] SIP/2.0
+      INVITE sip:+16173333456@jambonz.org SIP/2.0
       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
-      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
-      To: sut <sip:[service]@[remote_ip]:[remote_port]>
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
+      To: <sip:[email protected]>
       Call-ID: [call_id]
       CSeq: 1 INVITE
       Contact: sip:sipp@[local_ip]:[local_port]
       Max-Forwards: 70
-      Subject: uac-no-3pcc
-      Content-Type: application/sdp
+      Subject: uac-late-media
       Content-Length: 0
 
     ]]>
@@ -43,27 +42,79 @@
   <recv response="100" optional="true">
   </recv>
 
-  <recv response="488">
+  <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:16173333456@[remote_ip]:[remote_port] SIP/2.0
-      [last_Via]
+      ACK sip:16173333456@jambonz.org SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
       From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
-      To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      To: <sip:[email protected]>[peer_tag_param]
       Call-ID: [call_id]
       CSeq: 1 ACK
-      Subject: uac-no-3pcc
+      Max-Forwards: 70
+      Subject: uac-late-media
+      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 8 101
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:101 telephone-event/8000
+      a=fmtp:101 0-11,16
+
+
+    ]]>
+  </send>
+
+  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
+  <nop>
+    <action>
+      <exec play_pcap_audio="pcap/g711a.pcap"/>
+    </action>
+  </nop>
+
+  <!-- Pause briefly      -->
+  <pause milliseconds="3000"/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:[email protected] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
+      To: <sip:[email protected]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Subject: uac-late-media
       Content-Length: 0
 
     ]]>
   </send>
 
-    <!-- definition of the response time repartition table (unit is ms)   -->
+  <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>
+
+</scenario>
+