TT#140851 scenarios extended for P-Early-Media - invite_allowip_soundset

invite_allowip_soundset has also been extended to demonstrate kamailio proxy handling P-Early-Media SIP Header. - The first call is the same and demonstrates if the caller does not send P-Early-Media header, ngcp will assume he's supporting early media and he's gonna play the announce. - The Second call is the same case but since P-Early-Media is "inactive" kamailio wont sent the audio message and justs sends an error. Change-Id: Ib989809ada070101b932deaf9ba8d2f5c1a92fe0
4 years ago · 169be79061
parent 0108575e95
commit 169be79061
4 changed files with 241 additions and 0 deletions
--- a/scenarios/invite_allowip_soundset/0009_test.yml.tt2
+++ b/scenarios/invite_allowip_soundset/0009_test.yml.tt2
@ -0,0 +1,108 @@
+flow:
+  - start|DEFAULT_ROUTE:
+  - start|ROUTE_ENTRY:
+  - return|ROUTE_ENTRY:
+  - start|ROUTE_NET_INFO:
+  - return|ROUTE_NET_INFO:
+  - start|ROUTE_CHECK_TEL_URI:
+  - return|ROUTE_CHECK_TEL_URI:
+  - start|ROUTE_PRX_REQUEST:
+  - start|ROUTE_INITVARS:
+  - return|ROUTE_INITVARS:
+  - start|ROUTE_INVITE:
+  - start|ROUTE_LOAD_CALLEE_DOMAIN_PREF:
+  - start|ROUTE_CLEAR_CALLEE_DOMAIN_PREF:
+  - return|ROUTE_CLEAR_CALLEE_DOMAIN_PREF:
+  - return|ROUTE_LOAD_CALLEE_DOMAIN_PREF:
+  - start|ROUTE_FIND_CALLER:
+  - start|ROUTE_AUTH:
+  - start|ROUTE_AUTH_HELPER:
+  - return|ROUTE_AUTH_HELPER:
+  - start|ROUTE_ADD_CALLINFO_REPLY:
+  - start|ROUTE_ADD_CALLINFO_CALLER_PRIMARY:
+  - return|ROUTE_ADD_CALLINFO_CALLER_PRIMARY:
+  - return|ROUTE_ADD_CALLINFO_REPLY:
+  - return|ROUTE_AUTH:
+  - return|ROUTE_FIND_CALLER:
+  - start|ROUTE_LOAD_CALLER_PREF:
+  - start|ROUTE_CLEAR_CALLER_PREF:
+  - return|ROUTE_CLEAR_CALLER_PREF:
+  - start|ROUTE_CHECK_UA:
+  - return|ROUTE_CHECK_UA:
+  - start|ROUTE_LOAD_CALLER_CONTRACT_PREF:
+  - return|ROUTE_LOAD_CALLER_CONTRACT_PREF:
+  - return|ROUTE_LOAD_CALLER_PREF:
+  - start|ROUTE_DLG_MANAGE:
+  - return|ROUTE_DLG_MANAGE:
+  - start|ROUTE_PARSE_P_EARLY_MEDIA:
+  - start|ROUTE_DISABLE_SEMS_EARLY_MEDIA:
+  - return|ROUTE_DISABLE_SEMS_EARLY_MEDIA:
+  - return|ROUTE_PARSE_P_EARLY_MEDIA:
+      $var(pem_flags): 1
+      $var(sems_early_media_disabled): 1
+  - start|ROUTE_APPLY_HEADER_RULES:
+  - return|ROUTE_APPLY_HEADER_RULES:
+  - start|ROUTE_GET_CALLER_CLI:
+  - return|ROUTE_GET_CALLER_CLI:
+  - start|ROUTE_GET_FORWARDER_CLI:
+  - return|ROUTE_GET_FORWARDER_CLI:
+  - start|ROUTE_APPLY_REWRITE_RULE:
+  - return|ROUTE_APPLY_REWRITE_RULE:
+  - start|ROUTE_ACC_FAILURE:
+  - start|ROUTE_ACC_CALLER:
+  - start|ROUTE_SET_ACC_CALLER_CALL_COUNTERS:
+  - return|ROUTE_SET_ACC_CALLER_CALL_COUNTERS:
+  - return|ROUTE_ACC_CALLER:
+  - start|ROUTE_ACC_CALLEE:
+  - start|ROUTE_SET_ACC_CALLEE_CALL_COUNTERS:
+  - return|ROUTE_SET_ACC_CALLEE_CALL_COUNTERS:
+  - return|ROUTE_ACC_CALLEE:
+  - return|ROUTE_ACC_FAILURE:
+  - start|ROUTE_EARLY_REJECT:
+  - start|ROUTE_ADD_CALLINFO_REPLY:
+  - start|ROUTE_ADD_CALLINFO_CALLER_PRIMARY:
+  - return|ROUTE_ADD_CALLINFO_CALLER_PRIMARY:
+  - start|ROUTE_ADD_CALLINFO_CALLEE_PRIMARY:
+  - return|ROUTE_ADD_CALLINFO_CALLEE_PRIMARY:
+  - return|ROUTE_ADD_CALLINFO_REPLY:
+  - start|ROUTE_LOCAL_REPLY:
+  - start|ROUTE_APPLY_HEADER_RULES:
+  - return|ROUTE_APPLY_HEADER_RULES:
+  - start|dialog:failed:
+  - return|dialog:failed:
+  - return|dialog:failed:
+  - start|ROUTE_EXIT:
+  - start|ROUTE_RUNTIME:
+  - return|ROUTE_RUNTIME:
+  - exit|ROUTE_EXIT:
+sip_in:
+  - 'INVITE sip:[% scenarios.1.responders.0.username %]@invite-allowip-soundset.scenarios.test SIP/2.0'
+  - 'From: <sip:[% scenarios.1.username %]@invite-allowip-soundset.scenarios.test>;tag=[\w-]+'
+  - 'To: <sip:[% scenarios.1.responders.0.username %]@invite-allowip-soundset.scenarios.test>'
+  - 'CSeq: \d+ INVITE'
+  - 'Proxy-Authorization: Digest username="testuser1002",realm="invite-allowip-soundset.scenarios.test",uri="sip:[% server_ip %]:5060",nonce=".+",response=".+",algorithm=MD5'
+  - 'P-Early-Media: supported,inactive'
+  - 'Content-Type: application/sdp'
+  - 'Content-Length:\s+\d+'
+sip_out:
+  - [
+      'SIP/2.0 100 Trying',
+      'From: <sip:[% scenarios.1.username %]@invite-allowip-soundset.scenarios.test>;tag=[\w-]+',
+      'To: <sip:[% scenarios.1.responders.0.username %]@invite-allowip-soundset.scenarios.test>',
+      'CSeq: \d+ INVITE',
+      'Content-Length: 0',
+    ]
+  - [
+      'SIP/2.0 101 Connecting',
+      'From: <sip:testuser1002@invite-allowip-soundset.scenarios.test>;tag=[\w-]+',
+      'To: <sip:[% scenarios.1.responders.0.username %]@invite-allowip-soundset.scenarios.test>',
+      'CSeq: \d+ INVITE',
+      'Content-Length: 0',
+    ]
+  - [
+      'SIP/2.0 403 Unauthorized IP detected',
+      'From: <sip:testuser1002@invite-allowip-soundset.scenarios.test>;tag=[\w-]+',
+      'To: <sip:[% scenarios.1.responders.0.username %]@invite-allowip-soundset.scenarios.test>;tag=[\w-]+',
+      'CSeq: \d+ INVITE',
+      'Content-Length: 0',
+    ]
--- a/scenarios/invite_allowip_soundset/cdr_test.yml.tt2
+++ b/scenarios/invite_allowip_soundset/cdr_test.yml.tt2
@ -7,4 +7,13 @@ cdr:
    destination_user_dialed: 'testuser1003'
    call_type: 'call'
    call_status: 'other'
+    call_code: '403'
+  - source_user: 'testuser1002'
+    source_domain: 'invite-allowip-soundset.scenarios.test'
+    source_cli: '0'
+    destination_user: 'testuser1004'
+    destination_domain: 'invite-allowip-soundset.scenarios.test'
+    destination_user_dialed: 'testuser1004'
+    call_type: 'call'
+    call_status: 'other'
    call_code: '403'
--- a/scenarios/invite_allowip_soundset/scenario.yml
+++ b/scenarios/invite_allowip_soundset/scenario.yml
@ -21,6 +21,12 @@ subscribers:
      cc: 43
      ac: 1
      sn: 1003
+    testuser1004:
+      customer: 'customer.test'
+      password: testuser
+      cc: 43
+      ac: 1
+      sn: 1003
    testuser1002:
      customer: 'customer.test'
      password: testuser
@ -37,3 +43,12 @@ scenarios:
        domain: invite-allowip-soundset.scenarios.test
        register: 'no'
        active: 'no'
+  - ip: 127.126.0.1
+    username: testuser1002
+    domain: invite-allowip-soundset.scenarios.test
+    responders:
+      - ip: 127.1.0.1
+        username: testuser1004
+        domain: invite-allowip-soundset.scenarios.test
+        register: 'no'
+        active: 'no'
--- a/scenarios/invite_allowip_soundset/sipp_scenario01.xml
+++ b/scenarios/invite_allowip_soundset/sipp_scenario01.xml
@ -0,0 +1,109 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Sipwise NGCP Benchmark UAC Caller">
+  <send start_rtd="1" start_rtd="2">
+    <![CDATA[
+
+      INVITE sip:[field0 file="callee.csv" line=1]@[field3 file="callee.csv" line=1] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: <sip:[field0 file="caller.csv" line=1]@[field2 file="caller.csv" line=1]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:[field0 file="callee.csv" line=1]@[field3 file="callee.csv" line=1]>
+      Call-ID: NGCP%[field4 file="callee.csv" line=1]%///[call_id]
+      CSeq: 1 INVITE
+      Contact: <sip:[field0 file="caller.csv" line=1]@[local_ip]:[local_port]>
+      P-Early-Media: supported,inactive
+      Max-Forwards: 70
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 8
+      a=rtpmap:8 PCMA/8000
+      a=ptime:50
+
+    ]]>
+  </send>
+
+  <recv response="100" rtd="1" optional="true">
+  </recv>
+
+  <recv response="407" rtd="2" auth="true"/>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:[field0 file="callee.csv" line=1]@[field3 file="callee.csv" line=1]:[remote_port] SIP/2.0
+      [last_Via:]
+      From: <sip:[field0 file="caller.csv" line=1]@[field2 file="caller.csv" line=1]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:[field0 file="callee.csv" line=1]@[field3 file="callee.csv" line=1]>[peer_tag_param]
+      Call-ID: NGCP%[field4 file="callee.csv" line=1]%///[call_id]
+      CSeq: 1 ACK
+      Contact: <sip:[field0 file="caller.csv" line=1]@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause milliseconds="500"/>
+
+  <send start_rtd="3">
+    <![CDATA[
+
+      INVITE sip:[field0 file="callee.csv" line=1]@[field3 file="callee.csv" line=1] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: <sip:[field0 file="caller.csv" line=1]@[field2 file="caller.csv" line=1]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:[field0 file="callee.csv" line=1]@[field3 file="callee.csv" line=1]>
+      Call-ID: NGCP%[field4 file="callee.csv" line=1]%///[call_id]
+      CSeq: 2 INVITE
+      Contact: <sip:[field0 file="caller.csv" line=1]@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      [field1 file="caller.csv" line=1]
+      P-Early-Media: supported,inactive
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 8
+      a=rtpmap:8 PCMA/8000
+      a=ptime:50
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="403" rtd="3">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:[field0 file="callee.csv" line=1]@[field3 file="callee.csv" line=1]:[remote_port] SIP/2.0
+      [last_Via:]
+      From: <sip:[field0 file="caller.csv" line=1]@[field2 file="caller.csv" line=1]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:[field0 file="callee.csv" line=1]@[field3 file="callee.csv" line=1]>[peer_tag_param]
+      Call-ID: NGCP%[field4 file="callee.csv" line=1]%///[call_id]
+      CSeq: 2 ACK
+      Contact: <sip:[field0 file="caller.csv" line=1]@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200, 500, 1000"/>
+
+<!-- <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> -->
+
+</scenario>
+