[Asterisk-code-review] Add test for pjsip transfer with 481 (testsuite[18])

Friendly Automation asteriskteam at digium.com
Tue Mar 23 09:33:24 CDT 2021


Friendly Automation has submitted this change. ( https://gerrit.asterisk.org/c/testsuite/+/15669 )

Change subject: Add test for pjsip transfer with 481
......................................................................

Add test for pjsip transfer with 481

Change-Id: Iee0e5c4fd31b4d4f4bc833391d972ef87ecd902e
---
A tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/configs/ast1/extensions.conf
A tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/configs/ast1/pjsip.conf
A tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/alice.xml
A tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/bob.xml
A tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/charlie.xml
A tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/test-config.yaml
M tests/channels/pjsip/transfers/blind_transfer/tests.yaml
7 files changed, 603 insertions(+), 0 deletions(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved
  Friendly Automation: Approved for Submit



diff --git a/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/configs/ast1/extensions.conf b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/configs/ast1/extensions.conf
new file mode 100644
index 0000000..8cc124e
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/configs/ast1/extensions.conf
@@ -0,0 +1,16 @@
+[general]
+
+[globals]
+
+[transfertest]
+exten => bob,1,NoOp()
+	same => n,Dial(PJSIP/bob)
+	same => n,Hangup()
+
+exten => charlie,1,NoOp()
+	same => n,Wait(1)
+	same => n,Progress()
+	same => n,Wait(1)
+	same => n,Dial(PJSIP/charlie)
+	same => n,Hangup()
+
diff --git a/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/configs/ast1/pjsip.conf b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..bb3e07c
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/configs/ast1/pjsip.conf
@@ -0,0 +1,66 @@
+[system]
+type=system
+timer_t1=100
+timer_b=6400
+
+[global]
+type=global
+debug=yes
+
+[local-transport]
+type=transport
+bind=127.0.0.1
+protocol=udp
+
+[alice]
+type=endpoint
+allow=g722,ulaw,alaw
+context=transfertest
+direct_media=no
+media_address=127.0.0.1
+aors=alice
+
+[alice]
+type=aor
+max_contacts=1
+contact=sip:alice at 127.0.0.2:5060\;transport=udp
+
+[bob]
+type=endpoint
+allow=g722,ulaw,alaw
+context=transfertest
+direct_media=no
+media_address=127.0.0.1
+aors=bob
+
+[bob]
+type=aor
+max_contacts=1
+contact=sip:bob at 127.0.0.3:5060\;transport=udp
+
+[bob_two]
+type=endpoint
+allow=ulaw
+context=transfertest
+direct_media=no
+media_address=127.0.0.1
+aors=bob_two
+
+[bob_two]
+type=aor
+max_contacts=1
+contact=sip:bob_two at 127.0.0.3:5060\;transport=udp
+
+[charlie]
+type=endpoint
+allow=g722,ulaw,alaw
+context=transfertest
+direct_media=no
+media_address=127.0.0.1
+aors=charlie
+
+[charlie]
+type=aor
+max_contacts=1
+contact=sip:charlie at 127.0.0.4:5060\;transport=udp
+
diff --git a/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/alice.xml b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/alice.xml
new file mode 100644
index 0000000..acd98f8
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/alice.xml
@@ -0,0 +1,232 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Send Call">
+
+  <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: <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag[call_number]
+      To: <sip:[service]@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: [cseq] INVITE
+      Contact: <sip:alice@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=- 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [custom_media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true" />
+  <recv response="180" optional="true" />
+  <recv response="183" optional="true" />
+
+  <recv response="200" rtd="true">
+    <!-- Save the To tag. We'll need it when we send REFER -->
+    <action>
+      <ereg regexp="(;tag=.*)"
+          header="To:"
+          search_in="hdr"
+          check_it="true"
+          assign_to="remote_tag"/>
+    </action>
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      Call-ID: [call_id]
+      CSeq: [cseq] ACK
+      Contact: <sip:alice@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- Send audio -->
+  <nop>
+    <action>
+      <exec rtp_stream="lib/python/asterisk/audio.ulaw,1,0"/>
+    </action>
+  </nop>
+
+  <pause milliseconds="1000" crlf="true" />
+  <nop>
+    <action>
+		<exec rtp_stream="pause" />
+    </action>
+  </nop>
+
+  <!-- Put call on hold -->
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      [last_Via:]2
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      CSeq: [cseq+2] INVITE
+      Contact: <sip:alice@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=- 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [custom_media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+      a=sendonly
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true" />
+
+  <recv response="200" rtd="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      [last_Via:]2
+      [last_From:]
+      [last_To:]
+      Call-ID: [call_id]
+      CSeq: [cseq+2] ACK
+      Contact: <sip:alice@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause milliseconds="1000" crlf="true"/>
+
+  <!-- Blind transfer bob to charlie -->
+  <send retrans="500">
+    <![CDATA[
+
+      REFER sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      [last_Via:]3
+      From: <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag[call_number]
+      To: <sip:[service]@[remote_ip]:[remote_port]>[$remote_tag]
+      Call-ID: [call_id]
+      CSeq: [cseq+4] REFER
+      Contact: sip:alice@[local_ip]:[local_port];transport=[transport]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Refer-To: sip:charlie@[remote_ip]:[remote_port];user=phone
+      Referred-By: sip:alice@[local_ip]:[local_port]
+      Content-Length: 0
+
+    ]]>
+
+  </send>
+
+  <recv response="202" rtd="true">
+  </recv>
+
+  <!-- We should receive a NOTIFY from Asterisk with a 100 trying sipfrag -->
+  <recv request="NOTIFY">
+  </recv>
+
+  <send  crlf="true">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      [last_Event:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+    ]]>
+  </send>
+
+
+  <!-- We should receive a NOTIFY from Asterisk with a '183 Session Progress' sipfrag -->
+  <recv request="NOTIFY" crlf="true">
+    <action>
+       <ereg regexp="(SIP/2.0 183 Session Progress)"
+          search_in="msg"
+          check_it="true"
+          assign_to="sip_frag_ok"/>
+       <ereg regexp="([0-9]+)"
+           header="CSeq:"
+           search_in="hdr"
+           check_it="true"
+           assign_to="progress_cseq"/>
+       <ereg regexp=" (.*)"
+           header="Via:"
+           search_in="hdr"
+           check_it="true"
+           assign_to="progress_via"/>
+       <ereg regexp=" (.*)"
+           header="To:"
+           search_in="hdr"
+           check_it="true"
+           assign_to="progress_to"/>
+    </action>
+  </recv>
+
+  <!-- Transfer should have successfully occurred so now we need to hang up -->
+  <send start_txn="bye">
+    <![CDATA[
+
+      BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      [last_Via:]4
+      From: <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag[call_number]
+      To: <sip:[service]@[remote_ip]:[remote_port]>[$remote_tag]
+      Call-ID: [call_id]
+      CSeq: [cseq+5] BYE
+      Contact: sip:alice@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" response_txn="bye">
+  </recv>
+
+    <send>
+    <![CDATA[
+
+      SIP/2.0 481 Subscription Does Not Exist
+      Via: [$progress_via]
+      [last_From:]
+      To: [$progress_to]
+      [last_Call-ID:]
+      CSeq: [$progress_cseq] NOTIFY
+      [last_Event:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <Reference variables="sip_frag_ok" />
+
+</scenario>
diff --git a/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/bob.xml b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/bob.xml
new file mode 100644
index 0000000..3870eac
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/bob.xml
@@ -0,0 +1,80 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+
+  <recv request="INVITE" crlf="true" />
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+      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 [custom_media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv request="ACK"
+    rtd="true"
+    crlf="true">
+  </recv>
+
+  <!-- Send audio -->
+  <nop>
+    <action>
+      <exec rtp_stream="lib/python/asterisk/audio.ulaw,3,0"/>
+    </action>
+  </nop>
+
+  <recv request="BYE" />
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/charlie.xml b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/charlie.xml
new file mode 100644
index 0000000..d14fb7a
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/sipp/charlie.xml
@@ -0,0 +1,100 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Basic UAS responder">
+
+  <recv request="INVITE" crlf="true">
+      <action>
+          <!-- Save the From tag. We'll need it when we send our BYE -->
+          <ereg regexp="(;tag=.*)"
+              header="From:"
+              search_in="hdr"
+              check_it="true"
+              assign_to="remote_tag"/>
+          <!-- Save the From user portion of URI. We'll need it when we send our BYE -->
+          <ereg regexp="(sip:bob)"
+              header="From:"
+              search_in="hdr"
+              check_it="true"
+              assign_to="remote_user"/>
+          <!-- Check the Referred-By header. -->
+          <ereg regexp="sip:alice at 127.0.0.2:5060"
+              header="Referred-By"
+              search_in="hdr"
+              check_it="true"
+              assign_to="referred_by"/>
+      </action>
+  </recv>
+
+  <!-- Answer inbound call -->
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:charlie@[local_ip]:[local_port];transport=[transport]>
+      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 [custom_media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv request="ACK"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <!-- Send audio -->
+  <nop>
+    <action>
+      <exec rtp_stream="lib/python/asterisk/audio.ulaw,1,0"/>
+    </action>
+  </nop>
+
+  <!-- Allow 5s of audio to be sent to bob -->
+  <pause milliseconds="1000" />
+  <nop>
+    <action>
+		<exec rtp_stream="pause" />
+    </action>
+  </nop>
+
+  <send retrans="0">
+    <![CDATA[
+
+      BYE [$remote_user]@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: <sip:charlie@[local_ip]:[local_port]>;tag=[pid]SIPpTag[call_number]
+      To: <[$remote_user]@[remote_ip]:[remote_port]>[$remote_tag]
+      Call-ID: [call_id]
+      CSeq: [cseq] BYE
+      Contact: sip:charlie@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" timeout="2000" crlf="true">
+  </recv>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+  <Reference variables="referred_by" />
+
+</scenario>
+
diff --git a/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/test-config.yaml b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/test-config.yaml
new file mode 100644
index 0000000..903272a
--- /dev/null
+++ b/tests/channels/pjsip/transfers/blind_transfer/caller_with_hold_481/test-config.yaml
@@ -0,0 +1,108 @@
+testinfo:
+    summary:     'Test SIP Blind Transfer (caller transfers with hold and a 481)'
+    description: |
+        'This test verifies a SIP Blind transfer with putting the callee on
+        hold before the transfer occurs. This uses a SIPp instance each for
+        "Alice", "Bob", and "Charlie". Alice calls Bob through Asterisk. Alice
+        puts Bob on hold and Alice then blind transfers Bob to Charlie.
+        Charlie's dialplan sends a Progress to Alice but Alice sends a BYE
+        then a 481 reponse to the NOTIFY 183 sipfrag because the subscription
+        no longer exists.'
+
+properties:
+    dependencies:
+        - python : 'twisted'
+        - python : 'starpy'
+        - asterisk : 'app_dial'
+        - asterisk : 'res_pjsip'
+        - sipp :
+            version : 'v3.4.1'
+    tags:
+        - pjsip
+        - transfer
+
+test-modules:
+    add-test-to-search-path: 'True'
+    test-object:
+        config-section: test-case-config
+        typename: 'sipp.SIPpTestCase'
+    modules:
+        -
+            config-section: 'ami-config'
+            typename: 'ami.AMIEventModule'
+
+test-case-config:
+    memcheck-delay-stop: 7
+    connect-ami: 'True'
+    fail-on-any: False
+    test-iterations:
+        # First iteration
+        -
+            scenarios:
+                # Charlie receives transfered call and sends audio to Bob.
+                - { 'key-args': {'scenario': 'charlie.xml', '-p': '5060', '-i': '127.0.0.4', '-timeout': '20s', '-mi': '127.0.0.4'},
+                    'ordered-args': ['-nd', '-timeout_error', '-key', 'custom_media_port', '6004'] }
+                # Bob receives call from Alice and sends audio.
+                - { 'key-args': {'scenario': 'bob.xml', '-p': '5060', '-i': '127.0.0.3', '-s': 'alice', '-timeout': '20s', '-mi': '127.0.0.3'},
+                    'ordered-args': ['-nd', '-timeout_error', '-key', 'custom_media_port', '6004'] }
+                # Alice calls Bob and sends audio. Alice then attempts to blind transfer Bob to Charlie.
+                - { 'key-args': {'scenario': 'alice.xml', '-p': '5060', '-i': '127.0.0.2', '-s': 'bob', '-timeout': '20s', '-mi': '127.0.0.2'},
+                    'ordered-args': ['-nd', '-timeout_error', '-key', 'custom_media_port', '6004'] }
+
+ami-config:
+        -
+            type: 'headermatch'
+            conditions:
+                match:
+                    Event: 'MusicOnHoldStart'
+                    Channel: 'PJSIP/bob-.{8}'
+            count: 1
+        -
+            type: 'headermatch'
+            conditions:
+                match:
+                    Event: 'MusicOnHoldStop'
+                    Channel: 'PJSIP/bob-.{8}'
+            count: 1
+        -
+            type: 'headermatch'
+            conditions:
+                match:
+                    Event: 'VarSet'
+                    Variable: 'SIPTRANSFER'
+                    Value: 'yes'
+            count: 1
+        -
+            type: 'headermatch'
+            conditions:
+                match:
+                    Event: 'VarSet'
+                    Channel: 'PJSIP/bob-.{8}'
+                    Variable: 'SIPREFERRINGCONTEXT'
+                    Value: 'transfertest'
+            count: 1
+        -
+            type: 'headermatch'
+            conditions:
+                match:
+                    Event: 'VarSet'
+                    Channel: 'PJSIP/bob-.{8}'
+                    Variable: '_{0,2}SIPREFERREDBYHDR'
+                    Value: 'sip:alice at 127.0.0.2:5060'
+            count: 1
+        -
+            type: 'headermatch'
+            conditions:
+                match:
+                    Event: 'VarSet'
+                    Variable: 'SIPREFERTOHDR'
+                    Value: 'sip:charlie at 127.0.0.1'
+            count: 1
+        -
+            type: 'headermatch'
+            conditions:
+                match:
+                    Event: 'BlindTransfer'
+                    TransfererChannel: 'PJSIP/alice-.{8}'
+            count: 1
+
diff --git a/tests/channels/pjsip/transfers/blind_transfer/tests.yaml b/tests/channels/pjsip/transfers/blind_transfer/tests.yaml
index 2c24a89..39a3cca 100644
--- a/tests/channels/pjsip/transfers/blind_transfer/tests.yaml
+++ b/tests/channels/pjsip/transfers/blind_transfer/tests.yaml
@@ -7,6 +7,7 @@
     - test: 'caller_direct_media'
     - test: 'caller_refer_only'
     - test: 'caller_with_hold'
+    - test: 'caller_with_hold_481'
     - test: 'caller_with_hold_drop_options'
     - test: 'disallow'
     - test: 'goto_on_blindxfr'

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Gerrit-Project: testsuite
Gerrit-Branch: 18
Gerrit-Change-Id: Iee0e5c4fd31b4d4f4bc833391d972ef87ecd902e
Gerrit-Change-Number: 15669
Gerrit-PatchSet: 2
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-MessageType: merged
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