asterisk · maximilianfridrich · Aug 24, 2023
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/configs/ast1/pjsip.conf b/tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/configs/ast1/pjsip.conf
@@ -5,6 +5,7 @@ timer_b=6400
 
 [global]
 all_codecs_on_empty_reinvite=yes
+debug=yes
 
 [local-transport-udp]
 type=transport
@@ -16,3 +17,4 @@ type=endpoint
 context=default
 media_address=127.0.0.1
 allow=!all,g722,alaw,ulaw,ilbc,opus
+codec_prefs_incoming_offer = prefer: configured, operation: intersect, keep: all, transcode: allow
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/configs/ast1/pjsip.conf b/tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/configs/ast1/pjsip.conf
@@ -5,6 +5,7 @@ timer_b=6400
 
 [global]
 all_codecs_on_empty_reinvite=yes
+debug=yes
 
 [local-transport-udp]
 type=transport
@@ -16,3 +17,4 @@ type=endpoint
 context=default
 media_address=127.0.0.1
 allow=!all,g722,alaw,ulaw,ilbc,opus
+codec_prefs_incoming_offer = prefer: configured, operation: intersect, keep: all, transcode: allow
diff --git a/...channels/pjsip/codec_negotiation/audio/common_codecs_default/configs/ast1/extensions.conf b/...channels/pjsip/codec_negotiation/audio/common_codecs_default/configs/ast1/extensions.conf
@@ -0,0 +1,8 @@
+[general]
+
+[globals]
+
+[calling]
+exten => bob,1,NoOp()
+	same => n,Dial(PJSIP/bob)
+	same => n,Hangup()
diff --git a/tests/channels/pjsip/codec_negotiation/audio/common_codecs_default/configs/ast1/pjsip.conf b/tests/channels/pjsip/codec_negotiation/audio/common_codecs_default/configs/ast1/pjsip.conf
@@ -0,0 +1,38 @@
+[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
+
+
+
+[endpoint-template](!)
+
+[alice](endpoint-template)
+type=endpoint
+allow=opus,g722,ulaw,alaw,h264
+context=calling
+direct_media=no
+media_address=127.0.0.1
+aors=alice
+
+[bob](endpoint-template)
+type=endpoint
+allow=opus,g722,ulaw,alaw,h264
+context=calling
+direct_media=no
+media_address=127.0.0.1
+aors=bob
+
+[bob]
+type=aor
+max_contacts=1
+contact=sip:[email protected]:5060\;transport=udp
diff --git a/tests/channels/pjsip/codec_negotiation/audio/common_codecs_default/sipp/alice.xml b/tests/channels/pjsip/codec_negotiation/audio/common_codecs_default/sipp/alice.xml
@@ -0,0 +1,101 @@
+<?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 107 9 8 0
+      a=rtpmap:107 opus/48000/2
+      a=fmtp:107 maxaveragebitrate=64000;sprop-stereo=1
+      a=rtpmap:9 G722/8000
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:0 PCMU/8000
+      a=ptime:20
+      a=maxptime:20
+      a=sendrecv
+      a=rtcp-mux
+    ]]>
+  </send>
+
+  <recv response="100" optional="true" />
+  <recv response="180" />
+
+  <recv response="200" rtd="true">
+    <!-- Save the To tag. -->
+    <action>
+      <ereg regexp="(;tag=.*)"
+          header="To:"
+          search_in="hdr"
+          check_it="true"
+          assign_to="remote_tag"/>
+      <ereg regexp="m=audio [0-9]{1,5} RTP/AVP 9 8 0"
+            search_in="body" check_it="true" assign_to="1"/>
+      <test assign_to="1" variable="1" compare="equal" value=""/>
+      <ereg regexp="G722"
+            search_in="body" check_it_inverse="false" assign_to="2"/>
+      <test assign_to="2" variable="2" compare="equal" value=""/>
+      <ereg regexp="PCMA"
+            search_in="body" check_it_inverse="false" assign_to="3"/>
+      <test assign_to="3" variable="3" compare="equal" value=""/>
+      <ereg regexp="OPUS"
+            search_in="body" check_it_inverse="true" assign_to="4"/>
+      <test assign_to="4" variable="4" compare="equal" value=""/>
+    </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>
+
+  <pause milliseconds="1000" />
+
+  <send retrans="500">
+    <![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] BYE
+      Contact: sip:alice@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Codec Negotiation Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+</scenario>
diff --git a/tests/channels/pjsip/codec_negotiation/audio/common_codecs_default/sipp/bob.xml b/tests/channels/pjsip/codec_negotiation/audio/common_codecs_default/sipp/bob.xml
@@ -0,0 +1,79 @@
+<?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>
+
+  <pause milliseconds="100" />
+
+  <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 9 8 0
+      a=rtpmap:9 G722/8000
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:0 PCMU/8000
+      a=ptime:20
+      a=maxptime:20
+      a=sendrecv
+    ]]>
+  </send>
+
+  <recv request="ACK"
+    rtd="true"
+    crlf="true">
+  </recv>
+
+  <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/codec_negotiation/audio/common_codecs_default/test-config.yaml b/tests/channels/pjsip/codec_negotiation/audio/common_codecs_default/test-config.yaml
@@ -0,0 +1,52 @@
+---
+testinfo:
+    summary:
+        'Test advanced codec negotiation in a call between two parties
+        with 3 common codecs '
+    description: |
+        'Two PJSIP endpoints are configured to support many codecs. 
+        Alice client is configured with Opus as additional codec.
+        All endpoints share the same codec settings in pjsip.conf
+        Alice calls Bob and in Bob\'s SDP answer he provides 3 codecs that Alice except Opus.
+        The common codecs are g722,pcma,pcmu.
+        Asterisk should not transcode in this case as the parties share at least one common codec.'
+
+properties:
+    dependencies:
+        - python: 'twisted'
+        - python: 'starpy'
+        - asterisk: 'app_dial'
+        - asterisk: 'res_pjsip'
+        - sipp:
+              version: 'v3.6.0'
+    tags:
+        - pjsip
+
+test-modules:
+    add-test-to-search-path: 'True'
+    test-object:
+        config-section: test-case-config
+        typename: 'sipp.SIPpTestCase'
+
+test-case-config:
+    memcheck-delay-stop: 7
+    # connect-ami: 'False'
+    fail-on-any: false
+    test-iterations:
+        # First iteration
+        -
+            scenarios:
+                # Bob receives call from Alice
+                - {'key-args':
+                       {'scenario': 'bob.xml', '-p': '5060', '-i': '127.0.0.3',
+                        '-s': 'alice', '-timeout': '20s', '-mi': '127.0.0.3'
+                       },
+                   'ordered-args':
+                       ['-timeout_error', '-key', 'custom_media_port', '6004']}
+                # Alice calls Bob
+                - {'key-args':
+                       {'scenario': 'alice.xml', '-p': '5060',
+                        '-i': '127.0.0.2', '-s': 'bob', '-timeout': '20s',
+                        '-mi': '127.0.0.2'},
+                   'ordered-args':
+                       ['-timeout_error', '-key', 'custom_media_port', '6004']}
diff --git a/...s/channels/pjsip/codec_negotiation/audio/common_codecs_union/configs/ast1/extensions.conf b/...s/channels/pjsip/codec_negotiation/audio/common_codecs_union/configs/ast1/extensions.conf
@@ -0,0 +1,8 @@
+[general]
+
+[globals]
+
+[calling]
+exten => bob,1,NoOp()
+	same => n,Dial(PJSIP/bob)
+	same => n,Hangup()
diff --git a/tests/channels/pjsip/codec_negotiation/audio/common_codecs_union/configs/ast1/pjsip.conf b/tests/channels/pjsip/codec_negotiation/audio/common_codecs_union/configs/ast1/pjsip.conf
@@ -0,0 +1,39 @@
+[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
+
+
+
+[endpoint-template](!)
+codec_prefs_outgoing_answer = prefer: pending, operation: union, keep: all, transcode: allow
+
+[alice](endpoint-template)
+type=endpoint
+allow=opus,g722,ulaw,alaw,h264
+context=calling
+direct_media=no
+media_address=127.0.0.1
+aors=alice
+
+[bob](endpoint-template)
+type=endpoint
+allow=opus,g722,ulaw,alaw,h264
+context=calling
+direct_media=no
+media_address=127.0.0.1
+aors=bob
+
+[bob]
+type=aor
+max_contacts=1
+contact=sip:[email protected]:5060\;transport=udp