[asterisk-commits] channels/pjsip/allow overlap: New test for RFC3578 overlap d... (testsuite[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Mar 22 14:15:14 CDT 2017
Anonymous Coward #1000019 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5277 )
Change subject: channels/pjsip/allow_overlap: New test for RFC3578 overlap dialling
......................................................................
channels/pjsip/allow_overlap: New test for RFC3578 overlap dialling
New test for the proposed implementation of RFC3578 overlap dialling support
in res_pjsip as detailed in ASTERISK-26864.
Simply checks whether partially matching INVITEs return 404 (when disabled)
or 484 (when enabled)
Change-Id: I60a497b920167a6d46c6c6fa823149b9843c01d7
---
A tests/channels/pjsip/allow_overlap/configs/ast1/extensions.conf
A tests/channels/pjsip/allow_overlap/configs/ast1/pjsip.conf
A tests/channels/pjsip/allow_overlap/sipp/nooverlap_ech.xml
A tests/channels/pjsip/allow_overlap/sipp/nooverlap_echo.xml
A tests/channels/pjsip/allow_overlap/sipp/nooverlap_echox.xml
A tests/channels/pjsip/allow_overlap/sipp/overlap_ech.xml
A tests/channels/pjsip/allow_overlap/sipp/overlap_echo.xml
A tests/channels/pjsip/allow_overlap/sipp/overlap_echox.xml
A tests/channels/pjsip/allow_overlap/test-config.yaml
M tests/channels/pjsip/tests.yaml
10 files changed, 801 insertions(+), 0 deletions(-)
Approvals:
Mark Michelson: Looks good to me, approved
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/tests/channels/pjsip/allow_overlap/configs/ast1/extensions.conf b/tests/channels/pjsip/allow_overlap/configs/ast1/extensions.conf
new file mode 100644
index 0000000..bcea565
--- /dev/null
+++ b/tests/channels/pjsip/allow_overlap/configs/ast1/extensions.conf
@@ -0,0 +1,16 @@
+[default]
+exten => echo,1,Answer()
+same => n,Echo()
+same => n,Hangup()
+
+exten => playback,1,Answer()
+same => n,Playback(hello-world)
+same => n,Hangup()
+
+exten => early,1,Progress()
+same => n,Playback(hello-world,noanswer)
+same => n,Hangup(INTERWORKING)
+
+;This dialstring can be altered once endpoints can be used directly
+exten => bob,1,Dial(PJSIP/sip:bob at 127.0.0.1:5062)
+same => n,Hangup()
diff --git a/tests/channels/pjsip/allow_overlap/configs/ast1/pjsip.conf b/tests/channels/pjsip/allow_overlap/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..cfa4298
--- /dev/null
+++ b/tests/channels/pjsip/allow_overlap/configs/ast1/pjsip.conf
@@ -0,0 +1,32 @@
+[local-transport-template](!)
+type=transport
+bind=127.0.0.1
+
+[local-transport-udp](local-transport-template)
+protocol=udp
+
+[endpoint-template-ipv4](!)
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+media_address=127.0.0.1
+
+[alice-ipv4-udp](endpoint-template-ipv4)
+auth=alice-auth
+allow_overlap=no
+
+[bob-ipv4-udp](endpoint-template-ipv4)
+auth=bob-auth
+;allow_overlap=yes ;(default)
+
+[auth-template](!)
+type=auth
+
+[alice-auth](auth-template)
+username=alice
+password=swordfish
+
+[bob-auth](auth-template)
+username=bob
+password=swordfish
+
diff --git a/tests/channels/pjsip/allow_overlap/sipp/nooverlap_ech.xml b/tests/channels/pjsip/allow_overlap/sipp/nooverlap_ech.xml
new file mode 100644
index 0000000..e1daf01
--- /dev/null
+++ b/tests/channels/pjsip/allow_overlap/sipp/nooverlap_ech.xml
@@ -0,0 +1,110 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to ech - expecting 404 response">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:ech@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="401" auth="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:ech@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:sipp@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:ech@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ [authentication username=alice password=swordfish]
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="404" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:ech@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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>
+
diff --git a/tests/channels/pjsip/allow_overlap/sipp/nooverlap_echo.xml b/tests/channels/pjsip/allow_overlap/sipp/nooverlap_echo.xml
new file mode 100644
index 0000000..39fb61f
--- /dev/null
+++ b/tests/channels/pjsip/allow_overlap/sipp/nooverlap_echo.xml
@@ -0,0 +1,132 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in initial INVITE">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="401" auth="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:sipp@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ [authentication username=alice password=swordfish]
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 3 BYE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <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>
+
diff --git a/tests/channels/pjsip/allow_overlap/sipp/nooverlap_echox.xml b/tests/channels/pjsip/allow_overlap/sipp/nooverlap_echox.xml
new file mode 100644
index 0000000..96bf879
--- /dev/null
+++ b/tests/channels/pjsip/allow_overlap/sipp/nooverlap_echox.xml
@@ -0,0 +1,110 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echox - expecting 404 response">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echox@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="401" auth="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echox@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:sipp@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echox@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ [authentication username=alice password=swordfish]
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="404" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echox@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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>
+
diff --git a/tests/channels/pjsip/allow_overlap/sipp/overlap_ech.xml b/tests/channels/pjsip/allow_overlap/sipp/overlap_ech.xml
new file mode 100644
index 0000000..6ee97c9
--- /dev/null
+++ b/tests/channels/pjsip/allow_overlap/sipp/overlap_ech.xml
@@ -0,0 +1,110 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to ech - expecting 484 response">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:ech@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="401" auth="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:ech@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:sipp@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:ech@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ [authentication username=bob password=swordfish]
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="484" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:ech@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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>
+
diff --git a/tests/channels/pjsip/allow_overlap/sipp/overlap_echo.xml b/tests/channels/pjsip/allow_overlap/sipp/overlap_echo.xml
new file mode 100644
index 0000000..97bd3eb
--- /dev/null
+++ b/tests/channels/pjsip/allow_overlap/sipp/overlap_echo.xml
@@ -0,0 +1,132 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in initial INVITE">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="401" auth="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:sipp@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ [authentication username=bob password=swordfish]
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 3 BYE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <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>
+
diff --git a/tests/channels/pjsip/allow_overlap/sipp/overlap_echox.xml b/tests/channels/pjsip/allow_overlap/sipp/overlap_echox.xml
new file mode 100644
index 0000000..956ccaf
--- /dev/null
+++ b/tests/channels/pjsip/allow_overlap/sipp/overlap_echox.xml
@@ -0,0 +1,110 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echox - expecting 404 response">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echox@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="401" auth="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echox@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:sipp@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echox@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ [authentication username=bob password=swordfish]
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="404" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echox@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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>
+
diff --git a/tests/channels/pjsip/allow_overlap/test-config.yaml b/tests/channels/pjsip/allow_overlap/test-config.yaml
new file mode 100644
index 0000000..6b80f20
--- /dev/null
+++ b/tests/channels/pjsip/allow_overlap/test-config.yaml
@@ -0,0 +1,48 @@
+testinfo:
+ summary: 'Tests incoming call behaviour with allow_overlap'
+ description: |
+ 'Run SIPp scenarios that send various calls to res_pjsip. Each
+ scenario matches a PJSIP endpoint that has the allow_overlap
+ option set to yes or no. The SIP response from PJSIP is checked
+ to ensure that the expected behavior matches what the option
+ should produce.'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ test-iterations:
+ # overlap disabled
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'nooverlap_ech.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice-ipv4-udp'} }
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'nooverlap_echo.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice-ipv4-udp'} }
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'nooverlap_echox.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice-ipv4-udp'} }
+ # overlap enabled
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'overlap_ech.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'bob-ipv4-udp'} }
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'overlap_echo.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'bob-ipv4-udp'} }
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'overlap_echox.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'bob-ipv4-udp'} }
+
+properties:
+ minversion: '13.15.0'
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'app_dial'
+ - asterisk : 'app_echo'
+ - asterisk : 'app_playback'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index c198cfe..7de1fdd 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -26,6 +26,7 @@
- dir: 'video_calls'
- test: 'accountcode'
- test: 'acl_call'
+ - test: 'allow_overlap'
- test: 'auth_security_events'
- test: 'call_pickup'
- test: 'dtmf_incompatible'
--
To view, visit https://gerrit.asterisk.org/5277
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I60a497b920167a6d46c6c6fa823149b9843c01d7
Gerrit-PatchSet: 2
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Richard Begg <asterisk at meric.id.au>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
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