[Asterisk-code-review] chan_pjsip: Extend existing test all_codecs_on_empty_reinvite (testsuite[18])
Henning Westerholt
asteriskteam at digium.com
Mon Apr 10 10:46:54 CDT 2023
Henning Westerholt has uploaded this change for review. ( https://gerrit.asterisk.org/c/testsuite/+/20048 )
Change subject: chan_pjsip: Extend existing test all_codecs_on_empty_reinvite
......................................................................
chan_pjsip: Extend existing test all_codecs_on_empty_reinvite
Extend existing test all_codecs_on_empty_reinvite to also test
for a scenario with an late offer. The existing test was moved
to the 'early-offer' directory, add a second directory for the
new tests. Change test include in tests.yaml to directory.
The late-offer test will fail until the bugfix is also merged.
ASTERISK-30473
Change-Id: I58f207172562e318d6ab9f352e3a695835b06bef
---
R tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/configs/ast1/extensions.conf
R tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/configs/ast1/pjsip.conf
R tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/sipp/empty-reinvite-all-codecs.xml
C tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/test-config.yaml
C tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/configs/ast1/extensions.conf
C tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/configs/ast1/pjsip.conf
A tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/sipp/empty-reinvite-all-codecs.xml
R tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/test-config.yaml
A tests/channels/pjsip/all_codecs_on_empty_reinvite/tests.yaml
M tests/channels/pjsip/tests.yaml
10 files changed, 214 insertions(+), 3 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/48/20048/1
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/configs/ast1/extensions.conf b/tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/configs/ast1/extensions.conf
similarity index 100%
rename from tests/channels/pjsip/all_codecs_on_empty_reinvite/configs/ast1/extensions.conf
rename to tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/configs/ast1/extensions.conf
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/configs/ast1/pjsip.conf b/tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/configs/ast1/pjsip.conf
similarity index 100%
rename from tests/channels/pjsip/all_codecs_on_empty_reinvite/configs/ast1/pjsip.conf
rename to tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/configs/ast1/pjsip.conf
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/sipp/empty-reinvite-all-codecs.xml b/tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/sipp/empty-reinvite-all-codecs.xml
similarity index 100%
rename from tests/channels/pjsip/all_codecs_on_empty_reinvite/sipp/empty-reinvite-all-codecs.xml
rename to tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/sipp/empty-reinvite-all-codecs.xml
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/test-config.yaml b/tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/test-config.yaml
similarity index 90%
copy from tests/channels/pjsip/all_codecs_on_empty_reinvite/test-config.yaml
copy to tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/test-config.yaml
index 26bf933..5295c44 100644
--- a/tests/channels/pjsip/all_codecs_on_empty_reinvite/test-config.yaml
+++ b/tests/channels/pjsip/all_codecs_on_empty_reinvite/early-offer/test-config.yaml
@@ -5,7 +5,8 @@
send an re-INVITE without SDP. Asterisk should send an SDP offer in the 200 OK
response containing all configured codecs on the endpoint if the parameter
all_codecs_on_empty_reinvite is activated. An re-INVITE with SDP should be
- answered normally.'
+ answered normally. In this scenario the usual early SDP offer in the INVITE is
+ used.'
test-modules:
test-object:
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/configs/ast1/extensions.conf b/tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/configs/ast1/extensions.conf
similarity index 100%
copy from tests/channels/pjsip/all_codecs_on_empty_reinvite/configs/ast1/extensions.conf
copy to tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/configs/ast1/extensions.conf
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/configs/ast1/pjsip.conf b/tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/configs/ast1/pjsip.conf
similarity index 100%
copy from tests/channels/pjsip/all_codecs_on_empty_reinvite/configs/ast1/pjsip.conf
copy to tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/configs/ast1/pjsip.conf
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/sipp/empty-reinvite-all-codecs.xml b/tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/sipp/empty-reinvite-all-codecs.xml
new file mode 100644
index 0000000..d11287b
--- /dev/null
+++ b/tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/sipp/empty-reinvite-all-codecs.xml
@@ -0,0 +1,189 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="empty reinvite all codecs">
+ <!-- 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:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="181"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ 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 0 8
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:0 PCMU/8000
+ a=ptime:20
+
+ ]]>
+ </send>
+
+ <pause milliseconds="500"/>
+
+<!-- For a re-INVITE with empty SDP Asterisk should return all codecs from the endpoint -->
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="m=audio [0-9]{1,5} RTP/AVP 9 8 0 97 107 101+..*"
+ search_in="body" check_it="true" assign_to="1"/>
+ <test assign_to="1" variable="1" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+<!-- Normal re-INIVTE, Asterisk should return negotiated codecs -->
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 3 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ 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 0 8
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:0 PCMU/8000
+ a=ptime:20
+
+ ]]>
+ </send>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="m=audio [0-9]{1,5} RTP/AVP 8 0+..*"
+ search_in="body" check_it="true" assign_to="1"/>
+ <test assign_to="1" variable="1" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 3 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+-->
+
+ <pause milliseconds="1000"/>
+
+ <send retrans="500">
+ <![CDATA[
+ BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 4 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv response="200"/>
+
+</scenario>
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/test-config.yaml b/tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/test-config.yaml
similarity index 91%
rename from tests/channels/pjsip/all_codecs_on_empty_reinvite/test-config.yaml
rename to tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/test-config.yaml
index 26bf933..e090da6 100644
--- a/tests/channels/pjsip/all_codecs_on_empty_reinvite/test-config.yaml
+++ b/tests/channels/pjsip/all_codecs_on_empty_reinvite/late-offer/test-config.yaml
@@ -5,7 +5,7 @@
send an re-INVITE without SDP. Asterisk should send an SDP offer in the 200 OK
response containing all configured codecs on the endpoint if the parameter
all_codecs_on_empty_reinvite is activated. An re-INVITE with SDP should be
- answered normally.'
+ answered normally. In this scenario a late SDP offer in the ACK is used.'
test-modules:
test-object:
diff --git a/tests/channels/pjsip/all_codecs_on_empty_reinvite/tests.yaml b/tests/channels/pjsip/all_codecs_on_empty_reinvite/tests.yaml
new file mode 100644
index 0000000..28afb2e
--- /dev/null
+++ b/tests/channels/pjsip/all_codecs_on_empty_reinvite/tests.yaml
@@ -0,0 +1,3 @@
+tests:
+ - test: 'early-offer'
+ - test: 'late-offer'
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index 3ae1bb2..34d7633 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -38,6 +38,7 @@
- dir: 'connected_line'
- dir: 'rtp_ptime'
- dir: 'ignore_183_wo_sdp'
+ - dir: 'all_codecs_on_empty_reinvite'
- test: 'accountcode'
- test: 'acl_call'
- test: 'allow_overlap'
@@ -70,4 +71,3 @@
- test: 'non_negotiated_frame_SSRC_change'
- test: 'content_disposition'
- test: 'reinvite_after_bye'
- - test: 'all_codecs_on_empty_reinvite'
--
To view, visit https://gerrit.asterisk.org/c/testsuite/+/20048
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Gerrit-Project: testsuite
Gerrit-Branch: 18
Gerrit-Change-Id: I58f207172562e318d6ab9f352e3a695835b06bef
Gerrit-Change-Number: 20048
Gerrit-PatchSet: 1
Gerrit-Owner: Henning Westerholt <hw at gilawa.com>
Gerrit-MessageType: newchange
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