[Asterisk-code-review] pjsip: Add SDP offer/answer test for bundle. (testsuite[master])
Jenkins2
asteriskteam at digium.com
Mon Jul 17 18:22:13 CDT 2017
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5982 )
Change subject: pjsip: Add SDP offer/answer test for bundle.
......................................................................
pjsip: Add SDP offer/answer test for bundle.
This change adds an SDP negotiation test for bundle to ensure
that when bundle is enabled we place the proper attributes in
the messages.
ASTERISK-27118
Change-Id: Ia7fcefcba248d1814b2fcbffb80b3a90cda13dd0
---
A tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/configs/ast1/extensions.conf
A tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/configs/ast1/pjsip.conf
A tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/sipp/uac-multiple-video-with-audio.xml
A tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/test-config.yaml
M tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/tests.yaml
5 files changed, 167 insertions(+), 0 deletions(-)
Approvals:
Joshua Colp: Looks good to me, approved; Verified
Kevin Harwell: Looks good to me, but someone else must approve
Jenkins2: Approved for Submit
diff --git a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/configs/ast1/extensions.conf b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/configs/ast1/extensions.conf
new file mode 100644
index 0000000..6955acb
--- /dev/null
+++ b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/configs/ast1/extensions.conf
@@ -0,0 +1,5 @@
+[default]
+
+exten => answer,1,NoOp()
+ same => n,Answer()
+ same => n,Hangup()
diff --git a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/configs/ast1/pjsip.conf b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..d328216
--- /dev/null
+++ b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/configs/ast1/pjsip.conf
@@ -0,0 +1,14 @@
+[local-transport-udp]
+type=transport
+bind=127.0.0.1
+protocol=udp
+
+[endpoint-template](!)
+type=endpoint
+context=default
+media_address=127.0.0.1
+max_video_streams=10
+bundle=yes
+
+[alice](endpoint-template)
+allow=!all,g722,ulaw,alaw,h264,h263
diff --git a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/sipp/uac-multiple-video-with-audio.xml b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/sipp/uac-multiple-video-with-audio.xml
new file mode 100644
index 0000000..4d52ad8
--- /dev/null
+++ b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/sipp/uac-multiple-video-with-audio.xml
@@ -0,0 +1,120 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:answer@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Codec 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
+ a=group:BUNDLE audio video
+ m=audio 6000 RTP/AVP 9 0 8 101
+ a=rtpmap:9 G722/8000
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=ptime:20
+ a=maxptime:20
+ a=sendrecv
+ a=mid:audio
+ m=video 6001 RTP/AVP 99 34
+ a=rtpmap:99 H264/90000
+ a=rtpmap:34 H263/90000
+ a=sendrecv
+ a=mid:video
+ m=video 6002 RTP/AVP 99
+ c=IN IP[media_ip_type] [media_ip]
+ a=rtpmap:99 H264/90000
+ a=sendrecv
+ a=mid:video
+ m=video 6003 RTP/AVP 34
+ a=rtpmap:34 H263/90000
+ a=sendrecv
+ a=mid:video
+
+ ]]>
+ </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">
+ <action>
+ <ereg regexp="a=group:BUNDLE audio video"
+ search_in="body" check_it="true" assign_to="1"/>
+ <test assign_to="1" variable="1" compare="equal" value=""/>
+ <ereg regexp="a=mid:audio"
+ search_in="body" check_it="true" assign_to="2"/>
+ <test assign_to="2" variable="2" compare="equal" value=""/>
+ <ereg regexp="a=mid:video"
+ search_in="body" check_it="true" assign_to="3"/>
+ <test assign_to="3" variable="3" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:answer@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:[service]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:alice@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Codec Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ 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/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/test-config.yaml b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/test-config.yaml
new file mode 100644
index 0000000..d3185f5
--- /dev/null
+++ b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/bundled/test-config.yaml
@@ -0,0 +1,27 @@
+testinfo:
+ summary: 'Test offers with multiple video streams/one audio stream and bundled'
+ description: |
+ This tests inbound offers that contain multiple video
+ media streams and a single audio stream with bundle enabled.
+ Asterisk should accept all the streams in a single bundle group.
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ fail-on-any: False
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'uac-multiple-video-with-audio.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice'} }
+
+properties:
+ minversion: '15.0.0'
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/tests.yaml b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/tests.yaml
index a2bb275..1a143f3 100644
--- a/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/tests.yaml
+++ b/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/multiple-media-stream/audio-video/accept/tests.yaml
@@ -1,5 +1,6 @@
# Enter tests here in the order they should be considered for execution:
tests:
+ - test: 'bundled'
- test: 'multiple-audio'
- test: 'multiple-video'
--
To view, visit https://gerrit.asterisk.org/5982
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: Ia7fcefcba248d1814b2fcbffb80b3a90cda13dd0
Gerrit-Change-Number: 5982
Gerrit-PatchSet: 1
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
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