[Asterisk-code-review] testsuite: validate rtcp-mux sdp (testsuite[master])
Jenkins2
asteriskteam at digium.com
Mon Aug 20 08:58:44 CDT 2018
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/9928 )
Change subject: testsuite: validate rtcp-mux sdp
......................................................................
testsuite: validate rtcp-mux sdp
add tests to verify the presence of rtcp-mux only if the remote offer has it
or if asterisk is generating the initial offer and the feature is enabled on
the endpoint
ASTERISK-28007 #close
Change-Id: I4d6f139c106add84819736c0c9e886d6cb10e1f0
---
A tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/configs/ast1/extensions.conf
A tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/configs/ast1/pjsip.conf
A tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/sipp/A_PARTY.xml
A tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/sipp/B_PARTY.xml
A tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/test-config.yaml
A tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/configs/ast1/extensions.conf
A tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/configs/ast1/pjsip.conf
A tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/sipp/A_PARTY.xml
A tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/sipp/B_PARTY.xml
A tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/test-config.yaml
M tests/channels/pjsip/rtcp/rtcp_mux_callee/tests.yaml
A tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/configs/ast1/extensions.conf
A tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/configs/ast1/pjsip.conf
A tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/sipp/A_PARTY.xml
A tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/test-config.yaml
A tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/configs/ast1/extensions.conf
A tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/configs/ast1/pjsip.conf
A tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/sipp/A_PARTY.xml
A tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/test-config.yaml
M tests/channels/pjsip/rtcp/rtcp_mux_caller/tests.yaml
20 files changed, 995 insertions(+), 0 deletions(-)
Approvals:
Joshua Colp: Looks good to me, but someone else must approve
George Joseph: Looks good to me, approved
Jenkins2: Approved for Submit
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/configs/ast1/extensions.conf b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/configs/ast1/extensions.conf
new file mode 100644
index 0000000..ad7b155
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/configs/ast1/extensions.conf
@@ -0,0 +1,11 @@
+[general]
+static=yes
+writeprotect=yes
+autofallthrough=yes
+clearglobalvars=no
+priorityjumping=yes
+
+[globals]
+
+[default]
+exten => _X.,1,Dial(pjsip/sbc,180)
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/configs/ast1/pjsip.conf b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..9cdc056
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/configs/ast1/pjsip.conf
@@ -0,0 +1,57 @@
+;--
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements start
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[general]
+sipdebug = yes
+
+[PEER_A]
+port = 5061
+
+[sbc]
+port = 5700
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements end
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+--;
+
+
+[global]
+type = global
+debug = yes
+
+[transport-udp]
+type = transport
+protocol = udp
+bind = 0.0.0.0:5060
+
+[PEER_A]
+type = aor
+contact = sip:127.0.0.1:5061
+
+[PEER_A]
+type = identify
+endpoint = PEER_A
+match = 127.0.0.1
+
+[PEER_A]
+type = endpoint
+context = default
+allow = ulaw
+direct_media = no
+aors = PEER_A
+
+[sbc]
+type = aor
+contact = sip:127.0.0.1:5700
+
+[sbc]
+type = endpoint
+context = callcontrol
+allow = ulaw
+direct_media = no
+aors = sbc
+rtcp_mux = yes
+
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/sipp/A_PARTY.xml b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/sipp/A_PARTY.xml
new file mode 100644
index 0000000..9c092b1
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/sipp/A_PARTY.xml
@@ -0,0 +1,104 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+<!-- -->
+<!-- Sipp 'uac' scenario with pcap (rtp) play -->
+<!-- -->
+<scenario name="CONTENT_TYPE_PARAMS">
+ <!-- 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:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test at voxbone.com>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 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[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 9000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <!-- By adding rrs="true" (Record Route Sets), the route sets -->
+ <!-- are saved and used for following messages sent. Useful to test -->
+ <!-- against stateful SIP proxies/B2BUAs. -->
+ <recv response="200" rtd="true" crlf="true">
+ </recv>
+
+ <!-- Packet lost can be simulated in any send/recv message by -->
+ <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="1000"/>
+ <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+</scenario>
+
+
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/sipp/B_PARTY.xml b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/sipp/B_PARTY.xml
new file mode 100644
index 0000000..cef056a
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/sipp/B_PARTY.xml
@@ -0,0 +1,88 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="CONTENT_TYPE_PARAMS">
+
+<User variables="dummy" />
+<recv request="INVITE" crlf="true" rrs="true">
+ <action>
+ <ereg regexp="a=rtcp-mux" search_in="body" check_it="true" assign_to="dummy" />
+ </action>
+</recv>
+<Reference variables="dummy" />
+
+
+
+<send>
+<![CDATA[
+
+SIP/2.0 100 Trying
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+
+<send>
+<![CDATA[
+
+SIP/2.0 180 Ringing
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+
+<send retrans="500">
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:];tag=[call_number]
+[last_CSeq:]
+Contact: <sip:736596371553211257@[local_ip]:[local_port];user=phone>
+Content-Type: application/sdp
+Content-Length: [len]
+
+v=0
+o=HuaweiSoftX3000 6644052 6644052 IN IP[local_ip_type] [local_ip]
+s=Sip Call
+c=IN IP[local_ip_type] [local_ip]
+t=0 0
+m=audio 8000 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+
+]]>
+</send>
+
+
+<recv request="ACK" rtd="true" crlf="true">
+</recv>
+
+<recv request="BYE">
+</recv>
+
+<send>
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_From:]
+[last_To:]
+[last_Call-ID:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+</scenario>
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/test-config.yaml b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/test-config.yaml
new file mode 100644
index 0000000..463bd15
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_enabled/test-config.yaml
@@ -0,0 +1,28 @@
+testinfo:
+ summary: 'Test that if Asterisk does put rtcp-mux in the INVITE it is enabled in the endpoint'
+ description: |
+ 'Asterisk has an enpoint with rtcp-mux enabled, when sending an invite using that endpoint
+ rtcp-mux should be present'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ memcheck-delay-stop: 7
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'A_PARTY.xml', '-i': '127.0.0.1', '-p': '5061', '-s': '3200000000'} }
+ - { 'key-args': {'scenario': 'B_PARTY.xml', '-i': '127.0.0.1', '-p': '5700'} }
+
+
+properties:
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/configs/ast1/extensions.conf b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/configs/ast1/extensions.conf
new file mode 100644
index 0000000..ad7b155
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/configs/ast1/extensions.conf
@@ -0,0 +1,11 @@
+[general]
+static=yes
+writeprotect=yes
+autofallthrough=yes
+clearglobalvars=no
+priorityjumping=yes
+
+[globals]
+
+[default]
+exten => _X.,1,Dial(pjsip/sbc,180)
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/configs/ast1/pjsip.conf b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..87d60d3
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/configs/ast1/pjsip.conf
@@ -0,0 +1,57 @@
+;--
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements start
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[general]
+sipdebug = yes
+
+[PEER_A]
+port = 5061
+
+[sbc]
+port = 5700
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements end
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+--;
+
+
+[global]
+type = global
+debug = yes
+
+[transport-udp]
+type = transport
+protocol = udp
+bind = 0.0.0.0:5060
+
+[PEER_A]
+type = aor
+contact = sip:127.0.0.1:5061
+
+[PEER_A]
+type = identify
+endpoint = PEER_A
+match = 127.0.0.1
+
+[PEER_A]
+type = endpoint
+context = default
+allow = ulaw
+direct_media = no
+aors = PEER_A
+
+[sbc]
+type = aor
+contact = sip:127.0.0.1:5700
+
+[sbc]
+type = endpoint
+context = callcontrol
+allow = ulaw
+direct_media = no
+aors = sbc
+rtcp_mux = no
+
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/sipp/A_PARTY.xml b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/sipp/A_PARTY.xml
new file mode 100644
index 0000000..9c092b1
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/sipp/A_PARTY.xml
@@ -0,0 +1,104 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+<!-- -->
+<!-- Sipp 'uac' scenario with pcap (rtp) play -->
+<!-- -->
+<scenario name="CONTENT_TYPE_PARAMS">
+ <!-- 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:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test at voxbone.com>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 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[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 9000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <!-- By adding rrs="true" (Record Route Sets), the route sets -->
+ <!-- are saved and used for following messages sent. Useful to test -->
+ <!-- against stateful SIP proxies/B2BUAs. -->
+ <recv response="200" rtd="true" crlf="true">
+ </recv>
+
+ <!-- Packet lost can be simulated in any send/recv message by -->
+ <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="1000"/>
+ <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+</scenario>
+
+
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/sipp/B_PARTY.xml b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/sipp/B_PARTY.xml
new file mode 100644
index 0000000..9c968fb
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/sipp/B_PARTY.xml
@@ -0,0 +1,86 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="CONTENT_TYPE_PARAMS">
+
+<User variables="dummy" />
+<recv request="INVITE" crlf="true" rrs="true">
+ <action>
+ <ereg regexp="a=rtcp-mux" search_in="body" check_it_inverse="true" assign_to="dummy" />
+ </action>
+</recv>
+<Reference variables="dummy" />
+
+<send>
+<![CDATA[
+
+SIP/2.0 100 Trying
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+
+<send>
+<![CDATA[
+
+SIP/2.0 180 Ringing
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+
+<send retrans="500">
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_Call-ID:]
+[last_From:]
+[last_To:];tag=[call_number]
+[last_CSeq:]
+Contact: <sip:736596371553211257@[local_ip]:[local_port];user=phone>
+Content-Type: application/sdp
+Content-Length: [len]
+
+v=0
+o=HuaweiSoftX3000 6644052 6644052 IN IP[local_ip_type] [local_ip]
+s=Sip Call
+c=IN IP[local_ip_type] [local_ip]
+t=0 0
+m=audio 8000 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+
+]]>
+</send>
+
+
+<recv request="ACK" rtd="true" crlf="true">
+</recv>
+
+<recv request="BYE">
+</recv>
+
+<send>
+<![CDATA[
+
+SIP/2.0 200 OK
+[last_Via:]
+[last_From:]
+[last_To:]
+[last_Call-ID:]
+[last_CSeq:]
+Content-Length: 0
+
+]]>
+</send>
+</scenario>
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/test-config.yaml b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/test-config.yaml
new file mode 100644
index 0000000..525ecd1
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/rtcp_mux_not_enabled/test-config.yaml
@@ -0,0 +1,28 @@
+testinfo:
+ summary: 'Test that if Asterisk does not put rtcp-mux in the INVITE it is not enabled in the endpoint'
+ description: |
+ 'Asterisk has an enpoint with rtcp-mux disabled, when sending an invite using that endpoint
+ rtcp-mux should not be present'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ memcheck-delay-stop: 7
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'A_PARTY.xml', '-i': '127.0.0.1', '-p': '5061', '-s': '3200000000'} }
+ - { 'key-args': {'scenario': 'B_PARTY.xml', '-i': '127.0.0.1', '-p': '5700'} }
+
+
+properties:
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_callee/tests.yaml b/tests/channels/pjsip/rtcp/rtcp_mux_callee/tests.yaml
index f587a6f..33a6d7e 100644
--- a/tests/channels/pjsip/rtcp/rtcp_mux_callee/tests.yaml
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_callee/tests.yaml
@@ -2,3 +2,5 @@
tests:
- test: 'direct'
- test: 'ice'
+ - test: 'rtcp_mux_enabled'
+ - test: 'rtcp_mux_not_enabled'
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/configs/ast1/extensions.conf b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/configs/ast1/extensions.conf
new file mode 100644
index 0000000..704532f
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/configs/ast1/extensions.conf
@@ -0,0 +1,13 @@
+[general]
+static=yes
+writeprotect=yes
+autofallthrough=yes
+clearglobalvars=no
+priorityjumping=yes
+
+[globals]
+
+[default]
+exten => _X.,1,Answer()
+exten => _X.,2,Echo()
+exten => h,1,Hangup()
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/configs/ast1/pjsip.conf b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..01e4843
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/configs/ast1/pjsip.conf
@@ -0,0 +1,57 @@
+;--
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements start
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[general]
+sipdebug = yes
+
+[PEER_A]
+port = 5061
+
+[sbc]
+port = 5700
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements end
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+--;
+
+
+[global]
+type = global
+debug = yes
+
+[transport-udp]
+type = transport
+protocol = udp
+bind = 0.0.0.0:5060
+
+[PEER_A]
+type = aor
+contact = sip:127.0.0.1:5061
+
+[PEER_A]
+type = identify
+endpoint = PEER_A
+match = 127.0.0.1
+
+[PEER_A]
+type = endpoint
+context = default
+allow = ulaw
+direct_media = no
+rtcp_mux = yes
+aors = PEER_A
+
+[sbc]
+type = aor
+contact = sip:127.0.0.1:5700
+
+[sbc]
+type = endpoint
+context = callcontrol
+allow = ulaw
+direct_media = no
+aors = sbc
+
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/sipp/A_PARTY.xml b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/sipp/A_PARTY.xml
new file mode 100644
index 0000000..ac6e666
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/sipp/A_PARTY.xml
@@ -0,0 +1,111 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+<!-- -->
+<!-- Sipp 'uac' scenario with pcap (rtp) play -->
+<!-- -->
+<scenario name="CONTENT_TYPE_PARAMS">
+ <!-- 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:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test at voxbone.com>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 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[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 9000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <!-- By adding rrs="true" (Record Route Sets), the route sets -->
+ <!-- are saved and used for following messages sent. Useful to test -->
+ <!-- against stateful SIP proxies/B2BUAs. -->
+ <User variables="dummy" />
+ <recv response="200" rtd="true" crlf="true">
+ <action>
+ <ereg regexp="a=rtcp-mux" search_in="body" check_it_inverse="true" assign_to="dummy" />
+ </action>
+ </recv>
+ <Reference variables="dummy" />
+
+
+ <!-- Packet lost can be simulated in any send/recv message by -->
+ <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="1000"/>
+ <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+</scenario>
+
+
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/test-config.yaml b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/test-config.yaml
new file mode 100644
index 0000000..e1a6e4d
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_not_offered/test-config.yaml
@@ -0,0 +1,27 @@
+testinfo:
+ summary: 'Test that if Asterisk does not put rtcp-mux in the response if it is not requested'
+ description: |
+ 'Endpôint A has ice enabled and sends an invite w/o rtcp-mux in the SDP. The
+ ANSWER from asterisk should not contain rtcp-mux'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ memcheck-delay-stop: 7
+ fail-on-any: False
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'A_PARTY.xml', '-i': '127.0.0.1', '-p': '5061', '-s': '3200000000'} }
+
+
+properties:
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/configs/ast1/extensions.conf b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/configs/ast1/extensions.conf
new file mode 100644
index 0000000..704532f
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/configs/ast1/extensions.conf
@@ -0,0 +1,13 @@
+[general]
+static=yes
+writeprotect=yes
+autofallthrough=yes
+clearglobalvars=no
+priorityjumping=yes
+
+[globals]
+
+[default]
+exten => _X.,1,Answer()
+exten => _X.,2,Echo()
+exten => h,1,Hangup()
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/configs/ast1/pjsip.conf b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..01e4843
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/configs/ast1/pjsip.conf
@@ -0,0 +1,57 @@
+;--
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements start
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[general]
+sipdebug = yes
+
+[PEER_A]
+port = 5061
+
+[sbc]
+port = 5700
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements end
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+--;
+
+
+[global]
+type = global
+debug = yes
+
+[transport-udp]
+type = transport
+protocol = udp
+bind = 0.0.0.0:5060
+
+[PEER_A]
+type = aor
+contact = sip:127.0.0.1:5061
+
+[PEER_A]
+type = identify
+endpoint = PEER_A
+match = 127.0.0.1
+
+[PEER_A]
+type = endpoint
+context = default
+allow = ulaw
+direct_media = no
+rtcp_mux = yes
+aors = PEER_A
+
+[sbc]
+type = aor
+contact = sip:127.0.0.1:5700
+
+[sbc]
+type = endpoint
+context = callcontrol
+allow = ulaw
+direct_media = no
+aors = sbc
+
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/sipp/A_PARTY.xml b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/sipp/A_PARTY.xml
new file mode 100644
index 0000000..0c5c66a
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/sipp/A_PARTY.xml
@@ -0,0 +1,112 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+<!-- -->
+<!-- Sipp 'uac' scenario with pcap (rtp) play -->
+<!-- -->
+<scenario name="CONTENT_TYPE_PARAMS">
+ <!-- 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:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test at voxbone.com>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 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[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 9000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+ a=rtcp-mux
+
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <!-- By adding rrs="true" (Record Route Sets), the route sets -->
+ <!-- are saved and used for following messages sent. Useful to test -->
+ <!-- against stateful SIP proxies/B2BUAs. -->
+ <User variables="dummy" />
+ <recv response="200" rtd="true" crlf="true">
+ <action>
+ <ereg regexp="a=rtcp-mux" search_in="body" check_it="true" assign_to="dummy" />
+ </action>
+ </recv>
+ <Reference variables="dummy" />
+
+
+ <!-- Packet lost can be simulated in any send/recv message by -->
+ <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="1000"/>
+ <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@voxbone.com>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+</scenario>
+
+
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/test-config.yaml b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/test-config.yaml
new file mode 100644
index 0000000..94f9d84
--- /dev/null
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_caller/rtcp_mux_offered/test-config.yaml
@@ -0,0 +1,27 @@
+testinfo:
+ summary: 'Test that if Asterisk does put rtcp-mux in the response if it is requested'
+ description: |
+ 'Endpôint A has ice enabled and sends an invite with rtcp-mux in the SDP. The
+ ANSWER from asterisk should contain rtcp-mux'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ memcheck-delay-stop: 7
+ fail-on-any: False
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'A_PARTY.xml', '-i': '127.0.0.1', '-p': '5061', '-s': '3200000000'} }
+
+
+properties:
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/rtcp/rtcp_mux_caller/tests.yaml b/tests/channels/pjsip/rtcp/rtcp_mux_caller/tests.yaml
index f587a6f..06ee2d4 100644
--- a/tests/channels/pjsip/rtcp/rtcp_mux_caller/tests.yaml
+++ b/tests/channels/pjsip/rtcp/rtcp_mux_caller/tests.yaml
@@ -2,3 +2,5 @@
tests:
- test: 'direct'
- test: 'ice'
+ - test: 'rtcp_mux_offered'
+ - test: 'rtcp_mux_not_offered'
--
To view, visit https://gerrit.asterisk.org/9928
To unsubscribe, or for help writing mail filters, visit https://gerrit.asterisk.org/settings
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: I4d6f139c106add84819736c0c9e886d6cb10e1f0
Gerrit-Change-Number: 9928
Gerrit-PatchSet: 1
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-code-review/attachments/20180820/4c2f97be/attachment-0001.html>
More information about the asterisk-code-review
mailing list