[Asterisk-code-review] tests/pjsip/hold: Add test checking codec order upon hold/unhold (testsuite[master])

Friendly Automation asteriskteam at digium.com
Wed Jan 5 12:38:48 CST 2022


Friendly Automation has submitted this change. ( https://gerrit.asterisk.org/c/testsuite/+/17732 )

Change subject: tests/pjsip/hold: Add test checking codec order upon hold/unhold
......................................................................

tests/pjsip/hold: Add test checking codec order upon hold/unhold

With the fix of ASTERISK-29320 the order of the codecs in the incoming
SDP is preserved. To check this a testcase is added, performing a
hold/unhold operation on an outgoing channel.

Change-Id: Id7249b174e5630b901d303a19cdd808651e455ea
---
M tests/channels/pjsip/hold/configs/ast1/pjsip.conf
M tests/channels/pjsip/hold/run-test
A tests/channels/pjsip/hold/sipp/phone_B_codec_order.xml
M tests/channels/pjsip/hold_inactive/test-config.yaml
4 files changed, 299 insertions(+), 1 deletion(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved
  Friendly Automation: Approved for Submit



diff --git a/tests/channels/pjsip/hold/configs/ast1/pjsip.conf b/tests/channels/pjsip/hold/configs/ast1/pjsip.conf
index 65519b0..62f2cae 100644
--- a/tests/channels/pjsip/hold/configs/ast1/pjsip.conf
+++ b/tests/channels/pjsip/hold/configs/ast1/pjsip.conf
@@ -24,5 +24,5 @@
 aors=phone_B
 context=default
 disallow=all
-allow=ulaw
+allow=ulaw,alaw,gsm
 direct_media=no
diff --git a/tests/channels/pjsip/hold/run-test b/tests/channels/pjsip/hold/run-test
index 9b32ffc..086ec53 100755
--- a/tests/channels/pjsip/hold/run-test
+++ b/tests/channels/pjsip/hold/run-test
@@ -45,6 +45,9 @@
                                   '-inf': INJECT_FILE},
                                  {'scenario': 'phone_A.xml',
                                   '-i': '127.0.0.2', '-p': '5060',
+                                  '-inf': INJECT_FILE},
+                                  {'scenario': 'phone_A.xml',
+                                  '-i': '127.0.0.2', '-p': '5060',
                                   '-inf': INJECT_FILE}]
         self.sipp_scn_phone_b = [{'scenario': 'phone_B_media_restrict.xml',
                                   '-i': '127.0.0.3', '-p': '5060',
@@ -63,6 +66,9 @@
                                   '-inf': INJECT_FILE},
                                  {'scenario': 'phone_B_IP_media_restrict.xml',
                                   '-i': '127.0.0.3', '-p': '5060',
+                                  '-inf': INJECT_FILE},
+                                  {'scenario': 'phone_B_codec_order.xml',
+                                  '-i': '127.0.0.3', '-p': '5060',
                                   '-inf': INJECT_FILE}]
 
         self.reactor_timeout = 60
diff --git a/tests/channels/pjsip/hold/sipp/phone_B_codec_order.xml b/tests/channels/pjsip/hold/sipp/phone_B_codec_order.xml
new file mode 100644
index 0000000..eb55b58
--- /dev/null
+++ b/tests/channels/pjsip/hold/sipp/phone_B_codec_order.xml
@@ -0,0 +1,291 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B Hold">
+	<Global variables="global_call_id"/>
+	<Global variables="prime_tag"/>
+
+	<recv request="INVITE" crlf="true">
+		<action>
+			<ereg regexp=".*"
+				header="Call-ID:"
+				search_in="hdr"
+				check_it="true"
+				assign_to="global_call_id"/>
+			<ereg regexp="tag=.*"
+				header="From:"
+				search_in="hdr"
+				check_it="true"
+				assign_to="prime_tag"/>
+		</action>
+	</recv>
+
+	<send>
+		<![CDATA[
+			SIP/2.0 100 Trying
+			[last_Via:]
+			[last_From:]
+			[last_To:];tag=[call_number]
+			[last_Call-ID:]
+			[last_CSeq:]
+			Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Content-Length: 0
+		]]>
+	</send>
+
+	<send>
+		<![CDATA[
+			SIP/2.0 180 Ringing
+			[last_Via:]
+			[last_From:]
+			[last_To:];tag=[call_number]
+			[last_Call-ID:]
+			[last_CSeq:]
+			Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Allow-Events: talk,hold,conference
+			Accept-Language: en
+			Content-Length: 0
+		]]>
+	</send>
+
+	<pause milliseconds="200"/>
+
+	<send retrans="500">
+		<![CDATA[
+			SIP/2.0 200 OK
+			[last_Via:]
+			[last_From:]
+			[last_To:];tag=[call_number]
+			[last_Call-ID:]
+			[last_CSeq:]
+			Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			Supported: 100rel,replaces
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=- 1325003603 1325003604 IN IP4 [local_ip]
+			s=Polycom IP Phone
+			c=IN IP4 [local_ip]
+			t=0 0
+			a=sendrecv
+			m=audio 2226 RTP/AVP 8 3 101
+			a=sendrecv
+			a=rtpmap:8 PCMA/8000
+			a=rtpmap:3 GSM/8000
+			a=rtpmap:101 telephone-event/8000
+		]]>
+	</send>
+
+	<!-- RECV ACK -->
+	<recv request="ACK"/>
+
+	<!-- Wait some period of time -->
+	<pause milliseconds="2000"/>
+
+	<!-- Modify RTP session to be send only -->
+	<send retrans="500">
+		<![CDATA[
+			INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+			To: [field0] <sip:[field1]@[remote_ip]>;[$prime_tag]
+			CSeq: [cseq] INVITE
+			Call-ID: [$global_call_id]
+			Contact: <sip:[field1]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Supported: 100rel,replaces
+			Allow-Events: talk,hold,conference
+			Max-Forwards: 70
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=- 1325003603 1325003604 IN IP4 [local_ip]
+			s=Polycom IP Phone
+			c=IN IP4 [local_ip]
+			t=0 0
+			a=sendonly
+			m=audio 2226 RTP/AVP 8 3 101
+			a=sendonly
+			a=rtpmap:8 PCMA/8000
+			a=rtpmap:3 GSM/8000
+			a=rtpmap:101 telephone-event/8000
+		]]>
+	</send>
+
+	<recv response="100" optional="true" />
+
+	<!-- Check that a-law has been selected after the reinvite -->
+	<recv response="200" rtd="true">
+		<action>
+			<ereg regexp="m=audio [0-9]{1,5} RTP/AVP 8 .*"
+					search_in="body" check_it="true" assign_to="1"/>
+			<test assign_to="1" variable="1" compare="equal" value=""/>
+		</action>
+	</recv>
+
+	<pause milliseconds="200"/>
+
+	<send>
+		<![CDATA[
+			ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+			To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+			CSeq: [cseq] ACK
+			Call-ID: [$global_call_id]
+			Contact: <sip:[field1]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Max-Forwards: 70
+			Content-Length: 0
+		]]>
+	</send>
+
+	<!-- Wait some period of time -->
+	<pause milliseconds="2000"/>
+
+	<!-- Modify RTP session to be send only -->
+	<send retrans="500">
+		<![CDATA[
+			UPDATE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+			To: [field0] <sip:[field1]@[remote_ip]>;[$prime_tag]
+			CSeq: [cseq] UPDATE
+			Call-ID: [$global_call_id]
+			Contact: <sip:[field1]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Supported: 100rel,replaces
+			Allow-Events: talk,hold,conference
+			Max-Forwards: 70
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=- 1325003603 1325003604 IN IP4 [local_ip]
+			s=Polycom IP Phone
+			c=IN IP4 [local_ip]
+			t=0 0
+			a=sendonly
+			m=audio 2226 RTP/AVP 8 3 101
+			a=sendonly
+			a=rtpmap:8 PCMA/8000
+			a=rtpmap:3 GSM/8000
+			a=rtpmap:101 telephone-event/8000
+		]]>
+	</send>
+
+	<recv response="100" optional="true" />
+
+	<!-- Check that a-law has been selected after the update -->
+	<recv response="200" rtd="true">
+		<action>
+			<ereg regexp="m=audio [0-9]{1,5} RTP/AVP 8 .*"
+				search_in="body" check_it="true" assign_to="1"/>
+			<test assign_to="1" variable="1" compare="equal" value=""/>
+		</action>
+	</recv>
+
+	<pause milliseconds="200"/>
+
+	<send>
+		<![CDATA[
+			ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+			To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+			CSeq: [cseq] ACK
+			Call-ID: [$global_call_id]
+			Contact: <sip:[field1]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Max-Forwards: 70
+			Content-Length: 0
+		]]>
+	</send>
+
+	<!-- Wait some period of time, then send the un-hold as reinvite without SDP -->
+	<pause milliseconds="2000"/>
+
+	<send retrans="500">
+		<![CDATA[
+			INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+			To: [field0] <sip:[field1]@[remote_ip]>;[$prime_tag]
+			CSeq: [cseq] INVITE
+			Call-ID: [$global_call_id]
+			Contact: <sip:[field1]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Supported: 100rel,replaces
+			Allow-Events: talk,hold,conference
+			Max-Forwards: 70
+			Content-Length: 0
+		]]>
+	</send>
+
+	<recv response="100" optional="true" />
+
+	<!-- Check that a-law has been selected after the reinvite -->
+	<recv response="200" rtd="true">
+		<action>
+			<ereg regexp="m=audio [0-9]{1,5} RTP/AVP 8 .*"
+				search_in="body" check_it="true" assign_to="1"/>
+			<test assign_to="1" variable="1" compare="equal" value=""/>
+		</action>
+	  </recv>
+
+	<send>
+		<![CDATA[
+			ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+			To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+			CSeq: [cseq] ACK
+			Call-ID: [$global_call_id]
+			Contact: <sip:[field1]@[local_ip]:[local_port]>
+			Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Max-Forwards: 70
+			Content-Length: 0
+		]]>
+	</send>
+
+	<!-- Wait some period of time -->
+	<pause milliseconds="2000"/>
+
+	<send>
+		<![CDATA[
+			BYE sip:[field1]@1[remote_ip]:[remote_port] SIP/2.0
+			Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
+			From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+			To: [field0] <sip:[field1]@[remote_ip]>[peer_tag_param]
+			CSeq: [cseq] BYE
+			Call-ID: [$global_call_id]
+			Contact: <sip:[field1]@[local_ip]:[local_port]>
+			User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+			Accept-Language: en
+			Max-Forwards: 70
+			Content-Length: 0
+		]]>
+	</send>
+
+
+</scenario>
diff --git a/tests/channels/pjsip/hold_inactive/test-config.yaml b/tests/channels/pjsip/hold_inactive/test-config.yaml
index 5a7c18c..790aa8c 100644
--- a/tests/channels/pjsip/hold_inactive/test-config.yaml
+++ b/tests/channels/pjsip/hold_inactive/test-config.yaml
@@ -5,6 +5,7 @@
         sending a re-INVITE with a modified SDP containing a restricted audio
         direction with and without an IP address of 0.0.0.0. Restricted audio
         direction is also tested for unholding by a re-INVITE without an SDP.
+        A scenario checks that the correct codec is selected after unholding.
 
 
 properties:

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Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Change-Id: Id7249b174e5630b901d303a19cdd808651e455ea
Gerrit-Change-Number: 17732
Gerrit-PatchSet: 2
Gerrit-Owner: Florentin Mayer <f.mayer at commend.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-MessageType: merged
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