[svn-commits] mmichelson: testsuite/asterisk/trunk r164 - in /asterisk/trunk/tests/sip_chan...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Mar 31 10:04:40 CDT 2010
Author: mmichelson
Date: Wed Mar 31 10:04:37 2010
New Revision: 164
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=164
Log:
Add unit test for testing SIP parameters retrieved from the CHANNEL dialplan function.
Review: https://reviewboard.asterisk.org/r/589
Added:
asterisk/trunk/tests/sip_channel_params/
asterisk/trunk/tests/sip_channel_params/configs/
asterisk/trunk/tests/sip_channel_params/configs/extensions.conf (with props)
asterisk/trunk/tests/sip_channel_params/configs/rtp.conf (with props)
asterisk/trunk/tests/sip_channel_params/configs/sip.conf (with props)
asterisk/trunk/tests/sip_channel_params/run-test (with props)
asterisk/trunk/tests/sip_channel_params/sipp/
asterisk/trunk/tests/sip_channel_params/sipp/call.xml (with props)
asterisk/trunk/tests/sip_channel_params/sipp/register.xml (with props)
asterisk/trunk/tests/sip_channel_params/test.lua (with props)
Added: asterisk/trunk/tests/sip_channel_params/configs/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_channel_params/configs/extensions.conf?view=auto&rev=164
==============================================================================
--- asterisk/trunk/tests/sip_channel_params/configs/extensions.conf (added)
+++ asterisk/trunk/tests/sip_channel_params/configs/extensions.conf Wed Mar 31 10:04:37 2010
@@ -1,0 +1,5 @@
+[channel_test]
+
+exten => test,1,Answer
+exten => test,n,Playback(demo-congrats)
+exten => test,n,Hangup
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Added: asterisk/trunk/tests/sip_channel_params/configs/rtp.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_channel_params/configs/rtp.conf?view=auto&rev=164
==============================================================================
--- asterisk/trunk/tests/sip_channel_params/configs/rtp.conf (added)
+++ asterisk/trunk/tests/sip_channel_params/configs/rtp.conf Wed Mar 31 10:04:37 2010
@@ -1,0 +1,3 @@
+[general]
+rtpstart=10000
+rtpend=10003
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Added: asterisk/trunk/tests/sip_channel_params/configs/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_channel_params/configs/sip.conf?view=auto&rev=164
==============================================================================
--- asterisk/trunk/tests/sip_channel_params/configs/sip.conf (added)
+++ asterisk/trunk/tests/sip_channel_params/configs/sip.conf Wed Mar 31 10:04:37 2010
@@ -1,0 +1,13 @@
+[general]
+udpbindaddr=127.0.0.1:5060
+canreinvite=no
+videosupport=yes
+
+[test1]
+type=friend
+host=dynamic
+dtmf=rfc2833
+context=channel_test
+disallow=all
+allow=ulaw
+allow=h263
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Added: asterisk/trunk/tests/sip_channel_params/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_channel_params/run-test?view=auto&rev=164
==============================================================================
--- asterisk/trunk/tests/sip_channel_params/run-test (added)
+++ asterisk/trunk/tests/sip_channel_params/run-test Wed Mar 31 10:04:37 2010
@@ -1,0 +1,3 @@
+#!/bin/bash -e
+
+asttest -a / -s tests/sip_channel_params
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Added: asterisk/trunk/tests/sip_channel_params/sipp/call.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_channel_params/sipp/call.xml?view=auto&rev=164
==============================================================================
--- asterisk/trunk/tests/sip_channel_params/sipp/call.xml (added)
+++ asterisk/trunk/tests/sip_channel_params/sipp/call.xml Wed Mar 31 10:04:37 2010
@@ -1,0 +1,95 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <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: test1 <sip:wienerschnitzel@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: Channel Param 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 6000 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ m=video 6002 RTP/AVP 34
+ a=rtpmap:34 H263/90000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ 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:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:wienerschnitzel@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- Pause long enough for our test to run -->
+ <pause milliseconds="4000" />
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:wienerschnitzel@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance 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>
+
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Added: asterisk/trunk/tests/sip_channel_params/sipp/register.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_channel_params/sipp/register.xml?view=auto&rev=164
==============================================================================
--- asterisk/trunk/tests/sip_channel_params/sipp/register.xml (added)
+++ asterisk/trunk/tests/sip_channel_params/sipp/register.xml Wed Mar 31 10:04:37 2010
@@ -1,0 +1,29 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<!--
+This scenario will execute a sip register with the given parameters.
+-->
+
+
+<scenario name="Register">
+ <send retrans="500">
+ <![CDATA[
+
+ REGISTER sip:test1@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:test1@[local_ip]:[local_port]>;tag=[call_number]
+ To: test1 <sip:test1@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 REGISTER
+ Contact: sip:test1@[local_ip]:[local_port]
+ Content-Length: 0
+ Expires: 120
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+ <recv response="200"/>
+
+</scenario>
+
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Added: asterisk/trunk/tests/sip_channel_params/test.lua
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_channel_params/test.lua?view=auto&rev=164
==============================================================================
--- asterisk/trunk/tests/sip_channel_params/test.lua (added)
+++ asterisk/trunk/tests/sip_channel_params/test.lua Wed Mar 31 10:04:37 2010
@@ -1,0 +1,144 @@
+function sipp_exec(scenario, name, local_port)
+ return proc.exec_io("sipp",
+ "127.0.0.1",
+ "-m", "1",
+ "-sf", scenario,
+ "-i", "127.0.0.1",
+ "-p", local_port,
+ "-timeout", "30",
+ "-trace_err"
+ )
+end
+
+function sipp_exec_and_wait(scenario, name, local_port)
+ return sipp_check_error(sipp_exec(scenario, name, local_port), scenario)
+end
+
+function sipp_check_error(p, scenario)
+ local res, err = p:wait()
+
+ if not res then error(err) end
+ if res ~= 0 then
+ error("error while executing " .. scenario .. " sipp scenario (sipp exited with status " .. res .. ")\n" .. p.stderr:read("*a"))
+ end
+
+ return res, err
+end
+
+function manager_setup(a)
+ m,err = a:manager_connect()
+ if not m then
+ fail("error connecting to asterisk: " .. err)
+ end
+
+ local login = ast.manager.action:new("login")
+ login["Username"] = "asttest"
+ login["Secret"] = "asttest"
+
+ local r = m(login)
+ if not r then
+ fail("error logging in to the manager: " .. err)
+ end
+
+ if r["Response"] ~= "Success" then
+ fail("error authenticating: " .. r["Message"])
+ end
+end
+
+function findpattern(text, pattern, start)
+ return string.sub(text, string.find(text, pattern, start))
+end
+
+function check_sip_params(sip_params, chan_name)
+ for key,value in pairs(sip_params) do
+ local variable = "CHANNEL(" .. key .. ")"
+ local getvar = ast.manager.action:new("GetVar")
+ getvar["Channel"] = chan_name
+ getvar["Variable"] = variable
+ local response, err = m(getvar)
+ if not response then
+ fail("error getting variable " .. key .. ": " .. err)
+ break
+ end
+ if response["Response"] ~= "Success" then
+ fail("Response failure: " .. response["Message"])
+ end
+ if response["Variable"] ~= variable then
+ fail("Wrong variable. Expected " .. variable .. " but got " .. response["Variable"])
+ end
+ --Helpful for debugging in case this test starts failing
+ --print ("Got value " .. response["Value"] .. " for variable " .. response["Variable"])
+ if not findpattern(response["Value"], value) then
+ fail("Expected " .. value .. " for variable " .. response["Variable"] .. " but got " .. response["Value"])
+ end
+ end
+end
+
+function get_channel_name()
+ local chan_name = nil
+ local function set_name(event)
+ chan_name = event["Channel"]
+ end
+ m:register_event("Newchannel", set_name)
+ local res, err = m:pump_messages()
+ if not res then
+ fail("Error pumping messages: " .. err)
+ end
+ m:process_events()
+ m:unregister_event("Newchannel", set_name)
+ return chan_name
+end
+
+function test_call(scenario, name, local_port, sip_params)
+ local t1 = sipp_exec(scenario, name, local_port)
+ local chan_name = nil
+
+ posix.sleep(1)
+ chan_name = get_channel_name()
+ if not chan_name then
+ fail("Failed to get the name of the channel\n");
+ else
+ print("Got channel name " .. chan_name)
+ end
+ check_sip_params(sip_params, chan_name)
+ -- We have what we want. Let the scenario conclude
+ sipp_check_error(t1, scenario)
+end
+
+function call_and_check_sip_params(name)
+ local a = ast.new()
+ local sip_params = {
+ ["peerip"] = "127.0.0.1",
+ ["recvip"] = "127.0.0.1",
+ ["from"] = "sip:wienerschnitzel at 127.0.0.1:5061",
+ ["uri"] = "sip:kartoffelsalat at 127.0.0.1:5061",
+ ["useragent"] = "Channel Param Test",
+ ["peername"] = "test1",
+ ["t38passthrough"] = "0",
+ ["rtpdest,audio"] = "127.0.0.1:6000",
+ ["rtpdest,video"] = "127.0.0.1:6002",
+ ["rtpdest,text"] = "",
+ --We have no way to force a specific source
+ --RTP port to be used; we can specify only a
+ --range. Thus, our rtp.conf file limits the
+ --audio and video RTP ports to a selection of
+ --either 10000 or 10002.
+ ["rtpsource,audio"] = "127.0.0.1:1000[02]",
+ ["rtpsource,video"] = "127.0.0.1:1000[02]",
+ ["rtpsource,text"] = ""
+ }
+
+ a:load_config("configs/sip.conf")
+ a:load_config("configs/extensions.conf")
+ a:load_config("configs/rtp.conf")
+ a:generate_manager_conf()
+ a:spawn()
+
+ manager_setup(a)
+ --register our peer
+ sipp_exec_and_wait("sipp/register.xml", name, "5061")
+
+ test_call("sipp/call.xml", name, "5061", sip_params)
+end
+
+call_and_check_sip_params("test1")
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