mirror of
https://github.com/jambonz/sbc-outbound.git
synced 2026-10-09 21:14:05 +00:00
156 lines
4.3 KiB
XML
156 lines
4.3 KiB
XML
<?xml version="1.0" encoding="ISO-8859-1" ?>
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<!DOCTYPE scenario SYSTEM "sipp.dtd">
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<!-- This program is free software; you can redistribute it and/or -->
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<!-- modify it under the terms of the GNU General Public License as -->
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<!-- published by the Free Software Foundation; either version 2 of the -->
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<!-- License, or (at your option) any later version. -->
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<!-- -->
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<!-- This program is distributed in the hope that it will be useful, -->
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<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
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<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
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<!-- GNU General Public License for more details. -->
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<!-- -->
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<!-- You should have received a copy of the GNU General Public License -->
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<!-- along with this program; if not, write to the -->
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<!-- Free Software Foundation, Inc., -->
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<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
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<!-- -->
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<!-- Sipp default 'uas' scenario. -->
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<!-- -->
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<scenario name="Basic UAS responder">
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<!-- By adding rrs="true" (Record Route Sets), the route sets -->
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<!-- are saved and used for following messages sent. Useful to test -->
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<!-- against stateful SIP proxies/B2BUAs. -->
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<recv request="INVITE" crlf="true">
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</recv>
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<send retrans="500">
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<![CDATA[
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SIP/2.0 401 Unauthorized
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[last_Via:]
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[last_From:]
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[last_To:];tag=[pid]SIPpTag01[call_number]
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[last_Call-ID:]
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[last_CSeq:]
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WWW-Authenticate: Digest realm="jambonz.org", nonce="4cdbb733645816512687270b83d2ae5d11e4d9d8"
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Content-Length: 0
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]]>
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</send>
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<recv request="ACK"
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rtd="true"
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crlf="true">
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</recv>
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<recv request="INVITE" crlf="true">
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<action>
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<verifyauth assign_to="authvalid" username="foo" password="bar" />
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</action>
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</recv>
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<nop hide="true" test="authvalid" next="goodauth" />
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<send retrans="500">
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<![CDATA[
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SIP/2.0 403 Forbidden
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[last_Via:]
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[last_From:]
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[last_To:];tag=[pid]SIPpTag01[call_number]
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[last_Call-ID:]
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[last_CSeq:]
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Contact: <sip:[local_ip]:[local_port];transport=[transport]>;expires=60
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Content-Type: application/sdp
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Content-Length: 0
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]]>
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</send>
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<recv request="ACK" next="endit"
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rtd="true"
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crlf="true">
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</recv>
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<label id="goodauth"/>
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<send>
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<![CDATA[
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SIP/2.0 100 Trying
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[last_Via:]
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[last_From:]
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[last_To:]
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[last_Call-ID:]
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[last_CSeq:]
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Content-Length: 0
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]]>
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</send>
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<send retrans="500">
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<![CDATA[
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SIP/2.0 200 OK
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[last_Via:]
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[last_From:]
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[last_To:];tag=[pid]SIPpTag01[call_number]
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[last_Call-ID:]
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[last_CSeq:]
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Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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Content-Type: application/sdp
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Content-Length: [len]
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v=0
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o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
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s=-
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c=IN IP[media_ip_type] [media_ip]
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t=0 0
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m=audio [media_port] RTP/AVP 0
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a=rtpmap:0 PCMU/8000
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]]>
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</send>
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<recv request="ACK"
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optional="true"
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rtd="true"
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crlf="true">
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</recv>
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<recv request="BYE">
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</recv>
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<send>
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<![CDATA[
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SIP/2.0 200 OK
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[last_Via:]
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[last_From:]
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[last_To:]
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[last_Call-ID:]
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[last_CSeq:]
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Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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Content-Length: 0
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]]>
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</send>
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<label id="endit"/>
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<!-- Keep the call open for a while in case the 200 is lost to be -->
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<!-- able to retransmit it if we receive the BYE again. -->
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<timewait milliseconds="4000"/>
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<!-- definition of the response time repartition table (unit is ms) -->
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<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
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<!-- definition of the call length repartition table (unit is ms) -->
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<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
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</scenario>
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