<?xml version="1.0" encoding="ISO-8859-1" ?>
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<!--
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-- Copyright (c) 2010-2019 Belledonne Communications SARL.
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--
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-- This file is part of Liblinphone.
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--
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-- This program is free software: you can redistribute it and/or modify
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-- it under the terms of the GNU General Public License as published by
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-- the Free Software Foundation, either version 3 of the License, or
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-- (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, see <http://www.gnu.org/licenses/>.
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-->
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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 'uac' scenario. -->
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<!-- -->
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<scenario name="Basic Sipstone UAC">
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<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
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<!-- generated by sipp. To do so, use [call_id] keyword. -->
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<send retrans="500">
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<![CDATA[
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INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
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To: sut <sip:[service]@[remote_ip]:[remote_port]>
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Call-ID: [call_id]
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CSeq: 1 INVITE
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Contact: sip:sipp@[local_ip]:[local_port]
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Max-Forwards: 70
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Subject: Performance Test
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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[media_ip_type] [media_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 96 97 0 8 101 98
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a=rtpmap:96 speex/16000
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a=fmtp:96 vbr=on
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a=rtpmap:97 speex/8000
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a=fmtp:97 vbr=on
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a=rtpmap:101 telephone-event/16000
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a=rtpmap:98 telephone-event/8000
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m=video [media_port+2] RTP/AVP 96
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a=rtpmap:96 VP8/90000
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]]>
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</send>
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<recv response="100"
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optional="true">
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</recv>
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<recv response="180" optional="true">
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</recv>
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<recv response="183" optional="true">
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</recv>
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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 response="200" rtd="true">
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</recv>
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<!-- Packet lost can be simulated in any send/recv message by -->
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<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
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<send>
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<![CDATA[
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ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
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To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
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Call-ID: [call_id]
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CSeq: 1 ACK
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Contact: sip:sipp@[local_ip]:[local_port]
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Max-Forwards: 70
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Subject: Performance Test
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Content-Length: 0
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]]>
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</send>
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<!-- This delay can be customized by the -d command-line option -->
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<!-- or by adding a 'milliseconds = "value"' option here. -->
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<pause/>
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<!-- The 'crlf' option inserts a blank line in the statistics report. -->
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<send retrans="500">
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<![CDATA[
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BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
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To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
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Call-ID: [call_id]
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CSeq: 2 BYE
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Contact: sip:sipp@[local_ip]:[local_port]
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Max-Forwards: 70
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Subject: Performance Test
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Content-Length: 0
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]]>
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</send>
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<recv response="200" crlf="true">
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</recv>
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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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