mirror of
https://github.com/jambonz/sbc-outbound.git
synced 2026-10-09 21:14:05 +00:00
Hoan Luu HuuandDave Horton
76558cc5c9
* add protocol for outbound call * add protocol for outbound call * update db-helper version * lcr fixes * fix cancel test --------- Co-authored-by: Dave Horton <daveh@beachdognet.com>
107 lines
3.8 KiB
XML
107 lines
3.8 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 '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" start_txn="invite">
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<![CDATA[
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INVITE sip:16173333456@127.0.0.1 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]SIPpTag09[call_number]
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To: <sip:16173333456@127.0.0.1>
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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: uac-cancel
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X-Account-Sid: ed649e33-e771-403a-8c99-1780eabbc803
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X-Call-Sid: fff49e33-e771-403a-8c99-1780eabbc803
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X-Jambonz-Routing: phone
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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[local_ip_type] [local_ip]
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t=0 0
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m=audio [auto_media_port] RTP/AVP 8 101
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a=rtpmap:8 PCMA/8000
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a=rtpmap:101 telephone-event/8000
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a=fmtp:101 0-11,16
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]]>
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<action>
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<ereg regexp=".*" search_in="hdr" header="Via:" assign_to="1" />
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<log message="Via is [last_Via]. read it as [$1]"/>
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</action>
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</send>
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<recv response="100" response_txn="invite"/>
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<pause milliseconds="1000"/>
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<send>
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<![CDATA[
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CANCEL sip:[service]@[remote_ip]:[remote_port] SIP/2.0
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Via: [$1]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
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To: [service] <sip:[service]@[remote_ip]:[remote_port]>
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Call-ID: [call_id]
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CSeq: 1 CANCEL
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Content-Length: 0
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]]>
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</send>
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<recv response="200" optional="true"/>
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<recv response="487" response_txn="invite">
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</recv>
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<send ack_txn="invite">
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<![CDATA[
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ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
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Via: [$1]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
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To: [service] <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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<!-- 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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