Hi everyone,
when i use the sipp send a call to softphone, i can not listen the voice
in phone.I have try many times,and feel very terrible. anyone can help me.
Thanks in advance!
The following is xml files:and the pcap files in attachment.
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword.
-->
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport]
[local_ip]:[local_port];branch=[branch]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: [field1] <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance 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[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="407" auth="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport]
[local_ip]:[local_port];branch=[branch]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: [field1]
<sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: [field1] <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: sip:[EMAIL PROTECTED]:[local_port]
[field2]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
t=0 0
c=IN IP[media_ip_type] [media_ip]
m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 PCMU/8000
]]>
</send>
<recv response="100" optional="true"/>
<recv response="180" optional="true"/>
<recv response="200" rtd="true" crlf="true"/>
<send >
<![CDATA[
ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport]
[local_ip]:[local_port];branch=[branch]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: [field1]
<sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- Play an pre-recorde PCAP file (RTP stream) for testing tone -->
<pause milliseconds="60000"/>
<nop>
<action>
<exec play_pcap_audio="pcap/g711u/g711u_01.pcap"/>
</action>
</nop>
<pause milliseconds="60000"/>
<pause milliseconds="60000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: [field1] <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<send>
<![CDATA[
ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport]
[local_ip]:[local_port];branch=[branch]
From: [field0]
<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: [field1]
<sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<pause milliseconds="2000"/>
<!-- 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>
--
Best regards!
jordan pan
Location:Shenzhen China Company:www.justcall.cn
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