Hi all,
I am doing Asterisks performance tests on a pico-itx box. I compiled SIPp
with pcap play support in a Debian Etch machine so I could send RTP packages
through a recorded pcap file for simulating traffic. I run SIPp in client
mode in one machine and I call to a VoIP phone (actually I wanted to call to
a SIPp in server mode but I am calling to a phone for debugging purposes).
The problem I have is that SIPp in client mode sends the RTP packages to
Asterisk (at least that is what SIPp shows in the "Total RTP pckts sent"),
but if I activate the RTP debug mode in Asterisk, I see that the
communication is only in one direction (communication is established but RTP
packages are only sent to the machine with SIPp client). SIP messages
between the boxes seems to arrive fine, and the communication after a while
closes without problem, telling the SIPp client that it was a Succesful
call. First I thought it could be Asterisk, but when I call from another
source (one more VoIP phone I have here) the communication is perfect and
RTP packages are running in both directions. So I think it can be SIPp. I
have used my own xml scenario based on uac_pcap.xml (it has little changes),
but even if I run the original uac_pcap, it doesn't work still. Anyway I add
my uac_pcap.xml:
<?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" start_rtd="1">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@
[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <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 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
]]>
</send>
<recv response="100" optional="true" rtd="1" start_rtd="2">
</recv>
<recv response="180" optional="true" rtd="2">
</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. -->
<recv response="200" crlf="true">
</recv>
<!-- 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: sipp <sip:sipp@
[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <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>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/call_g711a.pcap"/>
</action>
</nop>
<!-- Pause 3 minutes, which is approximately the duration of the -->
<!-- PCAP file -->
<pause milliseconds="180000"/>
<!-- Play an out of band DTMF '1' -->
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
</action>
</nop>
<pause milliseconds="1000"/>
<!-- 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: sipp <sip:sipp@
[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <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>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="50, 100, 200, 500, 1100, 2100, 3100, 4100,
5100, 6100, 10000"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
I am quite crazy about the whole problem, it makes me mad that it works
using two phones and that it doesn't when I generate the call with SIPp.
Maybe I have to add something to the SIP messages in the script that I am
not realizing or I don't know...
Thanks, greetings,
Jesús
PS. I am not using NAT, all the machines and phones are in the same private
network.
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