Pavel,
Thank you very much. I'm close to the solution now.
But, I'm going to ask a new question in the forum. I hope you can help me...
Thank you.
From: Šindelka Pavel [mailto:sinde...@ttc.cz]
Sent: terça-feira, 28 de novembro de 2017 17:12
To: gdcarvalh...@gmail.com
Cc: sipp-users@lists.sourceforge.net
Subject: Re: [Sipp-users] Help
At a quick overview on a smartphone, two points:
1) the To: tag in your ACK is constant so it cannot mirror the value assigned
by the opposite party. Hence the opposite party cannot map the message to the
dialog, so it repeats the 200 as required when no ACK is received.
2) if you want SIPp to send RTP, you have to put a corresponding statement to
your scenario, it does not happen automatically.
Pavel
Dne 28. 11. 2017 19:51 napsal uživatel gdcarvalh...@gmail.com
<mailto:gdcarvalh...@gmail.com> :
Folks,
I'm making a test on a IVR Machine from a softphone installed on my workstation
(10.0.4.147 = my workstation / 10.0.8.67 = IVR). Here is the result of
Wireshark:
I did the same test from SIPP to IVR and I used the scenario below:
<?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>
<![CDATA[
INVITE sip:12053@10.0.8.67 SIP/2.0
Via: SIP/2.0/UDP 10.0.4.147:5061;rport;branch=z9hGbk2426932
To: <sip:12053@10.0.8.67>
From: <sip:11971321122@10.0.4.147:5071>;tag=9519
Call-ID: 1511874074-26932-NVT4147@10.0.4.147
<mailto:1511874074-26932-NVT4147@10.0.4.147>
CSeq: 867 INVITE
Max-Forwards: 70
User-Agent: NCH Software Express Talk 4.28
Contact: <sip:11971321122@10.0.4.147:5061>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 328
v=0
o=NCHSoftware-Talk 1511874065 1511874074 IN IP4 10.0.4.147
s=Express Talk Call
c=IN IP4 10.0.4.147
t=0 0
m=audio 8000 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a:sendrecv
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="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. -->
<recv response="200" rtd="true" 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:12053@10.0.8.67:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.4.147:5061;rport;branch=z9hGbk2526932
From: <sip:11971321122@10.0.4.147:5071>;tag=9519
To: <sip:12053@10.0.8.67>;tag=7831-C233
Call-ID: 1511874074-26932-NVT4147@10.0.4.147
<mailto:1511874074-26932-NVT4147@10.0.4.147>
CSeq: 867 ACK
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.28
Content-Length: 0
]]>
</send>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send>
<![CDATA[
BYE sip:12053@10.0.8.67:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.4.147:5071;branch=z9hGbk2626932
From: <sip:11971321122@10.0.4.147:5071>;tag=9519
To: <sip:12053@10.0.8.67>;tag=7831-C233
Call-ID: 1511874074-26932-NVT4147@10.0.4.147
<mailto:1511874074-26932-NVT4147@10.0.4.147>
CSeq: 868 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.28
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- 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>
The result of this second test is just below:
The problem is that when I make the call through the softphone, the ura is
sending the RTP to me soon after the 183 (as print below). This is correct
behavior.
But when I am making the call through SIPP the same is not occurring and I get
the BYE upon receipt of several 200.
Is there anything wrong with this scenario? Please help me.
Thanks in advance for your help and attention.
Best regards.
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