Hi All,
I'm running into a problem sending out of band DTMF signals using pcap_play. I
tried sending the stock dtmf_2833_X.pcap files packaged with sipp. and from
what I read online, I should be be able to send them with no issues using
<exec play_pcap_audio="/usr/share/sipp/pcap/g711a.pcap"/>
But it seems as if the server isn't receiving the packets correctly or
something. I've tried all sorts of settings, to get a different result and
haven't been able to get it to work regardless of what I try.
This is the scenario file I'm using: Sipp registers with our OpenSips server,
authenticates, then sends an invite to conference number. Once the invite is
accepted, it sends the digits and hangs up. I need to do this to test
conference functionality, and at this stage, just stuck in not being able to
see the DTMF coming in on the OpenSips server. I've also tested it by calling
my cell phone using this scenario, and I don't hear any digits (although I
believe I wouldn't be able to hear the digits anyway)
Is there any other way I can send the digits? or is there something I'm doing
wrong? If you need more info, please let me know, I'd be glad to help with
troubleshooting as much as possible.
Oh, and i'm running the test using the following command:
sipp -r 5 -sf "Scenario_file_path" -m 1 -inf "parameters_file_path"
164.177.159.216 (parameters file contains things like authentication info and
destination number to call)
<!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 default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- 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[
REGISTER sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [field3]:[field4];rport;alias
To: sip:[field0]@[field1]
From: sip:[field0]@[field1];tag=[pid]SIPpTag00[call_number]
Contact: sip:[field0]@[field3]:[field4]
Max-Forwards: 70
Expires: 300
Call-ID: [call_id]
CSeq: 1 REGISTER
Content-Length: 0
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send>
<![CDATA[
ACK sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [field3]:[field4]
From: sip:[field0]@[field1];tag=[pid]SIPpTag00[call_number]
To: sip:[field0]@[field1];tag=[pid]SIPpTag00[call_number]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[field0]@[field3]:[field4]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
REGISTER sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [field3]:[field4];rport;alias
To: sip:[field0]@[field1]
From: sip:[field0]@[field1];tag=[pid]SIPpTag00[call_number]
Contact: sip:[field0]@[field3]:[field4]
[field2]
Expires: 300
Max-Forwards: 70
Call-ID: [call_id]
CSeq: 2 REGISTER
Content-Length: 0
]]>
</send>
<recv response="100"
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="1">
</recv>
<send start_rtd="1">
<![CDATA[
INVITE sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [field3]:[field4];branch=[branch]
From: sip:[field0]@[field1];tag=[pid]SIPpTag00[call_number]
To: sip:[field5]
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
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 0 9 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]]>
</send>
<recv response="100">
</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" rrs="true" >
</recv>
<send>
<![CDATA[
ACK sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [field3]:[field4]
From: sip:[field0]@[field1];tag=[pid]SIPpTag00[call_number]
<!-- From: sipp
<sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] -->
To: sip:[field5]
Call-ID: INV///[call_id]
CSeq: 1 ACK
Contact: sip:[field0]@[field3]:[field4]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<nop>
<action>
<exec play_pcap_audio="/usr/share/sipp/pcap/DTMF_6.pcap" />
</action>
</nop>
<pause milliseconds="1000"/>
<nop>
<action>
<exec play_pcap_audio="/usr/share/sipp/pcap/dtmf_2833_9.pcap" />
</action>
</nop>
<pause milliseconds="1000"/>
<nop>
<action>
<exec play_pcap_audio="/usr/share/sipp/pcap/dtmf_2833_7.pcap" />
</action>
</nop>
<pause milliseconds="1000"/>
<nop>
<action>
<exec play_pcap_audio="/usr/share/sipp/pcap/dtmf_2833_7.pcap" />
</action>
</nop>
<pause milliseconds="1000"/>
<nop>
<action>
<exec play_pcap_audio="/usr/share/sipp/pcap/dtmf_2833_2.pcap" />
</action>
</nop>
<pause milliseconds="1000"/>
<nop>
<action>
<exec play_pcap_audio="/usr/share/sipp/pcap/dtmf_2833_5.pcap" />
</action>
</nop>
<pause milliseconds="1000"/>
<nop>
<action>
<exec play_pcap_audio="/usr/share/sipp/pcap/dtmf_2833_1.pcap" />
</action>
</nop>
<pause milliseconds="1000"/>
<nop>
<action>
<exec
play_pcap_audio="/usr/share/sipp/pcap/dtmf_2833_pound.pcap" />
</action>
</nop>
<pause milliseconds="1000"/>
<send rtd="1">
<![CDATA[
ACK sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [field3]:[field4];branch=[branch]
From: sip:[field0]@[field1];tag=[pid]SIPpTag00[call_number]
<!-- From: sipp
<sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] -->
To: sip:[field5];[peer_tag_param]
Call-ID: INV///[call_id]
CSeq: 1 ACK
Contact: sip:[field0]@[field3]:[field4]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200">
</recv>
<send rtd="1">
<![CDATA[
ACK sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [field3]:[field4];branch=[branch]
From: sip:[field0]@[field1];tag=[pid]SIPpTag00[call_number]
<!-- From: sipp
<sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] -->
To: sip:[field5];[peer_tag_param]
Call-ID: INV///[call_id]
CSeq: 1 ACK
Contact: sip:[field0]@[field3]:[field4]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500" start_rtd="2">
<![CDATA[
BYE sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [field3]:[field4];branch=[branch]
From:
<sip:[field0]@[remote_ip]:[remote_port]>;tag=[pid]SIPpTag00[call_number]
<!-- From: sipp
<sip:[field0]@[local_ip]:[lo;tag=[pid]SIPpTag00[call_number] -->
To: sip:[field5];[peer_tag_param]
Call-ID: INV///[call_id]
CSeq: 2 BYE
Contact: sip:[field0]@[field3]:[field4]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true" rtd="2">
</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>
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