Call hold I tried, the call was holding also I have the script for call
transfer (downloaded) as well which I have not used, please try that and
moreover all the test I did so far with Asterisk server. Attached is the
files
On Mon, May 30, 2011 at 3:41 PM, Praveen Raj Swaminathan <
praveenraj_...@infosys.com> wrote:
> Hi All,
>
> Please let me know if any body tried call forwarding, call waiting
> scenarios successfully using open-ims, sipp etc., May I know if 3pcc is
> required to invoke these services.
>
> Also if you have sample xml input file for call forwarding/call waiting
> please attach with your reply.
>
> Thanks in Advance.
>
> Regards,
> Praveen Raj.
>
> ________________________________
> From: mayamatakeshi [mayamatake...@gmail.com]
> Sent: Tuesday, October 20, 2009 7:35 AM
> To: Koopmann, Jan-Peter
> Cc: sipp-users@lists.sourceforge.net
> Subject: Re: [Sipp-users] Testing Registers and Subscriptions
>
>
> On Tue, Oct 20, 2009 at 1:46 AM, Koopmann, Jan-Peter <
> jan-pe...@koopmann.eu> wrote:
> Hi,
>
> I would like to use sipp to test a few thousand registrations and
> subscriptions. I have succeeded in convincing sipp to do x concurrent
> calls of y seconds etc. However our current tests with registrations
> all seem to register the phone and after working through the XML file
> drop the connection. How do I teach sipp to get 2000 registrations
> going, keep them up, do subscriptions, answer to options (asterisk)? I
> am aware that you are not going to give me detailed step by step
> instructions but if you could put me into the right way of sipp-
> thinking it would help tremendously!
>
> As far as I know, SIPp would not be able to handle Registration,
> Subscription and OPTIONS messages in a single scenario unless you can ensure
> all of the messages involved come and go with the same Call-ID for each
> "session" doing REGISTER/SUBSCRIBE/OPTIONS.
> If I had to do it I would write 3 different scenarios and run them
> concurrently (start one instance of SIPp for each of them).
> To keep registering and subscribing, you just have to use a loop:
>
> <label id="start_of_loop" />
>
> ... REGISTER or SUBSCRiBE processing goes here ...
>
> <pause/>
>
> <nop next="start_of_loop" />
>
>
> Then, if the OPTIONS is sent by Asterisk due to Registration (I don't know,
> I never used Asterisk), you must set the IP/Port of a third instance of SIPp
> running a UAS scenario and pass this info in the header Contact of REGISTER
> (or SUBSCRIBE).
>
> And maybe ims_bench could handle this better (I don't know, I never used
> that either)
> http://sipp.sourceforge.net/ims_bench/
> as it says it improves on SIPp by adding (among other things)
> "multi-scenario support".
>
> (Obs: the REGISTER and SUBSCRIBE scenarios, could be merged in a single
> one).
>
>
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> _______________________________________________
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--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com
<?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 default 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder">
<recv request="INVITE" rrs="true" crlf="true">
</recv>
<!-- The '[last_*]' keyword automatically is replaced by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been recieved, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurences are concatenated (CRLF seperated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
[last_Record-Route:]
Contact: <sip:sameer@[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
[last_Record-Route:]
Contact: <sip:sameer@[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=root 3793000 3793000 IN IP4 172.16.17.151
s=call
c=IN IP4 172.16.17.151
t=0 0
m=audio 5064 RTP/AVP 8 0 2 3 97 110 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
]]>
</send>
<recv request="ACK" rrs="true" rtd="true">
<action>
<ereg regexp=".*"
search_in="hdr"
header="Call-ID:"
assign_to="4"/>
<ereg regexp="tag=([[:alnum:]._]*)" search_in="hdr"
header="To:" assign_to="5" />
<ereg regexp="tag=([[:alnum:]._]*)" search_in="hdr"
header="From:" assign_to="6" />
</action>
</recv>
<pause milliseconds="500"/>
<send>
<![CDATA[
INVITE sip:ripunjay@xxxxxxxxxxxxx:5064 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK2293940223
Max-Forwards: 70
Route: <sip:172.16.20.196:5067;21.ccpu.com;lr>
To: "ripunjay" <sip:ripunjay@xxxxxxxx>;[$6]
From: "sameer" <sip:sameer@xxxxxxxx>;[$5]
[last_Call-ID:]
CSeq: 2 INVITE
Contact: <sip:sameer@[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=root 3793000 3793001 IN IP4 172.16.17.151
s=-
t=0 0
c=IN IP4 0.0.0.0
m=audio 8000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100" crlf="true">
</recv>
<recv response="200" rrs="true" crlf="true" rtd="true" >
</recv>
<send>
<![CDATA[
ACK sip:ripunjay@xxxxxxxxxxxxx:5064 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK1235
[routes:]
[last_To:]
[last_From:]
[last_Call-ID:]
Cseq: 2 ACK
Contact: sip:sameer@[local_ip]:[local_port]
Max-Forwards: 70
Content-Length: [len]
]]>
</send>
<send>
<![CDATA[
INVITE sip:sumathy@xxxxxxxxxxxxx:5067 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK2293940223
Max-Forwards: 70
Route: <sip:172.16.20.196:5067;21.ccpu.com;lr>
To: "sumathy" <sip:sumathy@xxxxxxxx>
From: "sameer" <sip:sameer@xxxxxxxx>;tag=[call_number]
Call-ID: 1234567890///[call_id]
CSeq: 5 INVITE
Contact: <sip:sameer@[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 55655765 2453687637 IN IP4 127.0.0.1
s=-
t=0 0
c=IN IP4 [media_ip]
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" rrs="true" crlf="true">
</recv>
<recv response="200" rrs="true" crlf="true" rtd="true" >
</recv>
<send>
<![CDATA[
ACK sip:sumathy@xxxxxxxxxxxxx:5064 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK2293940444
[last_From:]
[routes:]
[last_To:]
[last_Call-ID:]
Cseq: 5 ACK
Contact: sip:sameer@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: [len]
]]>
</send>
<pause milliseconds="500"/>
<send>
<![CDATA[
INVITE sip:sumathy@xxxxxxxxxxxxx:5064 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK2293940555
Max-Forwards: 70
[routes:]
[last_To:]
[last_From:]
[last_Call-ID:]
CSeq: 6 INVITE
Contact: <sip:sameer@[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687638 IN IP4 127.0.0.1
s=-
t=0 0
c=IN IP4 0.0.0.0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendonly
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="200" rrs="true" crlf="true" rtd="true" >
<action>
<ereg regexp="tag=([[:alnum:]._]*)" search_in="hdr"
header="From:" assign_to="10" />
<ereg regexp="tag=([[:alnum:]._]*)" search_in="hdr"
header="To:" assign_to="11" />
<ereg regexp=".*"
search_in="hdr"
header="Call-ID:"
check_it="true"
assign_to="12"/>
</action>
</recv>
<send>
<![CDATA[
ACK sip:sumathy@xxxxxxxxxxxxx:5064 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK2293940666
[last_From:]
[routes:]
[last_To:]
[last_Call-ID:]
Cseq: 6 ACK
Contact: sip:sameer@[local_ip]:[local_port]
Max-Forwards: 70
Content-Length: [len]
]]>
</send>
<send>
<![CDATA[
REFER sip:ripunjay@xxxxxxxxxxxxx:5064 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK2293940223
Max-Forwards: 70
Route: <sip:172.16.20.196:5067;21.ccpu.com;lr>
To: "ripunjay" <sip:ripunjay@xxxxxxxx>;[$6]
From: "sameer" <sip:sameer@xxxxxxxx>;[$5]
Call-ID: [$4]
CSeq: 31 REFER
Refer-To: <sip:sumathy@xxxxxxxx?Replaces=[$12];to-tag=[$11];from-tag=[$10]>
Contact: <sip:sameer@[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<recv response="202" crlf="true" rtd="true" >
</recv>
<recv request="NOTIFY" rrs="true" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:sameer@[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<recv request="BYE" rrs="true" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:sameer@[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<recv request="NOTIFY" rrs="true" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:sameer@[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
BYE sip:ripunjay@xxxxxxxxxxxxx:5064 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
[routes:]
To: "ripunjay" <sip:ripunjay@xxxxxxxx>;[$6]
From: "sameer" <sip:sameer@xxxxxxxx>;[$5]
Call-ID: [$4]
CSeq: 53 BYE
Contact: <sip:sameer@[local_ip]:[local_port];transport=[transport]>
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true" rtd="true" >
</recv>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<pause milliseconds="4000"/>
<!-- 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><?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!--
SEQUENTIAL
Gopal;2001
Murali;1777
-->
<!--./sipp -i 192.168.0.87 -p 6060 -sf call_hold.xml -inf data_hold.txt -m 2 -l 1 -d 2000 192.168.0.54 -trace_err-->
<scenario name="Call Hold">
<send retrans="500">
<![CDATA[
INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: [field0] <sip:[field1]@[remote_ip]:[remote_port]>;tag=[call_number]
To: [field0] <sip:[field1]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[field1]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
t=0 0
c=IN IP4 [media_ip]
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200" rtd="true">
</recv>
<send>
<![CDATA[
ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[field1]@[remote_ip]:[remote_port]>;tag=[call_number]
To: [field0] <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[field1]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<pause milliseconds="5000"/>
<!-- invite on hold-->
<send retrans="500">
<![CDATA[
INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: [field0] <sip:[field1]@[remote_ip]:[remote_port]>;tag=[call_number]
To: [field0] <sip:[field1]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: sip:[field1]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=alice 2890844526 2890844526 IN IP4 [media_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=sendonly
]]>
</send>
<recv response="200" rtd="true">
</recv>
<pause milliseconds="5000"/>
<send>
<![CDATA[
ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[field1]@[remote_ip]:[remote_port]>;tag=[call_number]
To: [field0] <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:[field1]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- no rtp -->
<send retrans="500">
<![CDATA[
INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: [field0] <sip:[field1]@[remote_ip]:[remote_port]>;tag=[call_number]
To: [field0] <sip:[field1]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 3 INVITE
Contact: sip:[field1]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
t=0 0
c=IN IP4 [media_ip]
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="200" rtd="true">
</recv>
<send>
<![CDATA[
ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[field1]@[remote_ip]:[remote_port]>;tag=[call_number]
To: [field0] <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 ACK
Contact: sip:[field1]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<pause milliseconds="5000"/> <!-- media established-->
<send retrans="500">
<![CDATA[
BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[field1]@[remote_ip]:[remote_port]>;tag=[call_number]
To: [field0] <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 4 BYE
Contact: sip:[field1]@[local_ip]:[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="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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