sipp-users — sipp users

[Sipp-users] SIPp 3.7.3 - TLS key logging

[Olivier T. <oli...@gm...>]

Hello,

I'm trying to use the TLS key logging with version 3.7.3, but I can't get
it to work.
Is it supposed to be supported in this release?

I built SIPp with "OpenSSL 1.1.1d".

Before running SIPp, I set the SSLKEYLOGFILE environment variable as
follows:
$ export SSLKEYLOGFILE=/home/<myAccount>/ssl_keys.log

However, the ssl_keys.log is never created...

Any idea?

---

*To enable TLS key logging pass -DUSE_SSL=KL to cmake.*

*TLS key logging records the TLS Session Keys to a key log file when
the SSLKEYLOGFILE environment variable is set. It allows to decrypt SIPS
traffic generated or received by SIPp using Wireshark. For more details
see: https://wiki.wireshark.org/TLS <https://wiki.wireshark.org/TLS>*

*You need to compile with OpenSSL>=1.1.1 in order to use TLS key logging.*
Olivier T.

[Sipp-users] RTP traffic Problem

[Imen E. <ime...@gm...>]

Hello,
    I'm trying to generate 2500 concurrent calls. the caller (UAC) will
send 5 min RTP traffic using the G711 codec. The called (UAS) will respond
with 4.5 min RTP traffic using the G729 codec. The caller and the called
are runing on the same VM. I have another VM who will do the transcoding.
SIPP indicates that all calls were finished successfully.
    The problem is in the caller screen, the audio RTP out reaches only
about 4000KB/s , which is so low for 2500 concurrent calls with G711.
    Also in the called screen, the RTP out reaches only about 800 KB/s
which is again lower than the expected for 2500 concurrent calls with G729.
    The VM CPU is normal approximately 25%, and the transmitted traffic is
approximately 6 MebiBytes which is lower than the expected again.
    Are there some settings in SIPP that might be limiting throughput? Is
it because of my VM capacity and configuration?
    I’d appreciate any insights or suggestions.
 Setup:
* SIPP Version : SIPp v3.7.3-TLS-SCTP-PCAP-SHA256.
* Caller Command: sipp -sf
/sipp/uac/lte-sp-c01-uac-173-5MinRTP-ConcurrentCall.xml -inf
/sipp/uac/a-numbers-5minRTP-concurrent-calls-from-header.csv -inf
/sipp/uac/b-numbers-1.csv -inf
/sipp/uac/a-numbers-5minRTP-concurrent-calls-pai-header.csv -cid_str
%u-lte-sp-c01-uac-65-%p@%s -deadcall_wait 10000 -watchdog_reset 10000 -r
1000 -rp 1000 -l 2500 -m 2500 -i <local-ip> -p <local-port> -mi <local-ip>
-min_rtp_port 10000 -max_rtp_port 20000  -trace_err <remoteIP:RemotePort>
*  Called Command : sudo /opt/linxa/bin/sipp -sf
/opt/linxa/lib/sipp/uas/lte-sp-c01-uas-174-5minRTP.xml -deadcall_wait 10000
-watchdog_reset 10000 -i <local-ip> -p <local-port> -mi <local-ip>
 -min_rtp_port 20000 -max_rtp_port 30000
* Caller Scenario: Custom XML with <exec
rtp_stream="sipp/uac/rtp_G711_A.wav,-1,8,PCMA/8000"/>  for infinite
playback, followed by a 5-minute timer (<pause milliseconds="300000"/>) and
then RTP pause.
* Called Scenario:  Custom XML with <exec
rtp_stream="sipp/uac/rtp_G729.G729,-1,18,G729/8000"/>  for infinite
playback, followed by a 4.5-minute timer (<pause milliseconds="270000"/>)
and then RTP pause.
* Environment: VM with 28 vCPUs, 25% CPU usage during test, monitored via
bmon.
* RTP Settings: Using defaults SIPP RTP settings.

[Sipp-users] Creating a dialup Audio/Video loopback: Registering SIPp with Webex Control Hub, UAS and -rtp_echo

[Economides, T. <tec...@ve...>]

My goal here is to use SIPp in server mode, have it register to Webex Control Hub and wait for calls to be routed to it (via an internal extension). When calls come to SIPp it should echo back all RTP streams that it receives.  We use such a loopback scenario (using hardware videoconf systems, at the moment, to receive and loop back the calls) to test conference room videoconf systems, by letting them dial up the loopback and check that audio and video from the room is looped back.

Anyhow, I'm looking for anyone who's successfully gotten SIPp to talk to Webex Control Hub, using TLS encryption, that can give me some guidance.

Right now I am puzzled at how to set up both Control Hub and SIPp so that they actually talk to one another.  Here's what I've tried:


  *   In Control Hub (CH), I created a 'Generic IPPhone Customer Managed" device, and that device is contained within a Workspace with Webex Calling enabled. CH also allows for a 'Generic IPgateway Customer Managed" but I haven't tried using that configuration yet.)
  *   I've assigned an extension to the Workspace (x1277)
  *   CH provided a username and password, along with an outbound proxy server name and a 'LineID':
     *   Outbound Proxy is: jfk06.hosted-us.bcld.webex.com. CH docs state that this is DNS resolved with SRV records, and SIPp doesn't support SRV lookup from what I can tell. So I manually resolved this address to 23.89.40.55 and port 8934
     *   Line ID is of the form: <local-name-of-random-letters-and-numbers>@83549438.cisco-bcld.com
  *   UAS: Because I want SIPp to listen for calls and respond to them, it seems that SIPp should be in UAS mode. Right?
  *   My SIPp command line looks like: sudo ./sipp 23.89.40.55:8934 -i <local, NATted IP addy> -p 5060 -sn UAS -t 1n -max_socket 1023 -au <SIP username> -ap <SIP passwd> -s <LineID user part, left of @> -rtp_echo -trace_err -trace_msg -trace_error_codes -trace_calldebug -trace_logs -trace_screen
  *   When I run the command, I can see the Scenario Screen, and it shows that the REGISTER -----> messages quantity is 30 (I'll post a screen shot but I'm guessing it won't come through the listserve)
[cid:image001.png@01DB7EC7.4DE05220]

When I do this, there's no indication that SIPp is talking to CH.  I use netstat to watch for traffic on local port 5060 when I run SIPp, but no activity is detected.  Further, when I try dialing the 1277 extension from another device registered to CH, I get an error recording saying "we're sorry, your call cannot be completed."

There is a firewall between the local VM where I'm running SIPp and the Internet, but there's no blocking of packets showing up there at all.

I have a feeling that I'm trying to do something that SIPp is capable of (registering to another SIP server and waiting for calls to be forwarded to it) but I need some help.

Cheers!

Theo Economides, AV / Multimedia Supervisor

VedderPrice
T +1 312 609 4597
222 North LaSalle Street, Chicago, Illinois 60601
web<http://www.vedderprice.com/> | email<mailto:tec...@ve...> | offices<http://www.vedderprice.com/locations/>

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d250106us

Re: [Sipp-users] REGISTER and Receive an incoming call on SIPp over TLS

[Jyotirmaya M. <jyo...@gm...>]

Hello Pavel,

I have separated my scripts into two parts: one for REGISTER and another
for Incoming INVITE. In the REGISTER script, I have hardcoded the Contact
header with a specific port, and I have started the UAS (User Agent Server)
script on the same address and port specified in the Contact header.

Also I have added pause milliseconds="120000" to make the REGISTER session
alive.

However, the issue persists regardless of whether I hardcode the Contact
header or not. When I initiate a call, the INVITE is still sent to the
REGISTER SIPp instance instead of being directed to the INVITE SIPp
instance.


#########
#########
/usr/local/src/sipp-3.5.3/sipp -sf reg_with_contact.xml -inf test_itly.csv
-i 88.12.45.295 -p 5061 78.12.45.256:5061 -t ln -tls_key $PWD/myx_key.pem
-tls_cert myx_cert.pem -m 1
#########
#########

 cat reg_with_separate_contact.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="UAC with media">

  <send retrans="500">
    <![CDATA[

      REGISTER sip:[field1];transport=tls SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[field0]@[field1]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: sip:user_yyyKuu@88.12.45.295:5069
      Max-Forwards: 5
      Expires: 3600

Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
      k:timer,path,replaces
      User-Agent: SIPp
      Content-Length: 0

    ]]>
  </send>

  <recv response="401" auth="true">
  </recv>

  <send retrans="500">
    <![CDATA[

      REGISTER sip:[field1];transport=tls SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[field0]@[field1]>
      Call-ID: [call_id]
      CSeq: 2 REGISTER
      Contact: sip:user_KvwJyyyKuu@88.12.45.295:5069
      [field2]
      Max-Forwards: 5
      Expires: 3600

Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
      k:timer,path,replaces
      User-Agent: SIPp
      Content-Length: 0

    ]]>
  </send>

 <!--recv response="100" auth="true">
        </recv-->

<recv response="100" optional="true">
 </recv>
  <recv response="200"></recv>
  <pause milliseconds="120000">

  <send retrans="500">
    <![CDATA[

      REGISTER sip:[field1];transport=tls SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[field0]@[field1]>;tag=[call_number]
      To: <sip:[field0]@[field1]>
      Call-ID: [call_id]
      CSeq: 3 REGISTER
      Contact: sip:user_KvwJyyyKuu@88.12.45.295:5069
      [field2]
      Max-Forwards: 5
      Expires: 3600

Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
      k:timer,path,replaces
      User-Agent: SIPp
      Content-Length: 0

    ]]>
  </send>

 <!--recv response="100" auth="true">
              </recv-->

<recv response="100" optional="true">
 </recv>
  <recv response="200"></recv>

<pause milliseconds="120000"/>

  <!-- 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>


#######
#######
Incoming Call Script:
/usr/local/src/sipp-3.5.3/sipp -sf 200OK_Media-caller-bye.xml -i
88.12.45.295 -p 5069 -t ln -tls_key $PWD/myx_key.pem -tls_cert myx_cert.pem
#######
#######

 cat 200OK_Media-caller-bye.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 default 'uas' scenario.                       -->
<!--                                                                    -->

<scenario name="Basic UAS responder">
  <!-- 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 request="INVITE" crlf="true">
      <action>
     <!--ereg regexp="\<(.*)\>" search_in="hdr" header="Contact:"
check_it="true" assign_to="new" />
               <ereg regexp=".*" search_in="hdr" header="From:"
assign_to="remote_from"/>
     <ereg regexp=".*" search_in="hdr" header="To:"
assign_to="remote_to"/-->
     <ereg regexp=".*" search_in="hdr" header="Record-Route:" occurence="1"
assign_to="2"/>
     <ereg regexp=".*" search_in="hdr" header="Record-Route:" occurence="2"
assign_to="3"/>
     </action>


  </recv>

  <!-- The '[last_*]' keyword is replaced automatically 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 received, 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/2.0 100 Trying
      Record-Route: [$2]
      Record-Route: [$3]
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0
    ]]>
  </send-->


  <send>
    <![CDATA[

      SIP/2.0 180 Ringing
      Record-Route: [$2]
      Record-Route: [$3]
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
  </send>

  <send retrans="500">
    <![CDATA[

      SIP/2.0 200 OK
      Record-Route: [$2]
      Record-Route: [$3]
      [last_via]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s= Kumartest
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0 101 13
      o=FreeSWITCH 1728427243 1728427244 IN IP4 30.30.30.30
      a=rtpmap:0 PCMU/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-15
      a=rtpmap:13 CN/8000
      a=ptime:20
    ]]>
  </send>

  <recv request="ACK"
        rtd="true"
        crlf="true">
  </recv>

  <!-- Play an RTP Stream                                     -->
  <nop>
    <action>
      <exec play_pcap_audio="playspeech.pcap"/>
    </action>
  </nop>

  <!--pause milliseconds="120000"/-->


  <recv request="BYE">
  </recv>

  <send>
    <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
  </send>

  <!-- 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>

Best Regards,
KJM

On Tue, Dec 10, 2024 at 12:12 PM Jyotirmaya Mohanty <jyo...@gm...>
wrote:

> Hello Pavel,
>
> Could you please review my script an suggest the required changes.
>
>  modified_reg_with_call.xml
>> <?xml version="1.0" encoding="ISO-8859-1" ?>
>> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>> <scenario name="REGISTRATION with INCOMING CALL">
>>   <send retrans="500">
>>     <![CDATA[
>>       REGISTER sip:[field1];transport=tls SIP/2.0
>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>       From: <sip:[field0]@[field1]>;tag=[call_number]
>>       To: <sip:[field0]@[field1]:[remote_port]>
>>       Call-ID: [call_id]
>>       CSeq: 1 REGISTER
>>       Contact: sip:[field0]@[local_ip]:[local_port]
>>       Max-Forwards: 5
>>       Expires: 3600
>>       Allow:
>> INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
>>       k: timer,path,replaces
>>       User-Agent: SIPp
>>       Content-Length: 0
>>     ]]>
>>   </send>
>>
>>   <recv response="401" auth="true"></recv>
>>
>>   <send retrans="500">
>>     <![CDATA[
>>       REGISTER sip:[field1];transport=tls SIP/2.0
>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>       From: <sip:[field0]@[field1]>;tag=[call_number]
>>       To: <sip:[field0]@[field1]>
>>       Call-ID: [call_id]
>>       CSeq: 2 REGISTER
>>       Contact: sip:[field0]@[local_ip]:[local_port]
>>       [field2]
>>       Max-Forwards: 5
>>       Expires: 3600
>>       Allow:
>> INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
>>       k: timer,path,replaces
>>       User-Agent: SIPp
>>       Content-Length: 0
>>     ]]>
>>   </send>
>>
>>   <recv response="100" optional="true"></recv>
>>   <recv response="200"></recv>
>>
>>   <recv request="INVITE" next="handle_INVITE"/>
>>   <label id="handle_INVITE"/>
>>
>>   <send>
>>     <![CDATA[
>>       SIP/2.0 100 Trying
>>       Record-Route: [$2]
>>       Record-Route: [$3]
>>       Via: [$4]
>>       Via: [$5]
>>       [last_From:]
>>       [last_To:]
>>       [last_Call-ID:]
>>       [last_CSeq:]
>>       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>>       Content-Length: 0
>>     ]]>
>>   </send>
>>
>>   <send>
>>     <![CDATA[
>>       SIP/2.0 180 Ringing
>>       Record-Route: [$2]
>>       Record-Route: [$3]
>>       Via: [$4]
>>       Via: [$5]
>>       [last_From:]
>>       [last_To:];tag=[call_number]
>>       [last_Call-ID:]
>>       [last_CSeq:]
>>       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>>       Content-Length: 0
>>     ]]>
>>   </send>
>>
>>   <send retrans="500">
>>     <![CDATA[
>>       SIP/2.0 200 OK
>>       Record-Route: [$2]
>>       Record-Route: [$3]
>>       Via: [$4]
>>       Via: [$5]
>>       [last_From:]
>>       [last_To:];tag=[call_number]
>>       [last_Call-ID:]
>>       [last_CSeq:]
>>       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>>       Content-Type: application/sdp
>>       Content-Length: [len]
>>
>>       v=0
>>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>>       s= Kumartest
>>       c=IN IP[media_ip_type] [media_ip]
>>       t=0 0
>>       m=audio [media_port] RTP/AVP 0 101 13
>>       a=rtpmap:0 PCMU/8000
>>       a=rtpmap:101 telephone-event/8000
>>       a=fmtp:101 0-15
>>       a=rtpmap:13 CN/8000
>>     ]]>
>>   </send>
>>
>>   <recv request="ACK" rtd="true" crlf="true"></recv>
>>
>>   <nop>
>>     <action>
>>       <exec play_pcap_audio="playspeech.pcap"/>
>>     </action>
>>   </nop>
>>
>>   <recv request="BYE"></recv>
>>
>>   <send>
>>     <![CDATA[
>>       SIP/2.0 200 OK
>>       [last_Via:]
>>       [last_From:]
>>       [last_To:]
>>       [last_Call-ID:]
>>       [last_CSeq:]
>>       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>>       Content-Length: 0
>>     ]]>
>>   </send>
>>
>>   <!-- 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>
>>
>>
> On Mon, Dec 9, 2024 at 8:18 PM Šindelka Pavel <sin...@tt...> wrote:
>
>> Hello KJM,
>>
>> I'd suggest you to read
>> https://sourceforge.net/p/sipp/mailman/message/34707334/ as it covers
>> most of your issue.
>>
>> The use of TLS may complicate things further, though - I hazily remember
>> that I had to patch something in the source code so that sipp could
>> initiate outgoing TCP sessions and also expect incoming ones. But on the
>> other hand, since you start by sending a REGISTER, the typical use case
>> would be that the remote party would not create a new TCP session to
>> deliver the INVITE to your scenario but rather reuse the existing one
>> created by the REGISTER, so that patch may not be necessary.
>>
>> Pavel
>>
>>
>> Dne 09.12.2024 v 14:48 Jyotirmaya Mohanty napsal(a):
>>
>> Hello Team,
>>
>> I have a query regarding SIPp script. want to REGISTER and Receive an
>> incoming call on SIPp over TLS. Is it possible?
>>
>>                                  Messages  Retrans   Timeout
>> Unexpected-Msg
>>     REGISTER ---------->         1         0         0
>>                 401 <----------         1         0         0         0
>>     REGISTER ---------->         1         0         0
>>                  100 <----------        1         0         0         0
>>                 200 <----------         1         0         0         0
>>            INVITE <----------         0         0         0         0
>>
>>                 100 ---------->         0         0
>>                 180 ---------->         0         0
>>                 200 ---------->         0         0         0
>>              ACK <----------  E-RTD1 0         0         0         0
>>
>>               [ NOP ]
>>          BYE <----------         0         0         0         0
>>          200 ---------->         0         0
>> ------- Waiting for active calls to end. Press [q] again to force exit.
>> -------
>>
>>
>> My Registration is successful, but while receiving INVITE I am getting
>> 2024-12-09      13:18:31.970086 1733750311.970086: Discarding message
>> which can't be mapped to a known SIPp call:
>> INVITE sip:user_KvwJyyyKuu@88.112.45.95:5069
>> <http://sip:user_KvwJyyyKuu@38.102.145.195:5069/> SIP/2.0
>>
>> For UDP transport and if I ran separate script for REGISTER and INVITE
>> UAS. It is working fine.
>>
>> Please let me know and Thanks in advance, how to run it for TLS
>>
>> Best Regards
>> KJM
>>
>>
>> _______________________________________________
>> Sipp-users mailing lis...@li...://lists.sourceforge.net/lists/listinfo/sipp-users
>>
>> --
>>
>>
>> *Ing. Pavel Šindelka * senior specialista
>>
>>
>>
>>
>>
>> TTC MARCONI s. r. o.
>> Třebohostická 987/5, 100 00  Praha 10
>> +420 234 051 712, +420 602 355 199
>> sin...@tt..., www.ttc-marconi.com
>>
>> _______________________________________________
>> Sipp-users mailing list
>> Sip...@li...
>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>
>

Re: [Sipp-users] REGISTER and Receive an incoming call on SIPp over TLS

[Jyotirmaya M. <jyo...@gm...>]

Hello Pavel,

Could you please review my script an suggest the required changes.

 modified_reg_with_call.xml
> <?xml version="1.0" encoding="ISO-8859-1" ?>
> <!DOCTYPE scenario SYSTEM "sipp.dtd">
> <scenario name="REGISTRATION with INCOMING CALL">
>   <send retrans="500">
>     <![CDATA[
>       REGISTER sip:[field1];transport=tls SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>       From: <sip:[field0]@[field1]>;tag=[call_number]
>       To: <sip:[field0]@[field1]:[remote_port]>
>       Call-ID: [call_id]
>       CSeq: 1 REGISTER
>       Contact: sip:[field0]@[local_ip]:[local_port]
>       Max-Forwards: 5
>       Expires: 3600
>       Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
>       k: timer,path,replaces
>       User-Agent: SIPp
>       Content-Length: 0
>     ]]>
>   </send>
>
>   <recv response="401" auth="true"></recv>
>
>   <send retrans="500">
>     <![CDATA[
>       REGISTER sip:[field1];transport=tls SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>       From: <sip:[field0]@[field1]>;tag=[call_number]
>       To: <sip:[field0]@[field1]>
>       Call-ID: [call_id]
>       CSeq: 2 REGISTER
>       Contact: sip:[field0]@[local_ip]:[local_port]
>       [field2]
>       Max-Forwards: 5
>       Expires: 3600
>       Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
>       k: timer,path,replaces
>       User-Agent: SIPp
>       Content-Length: 0
>     ]]>
>   </send>
>
>   <recv response="100" optional="true"></recv>
>   <recv response="200"></recv>
>
>   <recv request="INVITE" next="handle_INVITE"/>
>   <label id="handle_INVITE"/>
>
>   <send>
>     <![CDATA[
>       SIP/2.0 100 Trying
>       Record-Route: [$2]
>       Record-Route: [$3]
>       Via: [$4]
>       Via: [$5]
>       [last_From:]
>       [last_To:]
>       [last_Call-ID:]
>       [last_CSeq:]
>       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>       Content-Length: 0
>     ]]>
>   </send>
>
>   <send>
>     <![CDATA[
>       SIP/2.0 180 Ringing
>       Record-Route: [$2]
>       Record-Route: [$3]
>       Via: [$4]
>       Via: [$5]
>       [last_From:]
>       [last_To:];tag=[call_number]
>       [last_Call-ID:]
>       [last_CSeq:]
>       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>       Content-Length: 0
>     ]]>
>   </send>
>
>   <send retrans="500">
>     <![CDATA[
>       SIP/2.0 200 OK
>       Record-Route: [$2]
>       Record-Route: [$3]
>       Via: [$4]
>       Via: [$5]
>       [last_From:]
>       [last_To:];tag=[call_number]
>       [last_Call-ID:]
>       [last_CSeq:]
>       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>       Content-Type: application/sdp
>       Content-Length: [len]
>
>       v=0
>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>       s= Kumartest
>       c=IN IP[media_ip_type] [media_ip]
>       t=0 0
>       m=audio [media_port] RTP/AVP 0 101 13
>       a=rtpmap:0 PCMU/8000
>       a=rtpmap:101 telephone-event/8000
>       a=fmtp:101 0-15
>       a=rtpmap:13 CN/8000
>     ]]>
>   </send>
>
>   <recv request="ACK" rtd="true" crlf="true"></recv>
>
>   <nop>
>     <action>
>       <exec play_pcap_audio="playspeech.pcap"/>
>     </action>
>   </nop>
>
>   <recv request="BYE"></recv>
>
>   <send>
>     <![CDATA[
>       SIP/2.0 200 OK
>       [last_Via:]
>       [last_From:]
>       [last_To:]
>       [last_Call-ID:]
>       [last_CSeq:]
>       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>       Content-Length: 0
>     ]]>
>   </send>
>
>   <!-- 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>
>
>
On Mon, Dec 9, 2024 at 8:18 PM Šindelka Pavel <sin...@tt...> wrote:

> Hello KJM,
>
> I'd suggest you to read
> https://sourceforge.net/p/sipp/mailman/message/34707334/ as it covers
> most of your issue.
>
> The use of TLS may complicate things further, though - I hazily remember
> that I had to patch something in the source code so that sipp could
> initiate outgoing TCP sessions and also expect incoming ones. But on the
> other hand, since you start by sending a REGISTER, the typical use case
> would be that the remote party would not create a new TCP session to
> deliver the INVITE to your scenario but rather reuse the existing one
> created by the REGISTER, so that patch may not be necessary.
>
> Pavel
>
>
> Dne 09.12.2024 v 14:48 Jyotirmaya Mohanty napsal(a):
>
> Hello Team,
>
> I have a query regarding SIPp script. want to REGISTER and Receive an
> incoming call on SIPp over TLS. Is it possible?
>
>                                  Messages  Retrans   Timeout
> Unexpected-Msg
>     REGISTER ---------->         1         0         0
>                 401 <----------         1         0         0         0
>     REGISTER ---------->         1         0         0
>                  100 <----------        1         0         0         0
>                 200 <----------         1         0         0         0
>            INVITE <----------         0         0         0         0
>
>                 100 ---------->         0         0
>                 180 ---------->         0         0
>                 200 ---------->         0         0         0
>              ACK <----------  E-RTD1 0         0         0         0
>
>               [ NOP ]
>          BYE <----------         0         0         0         0
>          200 ---------->         0         0
> ------- Waiting for active calls to end. Press [q] again to force exit.
> -------
>
>
> My Registration is successful, but while receiving INVITE I am getting
> 2024-12-09      13:18:31.970086 1733750311.970086: Discarding message
> which can't be mapped to a known SIPp call:
> INVITE sip:user_KvwJyyyKuu@88.112.45.95:5069
> <http://sip:user_KvwJyyyKuu@38.102.145.195:5069/> SIP/2.0
>
> For UDP transport and if I ran separate script for REGISTER and INVITE
> UAS. It is working fine.
>
> Please let me know and Thanks in advance, how to run it for TLS
>
> Best Regards
> KJM
>
>
> _______________________________________________
> Sipp-users mailing lis...@li...://lists.sourceforge.net/lists/listinfo/sipp-users
>
> --
>
>
> *Ing. Pavel Šindelka * senior specialista
>
>
>
>
>
> TTC MARCONI s. r. o.
> Třebohostická 987/5, 100 00  Praha 10
> +420 234 051 712, +420 602 355 199
> sin...@tt..., www.ttc-marconi.com
>
> _______________________________________________
> Sipp-users mailing list
> Sip...@li...
> https://lists.sourceforge.net/lists/listinfo/sipp-users
>

Re: [Sipp-users] REGISTER and Receive an incoming call on SIPp over TLS

[Šindelka P. <sin...@tt...>]

Hello KJM,

I'd suggest you to read 
https://sourceforge.net/p/sipp/mailman/message/34707334/ as it covers 
most of your issue.

The use of TLS may complicate things further, though - I hazily remember 
that I had to patch something in the source code so that sipp could 
initiate outgoing TCP sessions and also expect incoming ones. But on the 
other hand, since you start by sending a REGISTER, the typical use case 
would be that the remote party would not create a new TCP session to 
deliver the INVITE to your scenario but rather reuse the existing one 
created by the REGISTER, so that patch may not be necessary.

Pavel


Dne 09.12.2024 v 14:48 Jyotirmaya Mohanty napsal(a):
> Hello Team,
>
> I have a query regarding SIPp script. want to REGISTER and Receive an 
> incoming call on SIPp over TLS. Is it possible?
>
>  Messages  Retrans   Timeout   Unexpected-Msg
>     REGISTER ---------->         1         0         0
>                 401 <----------         1         0   0         0
>     REGISTER ---------->         1         0         0
>                  100 <----------        1         0   0         0
>                 200 <----------         1         0   0         0
>            INVITE <----------         0         0 0         0
>
>                 100 ---------->         0         0
>                 180 ---------->         0         0
>                 200 ---------->         0         0   0
>              ACK <----------  E-RTD1 0         0 0         0
>
>               [ NOP ]
>          BYE <----------         0         0         0       0
>          200 ---------->         0         0
> ------- Waiting for active calls to end. Press [q] again to force 
> exit. -------
>
>
> My Registration is successful, but while receiving INVITE I am getting
> 2024-12-09      13:18:31.970086 1733750311.970086: Discarding message 
> which can't be mapped to a known SIPp call:
> INVITE sip:user_KvwJyyyKuu@88.112.45.95:5069 
> <http://sip:user_KvwJyyyKuu@38.102.145.195:5069/> SIP/2.0
>
> For UDP transport and if I ran separate script for REGISTER and INVITE 
> UAS. It is working fine.
>
> Please let me know and Thanks in advance, how to run it for TLS
>
> Best Regards
> KJM
>
>
> _______________________________________________
> Sipp-users mailing list
> Sip...@li...
> https://lists.sourceforge.net/lists/listinfo/sipp-users
-- 

*Ing. Pavel Šindelka
* senior specialista

TTC MARCONI s. r. o.
Třebohostická 987/5, 100 00  Praha 10
+420 234 051 712, +420 602 355 199
sin...@tt..., www.ttc-marconi.com

[Sipp-users] REGISTER and Receive an incoming call on SIPp over TLS

[Jyotirmaya M. <jyo...@gm...>]

Hello Team,

I have a query regarding SIPp script. want to REGISTER and Receive an
incoming call on SIPp over TLS. Is it possible?

                                 Messages  Retrans   Timeout
Unexpected-Msg
    REGISTER ---------->         1         0         0
                401 <----------         1         0         0         0
    REGISTER ---------->         1         0         0
                 100 <----------        1         0         0         0
                200 <----------         1         0         0         0
           INVITE <----------         0         0         0         0

                100 ---------->         0         0
                180 ---------->         0         0
                200 ---------->         0         0         0
             ACK <----------  E-RTD1 0         0         0         0

              [ NOP ]
         BYE <----------         0         0         0         0
         200 ---------->         0         0
------- Waiting for active calls to end. Press [q] again to force exit.
-------


My Registration is successful, but while receiving INVITE I am getting
2024-12-09      13:18:31.970086 1733750311.970086: Discarding message which
can't be mapped to a known SIPp call:
INVITE sip:user_KvwJyyyKuu@88.112.45.95:5069
<http://sip:user_KvwJyyyKuu@38.102.145.195:5069/> SIP/2.0

For UDP transport and if I ran separate script for REGISTER and INVITE UAS.
It is working fine.

Please let me know and Thanks in advance, how to run it for TLS

Best Regards
KJM

[Sipp-users] (no subject)

[Jyotirmaya M. <jyo...@gm...>]

Hello Team,

I have a query regarding SIPp script. Can want to REGISTER and Receive an
incoming call on SIPp over TLS. Is it possible?

[image: image.png]

Currently I am getting
2024-12-09      13:18:31.970086 1733750311.970086: Discarding message which
can't be mapped to a known SIPp call:
INVITE sip:user_KvwJyyyKuu@88.112.45.95:5069
<http://sip:user_KvwJyyyKuu@38.102.145.195:5069/> SIP/2.0


Please let me know and Thanks in advance.

Best Regards
KJM

[Sipp-users] IMS Bench SIPp source

[Danish B. <dan...@ng...>]

Hi,

I can't seem to be able to download IMS Bench SIPp source as suggested in
the documentation (
https://sipp.sourceforge.net/ims_bench/reference.html#Using+IMS+Bench+SIPp)
using the command:

svn co http://sipp.svn.sourceforge.net/svnroot/sipp/sipp/branches/ims_bench
 ims_bench
svn: E170013: Unable to connect to a repository at URL '
https://sourceforge.net/projects/sipp/'
svn: E175003: The server at 'https://sourceforge.net/projects/sipp/' does
not support the HTTP/DAV protocol

Could someone kindly share if the URL has changed?

Best Regards,

Danish

[Sipp-users] Cyber-suite AES-256 algorithm - SIPP support

[Chandrasekaran A. <amb...@al...>]

Hello all,
I need one information. Currently we are using AES-128 algorithm for SRTP in SIPP SIP TLS . But we want to test it with Cyber-suite AES-256 algorithm. SIPP is supported for AES-256 ? 

Thanks,





-----Original Message-----
From: Jaime J. Maiz <ma...@ho...> 
Sent: 25 October 2024 23:09
To: sip...@li...
Subject: EXT: [Sipp-users] Mixed Mode


** External email - Please consider with caution **


Hello,
I was looking for a feature that was discussed a while ago called MIXED mode. Where sips can act as both the client and server to closely match a real device behavior.

I just pulled the latest version of sipp and noticed this feature is not in the current code base. Was there any particular reason this feature is missing from the codebase?

Thanks,
Jaime

_______________________________________________
Sipp-users mailing list
Sip...@li...
https://lists.sourceforge.net/lists/listinfo/sipp-users

[Sipp-users] Mixed Mode

[Jaime J. M. <ma...@ho...>]

Hello, 
I was looking for a feature that was discussed a while ago called MIXED mode. Where sips can act as both the client and server to closely match a real device behavior.

I just pulled the latest version of sipp and noticed this feature is not in the current code base. Was there any particular reason this feature is missing from the codebase?

Thanks,
Jaime

Re: [Sipp-users] Assistance Required for Compiling SIPp with TLS Support

[Ihor O. <igo...@gm...>]

Hello,

I have SIPp 3.7.2 with TLS support (Dockerfile) as a part of my project

https://github.com/igorolhovskiy/volts/blob/main/build/Dockerfile.sipp

How it's called can be found in a following python code parts:

https://github.com/igorolhovskiy/volts/blob/main/build/src/sipp/sipp.py#L41
https://github.com/igorolhovskiy/volts/blob/main/build/src/sipp/sipp.py#L62

Hope, this can give a hint.

Le 26/08/2024 à 12:10, Jafar Sarif a écrit :
>
> Dear SIPp Community,
>
> I hope this message finds you well.
>
> I am currently facing an issue while running a SIPp XML scenario over 
> TLS. Upon execution, I receive the following error message:
> TLS_init_context: SSL_CTX_use_certificate_file failed: 
> error:02001002:system library:fopen:No such file or directory.
>
> It seems that SIPp is unable to locate the required certificate file 
> during the TLS initialization process. I have ensured that OpenSSL is 
> properly installed on the system and that all necessary header files 
> are present. However, despite these checks, the issue persists.
>
> Could you kindly assist me in identifying the cause of this error and 
> provide guidance on resolving the issue? Specifically, I would 
> appreciate your advice on how to correctly configure SIPp to load the 
> required certificate file for TLS communication.
>
>
> On Sat, Aug 24, 2024, 3:42 AM Jafar Sarif <sap...@gm...> wrote:
>
>     I will try installing the new version.
>     Thanks for your help.
>
>
>     On Sat, Aug 24, 2024, 3:40 AM Jafar Sarif <sap...@gm...>
>     wrote:
>
>         Hi,
>         I am using SIPp 3.3V which uses makefile for its build system.
>
>
>         On Sat, Aug 24, 2024, 3:25 AM mayamatakeshi
>         <may...@gm...> wrote:
>
>
>
>             On Sat, Aug 24, 2024 at 6:44 AM Jafar Sarif
>             <sap...@gm...> wrote:
>
>                 I am writing to seek assistance with compiling SIPp to
>                 enable TLS support. I have successfully installed SIPp
>                 but encountered difficulties when attempting to
>                 compile it with TLS support.
>
>                 *Issue* *Details:*
>
>                 Current Installation: I have SIPp version 3.3
>                 installed, but it was compiled without TLS support.
>
>                 OpenSSL Installation: I have confirmed that OpenSSL is
>                 installed and working correctly on my system. The
>                 OpenSSL libraries and headers are available, and I
>                 have verified their installation with the following
>                 commands:
>
>                 1. *openssl* *version* shows:
>                 OpenSSL 1.1.1k FIPS 25 Mar 2021
>
>                 2. The headers are present in /*usr/include/openss*
>
>                 3. The libraries are located in /*usr/lib*, including
>                 *libssl* and *libcrypto*
>
>
>                 *Compilation Attempt:*
>                  I set the environment variables for OpenSSL:
>
>                 *export CFLAGS="-I/usr/include/openssl"*
>                 *
>                 *
>                 *export LDFLAGS="-L/usr/lib -lssl -lcrypto"*
>
>                 After setting these variables, I ran *make* *clean*
>                 followed by *make* in the SIPp directory, but the
>                 issue persists.
>
>                 *Error Message:* When I attempt to use TLS with SIPp,
>                 it gives the message: “To use a TLS transport you must
>                 compile SIPp with OpenSSL.”
>
>                 Request:
>
>                 I would appreciate any guidance or steps to ensure
>                 that SIPp is compiled correctly with TLS support. If
>                 there are specific steps or configurations needed to
>                 resolve this issue, please let me know.
>
>                 Thank you for your assistance.
>
>
>             Hi,
>             I think you skipped the instruction from
>             https://github.com/SIPp/sipp?tab=readme-ov-file#building
>             There is no need to export such variables.
>             SIPp is built using cmake so you should do something like:
>
>             cmake . -DUSE_SSL=1
>             make
>             make install
>
>
>
>
>
>
> _______________________________________________
> Sipp-users mailing list
> Sip...@li...
> https://lists.sourceforge.net/lists/listinfo/sipp-users

Re: [Sipp-users] Assistance Required for Compiling SIPp with TLS Support

[Jafar S. <sap...@gm...>]

Dear SIPp Community,

I hope this message finds you well.

I am currently facing an issue while running a SIPp XML scenario over TLS.
Upon execution, I receive the following error message:
TLS_init_context: SSL_CTX_use_certificate_file failed:
error:02001002:system library:fopen:No such file or directory.

It seems that SIPp is unable to locate the required certificate file during
the TLS initialization process. I have ensured that OpenSSL is properly
installed on the system and that all necessary header files are present.
However, despite these checks, the issue persists.

Could you kindly assist me in identifying the cause of this error and
provide guidance on resolving the issue? Specifically, I would appreciate
your advice on how to correctly configure SIPp to load the required
certificate file for TLS communication.

On Sat, Aug 24, 2024, 3:42 AM Jafar Sarif <sap...@gm...> wrote:

> I will try installing the new version.
> Thanks for your help.
>
> On Sat, Aug 24, 2024, 3:40 AM Jafar Sarif <sap...@gm...> wrote:
>
>> Hi,
>> I am using SIPp 3.3V which uses makefile for its build system.
>>
>> On Sat, Aug 24, 2024, 3:25 AM mayamatakeshi <may...@gm...>
>> wrote:
>>
>>>
>>>
>>> On Sat, Aug 24, 2024 at 6:44 AM Jafar Sarif <sap...@gm...>
>>> wrote:
>>>
>>>> I am writing to seek assistance with compiling SIPp to enable TLS
>>>> support. I have successfully installed SIPp but encountered difficulties
>>>> when attempting to compile it with TLS support.
>>>>
>>>> *Issue* *Details:*
>>>>
>>>> Current Installation: I have SIPp version 3.3 installed, but it was
>>>> compiled without TLS support.
>>>>
>>>> OpenSSL Installation: I have confirmed that OpenSSL is installed and
>>>> working correctly on my system. The OpenSSL libraries and headers are
>>>> available, and I have verified their installation with the following
>>>> commands:
>>>>
>>>> 1. *openssl* *version* shows:
>>>> OpenSSL 1.1.1k FIPS 25 Mar 2021
>>>>
>>>> 2. The headers are present in /*usr/include/openss*
>>>>
>>>> 3. The libraries are located in /*usr/lib*, including *libssl* and
>>>> *libcrypto*
>>>>
>>>>
>>>> *Compilation Attempt:*
>>>>  I set the environment variables for OpenSSL:
>>>>
>>>> *export CFLAGS="-I/usr/include/openssl"*
>>>>
>>>> *export LDFLAGS="-L/usr/lib -lssl -lcrypto"*
>>>>
>>>> After setting these variables, I ran *make* *clean* followed by *make*
>>>> in the SIPp directory, but the issue persists.
>>>>
>>>> *Error Message:* When I attempt to use TLS with SIPp, it gives the
>>>> message: “To use a TLS transport you must compile SIPp with OpenSSL.”
>>>>
>>>> Request:
>>>>
>>>> I would appreciate any guidance or steps to ensure that SIPp is
>>>> compiled correctly with TLS support. If there are specific steps or
>>>> configurations needed to resolve this issue, please let me know.
>>>>
>>>> Thank you for your assistance.
>>>>
>>>>
>>> Hi,
>>> I think you skipped the instruction from
>>> https://github.com/SIPp/sipp?tab=readme-ov-file#building
>>> There is no need to export such variables.
>>> SIPp is built using cmake so you should do something like:
>>>
>>> cmake . -DUSE_SSL=1
>>> make
>>> make install
>>>
>>>
>>>
>>>
>>>
>>>
>>

Re: [Sipp-users] Assistance Required for Compiling SIPp with TLS Support

[Jafar S. <sap...@gm...>]

I will try installing the new version.
Thanks for your help.

On Sat, Aug 24, 2024, 3:40 AM Jafar Sarif <sap...@gm...> wrote:

> Hi,
> I am using SIPp 3.3V which uses makefile for its build system.
>
> On Sat, Aug 24, 2024, 3:25 AM mayamatakeshi <may...@gm...>
> wrote:
>
>>
>>
>> On Sat, Aug 24, 2024 at 6:44 AM Jafar Sarif <sap...@gm...> wrote:
>>
>>> I am writing to seek assistance with compiling SIPp to enable TLS
>>> support. I have successfully installed SIPp but encountered difficulties
>>> when attempting to compile it with TLS support.
>>>
>>> *Issue* *Details:*
>>>
>>> Current Installation: I have SIPp version 3.3 installed, but it was
>>> compiled without TLS support.
>>>
>>> OpenSSL Installation: I have confirmed that OpenSSL is installed and
>>> working correctly on my system. The OpenSSL libraries and headers are
>>> available, and I have verified their installation with the following
>>> commands:
>>>
>>> 1. *openssl* *version* shows:
>>> OpenSSL 1.1.1k FIPS 25 Mar 2021
>>>
>>> 2. The headers are present in /*usr/include/openss*
>>>
>>> 3. The libraries are located in /*usr/lib*, including *libssl* and
>>> *libcrypto*
>>>
>>>
>>> *Compilation Attempt:*
>>>  I set the environment variables for OpenSSL:
>>>
>>> *export CFLAGS="-I/usr/include/openssl"*
>>>
>>> *export LDFLAGS="-L/usr/lib -lssl -lcrypto"*
>>>
>>> After setting these variables, I ran *make* *clean* followed by *make*
>>> in the SIPp directory, but the issue persists.
>>>
>>> *Error Message:* When I attempt to use TLS with SIPp, it gives the
>>> message: “To use a TLS transport you must compile SIPp with OpenSSL.”
>>>
>>> Request:
>>>
>>> I would appreciate any guidance or steps to ensure that SIPp is compiled
>>> correctly with TLS support. If there are specific steps or configurations
>>> needed to resolve this issue, please let me know.
>>>
>>> Thank you for your assistance.
>>>
>>>
>> Hi,
>> I think you skipped the instruction from
>> https://github.com/SIPp/sipp?tab=readme-ov-file#building
>> There is no need to export such variables.
>> SIPp is built using cmake so you should do something like:
>>
>> cmake . -DUSE_SSL=1
>> make
>> make install
>>
>>
>>
>>
>>
>>
>

Re: [Sipp-users] Assistance Required for Compiling SIPp with TLS Support

[Jafar S. <sap...@gm...>]

Hi,
I am using SIPp 3.3V which uses makefile for its build system.

On Sat, Aug 24, 2024, 3:25 AM mayamatakeshi <may...@gm...> wrote:

>
>
> On Sat, Aug 24, 2024 at 6:44 AM Jafar Sarif <sap...@gm...> wrote:
>
>> I am writing to seek assistance with compiling SIPp to enable TLS
>> support. I have successfully installed SIPp but encountered difficulties
>> when attempting to compile it with TLS support.
>>
>> *Issue* *Details:*
>>
>> Current Installation: I have SIPp version 3.3 installed, but it was
>> compiled without TLS support.
>>
>> OpenSSL Installation: I have confirmed that OpenSSL is installed and
>> working correctly on my system. The OpenSSL libraries and headers are
>> available, and I have verified their installation with the following
>> commands:
>>
>> 1. *openssl* *version* shows:
>> OpenSSL 1.1.1k FIPS 25 Mar 2021
>>
>> 2. The headers are present in /*usr/include/openss*
>>
>> 3. The libraries are located in /*usr/lib*, including *libssl* and
>> *libcrypto*
>>
>>
>> *Compilation Attempt:*
>>  I set the environment variables for OpenSSL:
>>
>> *export CFLAGS="-I/usr/include/openssl"*
>>
>> *export LDFLAGS="-L/usr/lib -lssl -lcrypto"*
>>
>> After setting these variables, I ran *make* *clean* followed by *make*
>> in the SIPp directory, but the issue persists.
>>
>> *Error Message:* When I attempt to use TLS with SIPp, it gives the
>> message: “To use a TLS transport you must compile SIPp with OpenSSL.”
>>
>> Request:
>>
>> I would appreciate any guidance or steps to ensure that SIPp is compiled
>> correctly with TLS support. If there are specific steps or configurations
>> needed to resolve this issue, please let me know.
>>
>> Thank you for your assistance.
>>
>>
> Hi,
> I think you skipped the instruction from
> https://github.com/SIPp/sipp?tab=readme-ov-file#building
> There is no need to export such variables.
> SIPp is built using cmake so you should do something like:
>
> cmake . -DUSE_SSL=1
> make
> make install
>
>
>
>
>
>

Re: [Sipp-users] Assistance Required for Compiling SIPp with TLS Support

[mayamatakeshi <may...@gm...>]

On Sat, Aug 24, 2024 at 6:44 AM Jafar Sarif <sap...@gm...> wrote:

> I am writing to seek assistance with compiling SIPp to enable TLS support.
> I have successfully installed SIPp but encountered difficulties when
> attempting to compile it with TLS support.
>
> *Issue* *Details:*
>
> Current Installation: I have SIPp version 3.3 installed, but it was
> compiled without TLS support.
>
> OpenSSL Installation: I have confirmed that OpenSSL is installed and
> working correctly on my system. The OpenSSL libraries and headers are
> available, and I have verified their installation with the following
> commands:
>
> 1. *openssl* *version* shows:
> OpenSSL 1.1.1k FIPS 25 Mar 2021
>
> 2. The headers are present in /*usr/include/openss*
>
> 3. The libraries are located in /*usr/lib*, including *libssl* and
> *libcrypto*
>
>
> *Compilation Attempt:*
>  I set the environment variables for OpenSSL:
>
> *export CFLAGS="-I/usr/include/openssl"*
>
> *export LDFLAGS="-L/usr/lib -lssl -lcrypto"*
>
> After setting these variables, I ran *make* *clean* followed by *make* in
> the SIPp directory, but the issue persists.
>
> *Error Message:* When I attempt to use TLS with SIPp, it gives the
> message: “To use a TLS transport you must compile SIPp with OpenSSL.”
>
> Request:
>
> I would appreciate any guidance or steps to ensure that SIPp is compiled
> correctly with TLS support. If there are specific steps or configurations
> needed to resolve this issue, please let me know.
>
> Thank you for your assistance.
>
>
Hi,
I think you skipped the instruction from
https://github.com/SIPp/sipp?tab=readme-ov-file#building
There is no need to export such variables.
SIPp is built using cmake so you should do something like:

cmake . -DUSE_SSL=1
make
make install

[Sipp-users] Assistance Required for Compiling SIPp with TLS Support

[Jafar S. <sap...@gm...>]

I am writing to seek assistance with compiling SIPp to enable TLS support.
I have successfully installed SIPp but encountered difficulties when
attempting to compile it with TLS support.

*Issue* *Details:*

Current Installation: I have SIPp version 3.3 installed, but it was
compiled without TLS support.

OpenSSL Installation: I have confirmed that OpenSSL is installed and
working correctly on my system. The OpenSSL libraries and headers are
available, and I have verified their installation with the following
commands:

1. *openssl* *version* shows:
OpenSSL 1.1.1k FIPS 25 Mar 2021

2. The headers are present in /*usr/include/openss*

3. The libraries are located in /*usr/lib*, including *libssl* and
*libcrypto*


*Compilation Attempt:*
 I set the environment variables for OpenSSL:

*export CFLAGS="-I/usr/include/openssl"*

*export LDFLAGS="-L/usr/lib -lssl -lcrypto"*

After setting these variables, I ran *make* *clean* followed by *make* in
the SIPp directory, but the issue persists.

*Error Message:* When I attempt to use TLS with SIPp, it gives the message:
“To use a TLS transport you must compile SIPp with OpenSSL.”

Request:

I would appreciate any guidance or steps to ensure that SIPp is compiled
correctly with TLS support. If there are specific steps or configurations
needed to resolve this issue, please let me know.

Thank you for your assistance.

Best regards,
Jafar

Re: [Sipp-users] Help with scenario

[Šindelka P. <sin...@tt...>]

I had a look through my scenarios and I was always only sending REFER so 
far, never expecting it, but I don't think it should be treated in some 
special way.


Can you post the .pcap, please? From the presence of [peer_tag_param] in 
your scenario I assume that it is a mid-dialog REFER, so maybe there is 
something wrong in the REFER message itself.


Pavel


Dne 18.06.2024 v 14:53 Tom Johnson napsal(a):
> I am using:
>
> <recv request="REFER" />
> <send>
> <![CDATA[
> SIP/2.0 202 Accepted
> Via: SIP/2.0/TCP [remote_ip]:5060;branch=[branch]
> From: <sip:91...@ps...>[peer_tag_param]
> To: +14253244587 
> <sip:4253244587@[local_ip]:5060;isup-oli=62>;tag=[call_number]
> Call-ID: [call_id]
> CSeq: 102 REFER
> Content-Length: 0
> Server: INdigital-inesrp-1.26.6
>
> ]]>
> </send>
>
> However, from Wireshark, I see "603 Decline" and sipp does not get 
> past the waiting for REFER step.   Can't seem to find any examples of 
> receiving a "refer".  Any help would be greatly appreciated.
>
>
>
> _______________________________________________
> Sipp-users mailing list
> Sip...@li...
> https://lists.sourceforge.net/lists/listinfo/sipp-users
-- 

*Ing. Pavel Šindelka
* senior specialista

TTC MARCONI s. r. o.
Třebohostická 987/5, 100 00  Praha 10
+420 234 051 712, +420 602 355 199
sin...@tt..., www.ttc-marconi.com

[Sipp-users] Help with scenario

[Tom J. <TJo...@mi...>]

I am using:

<recv request="REFER" />
<send>
<![CDATA[
SIP/2.0 202 Accepted
Via: SIP/2.0/TCP [remote_ip]:5060;branch=[branch]
From: <sip:91...@ps...>[peer_tag_param]
To: +14253244587 <sip:4253244587@[local_ip]:5060;isup-oli=62>;tag=[call_number]
Call-ID: [call_id]
CSeq: 102 REFER
Content-Length: 0
Server: INdigital-inesrp-1.26.6

]]>
</send>

However, from Wireshark, I see "603 Decline" and sipp does not get past the waiting for REFER step.   Can't seem to find any examples of receiving a "refer".  Any help would be greatly appreciated.

[Sipp-users] Issues with SIPP Call transfer

[Chandrasekaran A. <amb...@al...>]

Dear all

Good day !

We have the following Scenario is not working

SIPP 1 user calling SIPP2  via PBX and Automated Attendant and Automated Attendant is transfer the call to Pbx and it will reach the SIPP2 extension which was registered with Pabx.

SIPP1 ---->  PBX---> (Automated Attendant ) -- > Transfer the call to Pabx--->  SIPP - 2 Extension is not working

It seems the call leg is not matching . If you have any xml file matching this scenario please send to me.

Thanks

Best regards,

Chandrasekaran Ambalavanan

[Sipp-users] SIP trunk

[GUILLAUME Ely-A. <ely...@sp...>]

Hello, I'm having trouble creating a SIP trunk between my SIPp server and a Mitel MICC system. Do you have an idea of something I could be doing wrong?

Ely-Anne GUILLAUME
Architecte DWP
Tél : +33 635 20 06 55
ely...@sp...<mailto:ely...@sp...>

SPIE ICS
Dir. Innovation Design et Expertise
DAIDF
148 avenue Pierre Brossolette
92240 MALAKOFF
www.spie-ics.com<http://www.spie-ics.com/>

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[Sipp-users] SIPp messages logged are not terminated by end of line ?

[Chaigneau, N. <nic...@ca...>]

Hello,


I have a question related to SIPp logging.
It seems that messages logged in the SIPp error file are not terminated by an end of line character.
This makes it difficult to read or parse.

See example below:

The following events occurred:
2023-11-09      14:46:52.564911 1699541212.564911: Call-Id: (...), receive timeout on message (...) UAC:1 without label to jump to (ontimeout attribute): aborting call2023-11-09     14:46:52.565879 1699541212.565879: Dead call 1-4148690@10.118.21.108 (aborted at index 1), received 'SIP/2.0 302 Moved Temporarily^M
(...)
Content-Length: 0^M
^M
'2023-11-09     14:46:53.564003 1699541213.564003: Call-Id:(...)



In this example, the corresponding code is in call.cpp at line 2054:

                WARNING("Call-Id: %s, receive timeout on message %s:%d without label to jump to (ontimeout attribute): aborting call",
                        id, curmsg->desc, curmsg->index);

I don't see a \n at the end of the log message (or for any call to WARNING or ERROR functions).
Shouldn't there be one ?

Or should the logging function add a \n ?


Regards,
Nicolas.

This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

Re: [Sipp-users] Building sipp with open ssl on windows10

[Ihor O. <igo...@gm...>]

Ozgur,

You can try built it inside WSL on Windows. Or docker. Not really native
way, but definitely something that will work

Le lun. 16 oct. 2023 à 09:29, özgür pektaş <ozg...@ho...> a
écrit :

> Hi all,
>
> I've been trying to use sipp with authentication but it fails due to
> openssl dependency. İ tried 3.1 and 3.2 executable installation versions
> but it seems open sll was not built in it. Then i installed cygwin64
> followed instructions from here
>
> https://siptestingknowledge.blogspot.com/2013/12/installing-sipp-on-windows.html?m=1
>
> And tried newer releases of sipp but I've never managed to get it working.
> İ highly appreciate if anyone help me in this.
>
> Thanks
> _______________________________________________
> Sipp-users mailing list
> Sip...@li...
> https://lists.sourceforge.net/lists/listinfo/sipp-users
>


-- 
Best regards,
Ihor (Igor)

[Sipp-users] Building sipp with open ssl on windows10

[özgür p. <ozg...@ho...>]

Hi all,

I've been trying to use sipp with authentication but it fails due to openssl dependency. İ tried 3.1 and 3.2 executable installation versions but it seems open sll was not built in it. Then i installed cygwin64 followed instructions from here
https://siptestingknowledge.blogspot.com/2013/12/installing-sipp-on-windows.html?m=1

And tried newer releases of sipp but I've never managed to get it working. İ highly appreciate if anyone help me in this.

Thanks

[Sipp-users] IMS Bench SIPp

[Andrej M. <xm...@st...>]

<html><body><div>Hi.<br><br></div><div>I am trying to install IMS Bench SIPp on Linux operating system following the instructions at https://sipp.sourceforge.net/ims_bench/reference.html#Installation<br>with command:</div><div><i>svn co https://sipp.svn.sourceforge.net/svnroot/sipp/sipp/branches/ims_bench ims_bench<br></i><br></div><div>But the command failed with the message <i>"Repository moved permanently to 'https://sourceforge.net/projects/sipp/'; please relocate "</i></div><div><i><br></i>I could not find another source of the IMS Bench SIPp program.<br>Thanks for help.</div><div>Andy</div></body></html>