Thank You, Rob

Now it works. Your hypothesis was correct.

Thank you a lot for the support and congratulation for the intuition.


Massimo


2014-03-28 11:08 GMT+01:00 Rob Day <[email protected]>:

> Massimo,
>
> Could you try using the -mi parameter to set the media IP address
> explicitly? I wonder if it's incorrectly defaulting to 127.0.0.1 or an
> IPv6 address.
>
> Best,
> Rob
>
> > On 27 March 2014 09:21, Massimo Santo <[email protected]> wrote:
> >>
> >> Il giorno 25/mar/2014 12:07, "Massimo Santo" <[email protected]>
> ha
> >> scritto:
> >>
> >>
> >>>
> >>> Rob,
> >>>
> >>> I installed version 3.4.1 and I repeated the test.
> >>>
> >>> This is what -v option returns:
> >>>
> >>> SIPp v3.4.1-SCTP-PCAP-RTPSTREAM built Mar 25 2014, 11:32:59.
> >>>
> >>> I started sipp with this command
> >>>
> >>> sudo sipp 192.168.5.2 -sf uac_pcapTEST.xml -r 1 -rp 10s -p 6082 -rsa
> >>> 192.168.5.2:6082 -trace_err
> >>>
> >>> The RTP problem is still present and this is the errlog output (is the
> >>> same of the previous SIPP version):
> >>>
> >>> sipp: The following events occured:
> >>> 2014-03-25 11:55:56.154920    1395744956.154920: send_packets.c: sendto
> >>> failed with error: Invalid argument.
> >>>
> >>>
> >>> I also attach the xml file content:
> >>>
> >>>
> >>> <?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.
> >>> -->
> >>>
> >>> <!--
> >>>   <recv request="REGISTER">
> >>>   </recv>
> >>> -->
> >>>
> >>>   <send retrans="500">
> >>>     <![CDATA[
> >>>
> >>>
> >>>       INVITE sip:[email protected]:6082 SIP/2.0
> >>>       Via: SIP/2.0/UDP 192.168.5.165:6082;branch=[branch]
> >>>       From:"5002" <sip:[email protected]:6082>;tag=[call_number]
> >>>       To: "Alpha" <sip:[email protected]:6082>
> >>>       Call-ID: [call_id]
> >>>       CSeq: 1 INVITE
> >>>       Contact: sip:[email protected]:6082
> >>>       Max-Forwards: 70
> >>>       Subject: Performance Test
> >>>       Content-Type: application/sdp
> >>>       Content-Length: [len]
> >>>       Priority: normal
> >>>       External-Enc: 0
> >>>       External-Cg:5002
> >>>       Service: group
> >>>       Source-Gw: 64250
> >>>       External-Cd: 6001
> >>>
> >>>       v=0
> >>>       o=user1 53655765 2353687637 IN IP4 192.168.5.165
> >>>       s=-
> >>>       c=IN IP[media_ip_type] [media_ip]
> >>>       t=0 0
> >>>       m=audio 4024 RTP/AVP 0 8
> >>>       a=rtpmap:0 PCMU/8000
> >>>       a=rtpmap:8 PCMU/8000
> >>>
> >>>     ]]>
> >>>   </send>
> >>>
> >>>   <recv response="100" optional="true">
> >>>   </recv>
> >>>
> >>>   <recv response="180" 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:sip:[email protected]:6082 SIP/2.0
> >>>       Via: SIP/2.0/UDP 192.168.5.165:6082;branch=[branch]
> >>>       From: "5002" <sip:[email protected]:6082>;tag=[call_number]
> >>>       To: "Alpha" <sip:[email protected]:6082>[peer_tag_param]
> >>>       Call-ID: [call_id]
> >>>       CSeq: 1 ACK
> >>>       Contact: sip:[email protected]:6082
> >>>       Max-Forwards: 70
> >>>       Subject: Performance Test
> >>>       Content-Length: 0
> >>>
> >>>     ]]>
> >>>   </send>
> >>>
> >>>   <!-- Play a pre-recorded PCAP file (RTP stream)
> >>> -->
> >>>
> >>>   <nop>
> >>>     <action>
> >>>       <exec play_pcap_audio="rtp4.pcap"/>
> >>>     </action>
> >>>   </nop>
> >>>
> >>>
> >>>   <!-- Pause 8 seconds, which is approximately the duration of the
> >>> -->
> >>>   <!-- PCAP file        -->
> >>>
> >>>
> >>>   <pause milliseconds="8000"/>
> >>>
> >>>   <!-- 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]:6082 SIP/2.0
> >>>       Via: SIP/2.0/UDP 192.168.5.165:6082;branch=[branch]
> >>>       From: "5002" <sip:[email protected]:6082>;tag=[call_number]
> >>>       To: "Alpha" <sip:[email protected]:6082>[peer_tag_param]
> >>>       Call-ID: [call_id]
> >>>       CSeq: 2 BYE
> >>>       Contact: sip:[email protected]:6082
> >>>       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>
> >>>
> >>>
> >>>
> >>> Thank You for your help.
> >>>
> >>> Regards,
> >>>
> >>> Massimo
> >>>
> >>>
> >>>
> >>> 2014-03-22 12:52 GMT+01:00 Massimo Santo <[email protected]>:
> >>>
> >>>> Thank you for the answer,
> >>>>
> >>>> I'm not in my lab today but, if I'm in right, I'm using 3.4.0
> downloaded
> >>>> from here:
> >>>>
> >>>> https://github.com/SIPp/sipp/releases
> >>>>
> >>>> I'll udpdate on tuesday.
> >>>>
> >>>> I'd like to add another detail:
> >>>>
> >>>> In my scenario I perform just one call, so the issue is not related to
> >>>> high traffic load and it immediatly rises at first RTP transmission
> attempt.
> >>>>
> >>>> I'll update you with the 3.4.1 errlog.
> >>>>
> >>>>
> >>>> Massimo
> >>>>
> >>>>
> >>>> 2014-03-22 12:11 GMT+01:00 Rob Day <[email protected]>:
> >>>>
> >>>>> Hi Massimo,
> >>>>>
> >>>>> Are you running SIPp v3.4.1 (which you can get from
> >>>>> https://github.com/SIPp/sipp/releases/tag/v3.4.1)? I improved the
> >>>>> logging of this error in that release - running with that should give
> >>>>> a better indication of what's going on (e.g. why the RTP send is
> >>>>> failing).
> >>>>>
> >>>>> Best,
> >>>>> Rob
> >>>>>
> >>>>> On 21 March 2014 00:37, Massimo Santo <[email protected]>
> wrote:
> >>>>> > Hi,
> >>>>> >
> >>>>> > I installed the current SIPp Version on ubuntu.
> >>>>> >
> >>>>> > I used SIPp to generate invite towards our device and it correctly
> >>>>> > works.
> >>>>> >
> >>>>> > When I try to play a pcap as non root user, I record the "Can't
> create
> >>>>> > raw
> >>>>> > socket (need to run as root?)." error.
> >>>>> >
> >>>>> > Unfortunately, If I use the sudo command, SIPp crashes immediatly
> >>>>> > after
> >>>>> > completing SIP signalling.
> >>>>> >
> >>>>> > The errlog file reports  sendto failed with error: Invalid
> argument.
> >>>>> >
> >>>>> > Note: Our device has a particular behaviour. I.e. it uses an IP
> >>>>> > address to
> >>>>> > manage SIP signalling and a different IP to manage RTP streams.
> >>>>> >
> >>>>> > Can you help me to fix this issue?
> >>>>> >
> >>>>> > Thanks,
> >>>>> >
> >>>>> > Massimo
> >>>>> >
> >>>>> >
> >>>>> >
> >>>>> >
> >>>>> >
> ------------------------------------------------------------------------------
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> >>>>> > applications. Written by three acclaimed leaders in the field,
> >>>>> > this first edition is now available. Download your free book today!
> >>>>> > http://p.sf.net/sfu/13534_NeoTech
> >>>>> > _______________________________________________
> >>>>> > Sipp-users mailing list
> >>>>> > [email protected]
> >>>>> > https://lists.sourceforge.net/lists/listinfo/sipp-users
> >>>>> >
> >>>>
> >>>>
> >>>
> >
> >
> >
> > --
> > Robert K. Day
> > [email protected]
>
>
> ------------------------------------------------------------------------------
> _______________________________________________
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