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 >>> > >>> > >>> > >>> > ------------------------------------------------------------------------------ >>> > Learn Graph Databases - Download FREE O'Reilly Book >>> > "Graph Databases" is the definitive new guide to graph databases and their >>> > 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 >>> > >> >> >
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