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 >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------------ >>>>> > 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 >>>>> > >>>> >>>> >>> > > > > -- > Robert K. Day > [email protected] ------------------------------------------------------------------------------ _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
