here's the one i use for making a call waiting x seconds and hanging up http://www.freeswitch.org/eg/load_test/dft_cap.xml
This requires that the sipp terminate all the calls. careful with sipp, it's like a roach motel, you can get stuck trying to make it work and never get it to produce real-life situations. On Tue, Aug 25, 2009 at 3:23 PM, Bradley Brashier <[email protected]>wrote: > Well, you'd have another nickel from over here, then. > If I can get this working before I'm tasked with something else I'll write > up something more on the wiki about "Freeswitch and SIPp", but I'm not sure > I'll get that chance. > > BB > > On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale < > [email protected]> wrote: > >> I wish I had a nickel for every guy struggling with sipp load testing vs >> real world traffic. >> >> >> >> On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <[email protected]>wrote: >> >>> Hello Takeshi, >>> >>> Thanks for your hint... it worked out... so to be precise: >>> >>> VIA header of both INVITE and ACK messages MUST be identical (IP:PORT + >>> branch)... and you are right... it might not be according to SIP >>> specification. Anyhow, i get FS understand my ACK message. >>> >>> >>> Finally, here is what i used and I'm getting some poor results .. but >>> this is another topic :) >>> >>> >>> Thanks for your help. >>> Tihomir. >>> >>> >>> sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 >>> -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000 >>> >>> >>> <?xml version="1.0" encoding="ISO-8859-1" ?> >>> <!DOCTYPE scenario SYSTEM "sipp.dtd"> >>> >>> >>> <scenario name="Basic Sipstone UAC"> >>> <send retrans="500" start_rtd="1" start_rtd="2"> >>> >>> <![CDATA[ >>> >>> INVITE sip:[servi...@[remote_ip]:[remote_port] SIP/2.0 >>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] >>> Max-Forwards: 70 >>> Contact: <sip:[fiel...@[local_ip]> >>> From: [field1] >>> <sip:[fiel...@[local_ip]:[local_port]>;tag=[call_number] >>> To: [service] <sip:[servi...@[remote_ip]:[remote_port]> >>> Call-ID: [call_id] >>> CSeq: 1 INVITE >>> Content-Type: application/sdp >>> Content-Length: [len] >>> >>> v=0 >>> o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] >>> s=- >>> c=IN IP[media_ip_type] [media_ip] >>> t=0 0 >>> m=audio [media_port] RTP/AVP 0 >>> a=rtpmap:0 PCMU/8000 >>> >>> ]]> >>> </send> >>> >>> <recv response="100" >>> optional="true" rtd="1"> >>> </recv> >>> >>> >>> <recv response="302" rtd="2"> >>> </recv> >>> >>> <send> >>> <![CDATA[ >>> >>> ACK sip:[servi...@[remote_ip]:[remote_port] SIP/2.0 >>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] >>> From: [field1] >>> <sip:[fiel...@1[local_ip]:[local_port]>;tag=[call_number] >>> To: [service] >>> <sip:[servi...@[remote_ip]:[remote_port]>[peer_tag_param] >>> Call-ID: [call_id] >>> CSeq: 1 ACK >>> Max-Forwards: 70 >>> 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 Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi >>> <[email protected]>wrote: >>> >>>> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<[email protected]> >>>> wrote: >>>> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<[email protected]> >>>> wrote: >>>> >> >>>> >> sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s >>>> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt >>>> -m 1 -l >>>> >> 4000 >>>> >> scenario file: uac_redirect.xml >>>> >> FS dialplan: public.xml >>>> >> SIP trace: trace.log >>>> > >>>> > The Via definition in your SIPp scenario differs between the INVITE >>>> and the ACK: >>>> > >>>> > INVITE: >>>> > Via: SIP/2.0/[transport] [local_ip];branch=[branch] >>>> > >>>> > ACK: >>>> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] >>>> > >>>> > >>>> > In the INVITE, you are not adding the [local_port] as you do in the >>>> ACK. >>>> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the >>>> ACK. >>>> > So it seems FS is not checking just the ACK's branch against the >>>> > INVITE's; it seems it is checking the whole Via header. >>>> > I don't know if this is in accordance to SIP specs. >>>> > Another thing, about the way you are calling SIPp: do no use "-sn uac" >>>> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx" >>>> > means "use the internal (embedded) scenario named xxx". So this >>>> > conflicts with the other parameter "-sf" which specifies an external >>>> > profile. >>>> >>>> I mean, an external scenario (file). >>>> >>>> It seems this doesn't cause any problem (probably because in >>>> > the sipp startup, -sf overrides -sn), but it is misleading. >>>> > >>>> > regards, >>>> > takeshi >>>> > >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> [email protected] >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> [email protected] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:[email protected] <msn%[email protected]> >> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:[email protected] <sip%[email protected]> >> iax:[email protected]/888 >> googletalk:[email protected]<googletalk%3aconf%[email protected]> >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> [email protected] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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