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
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