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 <mayamatake...@gmail.com>wrote: > On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatake...@gmail.com> > wrote: > > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculj...@gmail.com> > 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 > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
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