[Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

[Michael Jerris]

I beleive this is following the right rfc rules for dialog matching.   
If it is not, please open up a bug on jira.freeswitch.org with  
references of what exactly is not right.

Mike

On Aug 25, 2009, at 2:51 AM, Tihomir Culjaga <tculjaga at gmail.com> 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:[service]@[remote_ip]:[remote_port] SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>       Max-Forwards: 70
>       Contact: <sip:[field1]@[local_ip]>
>       From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag= 
> [call_number]
>       To: [service] <sip:[service]@[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:[service]@[remote_ip]:[remote_port] SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch= 
> [branch-3]
>       From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag= 
> [call_number]
>       To: [service] <sip:[service]@[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 <mayamatakeshi at gmail.com 
> > wrote:
> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com 
> > wrote:
> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir  
> Culjaga<tculjaga at 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
> >
>
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