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 <tculjaga@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@gmail.com>wrote:
On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi@gmail.com> wrote:
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga@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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