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 <bjbrashier@gmail.com>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 < anthony.minessale@gmail.com> 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 <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
_______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400