Hi,
no I haven´t used wireshark, yet. Just watched the
http://sipp.sourceforge.net/doc/uas.xml and
http://sipp.sourceforge.net/doc/uac_pcap.xml.html for the first time.
I thought those were complete SIP Calls, i.e. REGISTER, INVITE, BYE, ...
But they aren´t :-( Several steps have to be done:
{1. SIPP-Server registers
2. SIPP-Client registers
3. SIPP-Client invites SIPP-Server
3. SIPP Client sends the prerecorded file to the SIPP-Server
4. SIPP-Server echoes back what SIPP-Client formerly sent.
(5. SIPP-Client sends again, SIPP-Server replies again)
6. Session will be ended.
}* x-times
I hope this is everything, do I forget something?
Btw, are there anywhere more premade scenarios, or do I have to make it
on my own? (Haven´t done this yet before, so that would be quite long...)
Christian
Simon Flannery schrieb:
>
> Are you tracing using ethereal / wireshark? This will help with the
> capturing the unexpected messages. If you are tracing, what are the
> unexpected messages? Also, silly question, have you first registered
> the users and are they successfully registered?
>
> Simon
>
> On 8/15/07, *Christian Kraus* <[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>> wrote:
>
> Hi,
>
> I want to test my Asterisk-Server. Therefore I want to use the
> included
> uac_pcap.xml.
> I´m posting my configs and only changing the IPs. Hopefully anyone
> could
> help, as I can´t find something googling for hours/days.
>
>
> I set up:
>
> SIPP-Client: 89.111.111.111 <http://89.111.111.111>
> SIPP-Server: 89.222.222.222 <http://89.222.222.222>
> Asterisk: 89.555.555.555
>
>
> on SIPP-Server I run:
> ./sipp -sn uas -i 89.222.222.222 <http://89.222.222.222>
> 89.222.222.222 <http://89.222.222.222>
>
> and an SIPP-Client:
> ./sipp -sf uac_pcap.xml -d 100000 -s 2000 89.555.555.555 -l 1000 -r 10
> -rp 1000
>
>
> To the asterisk´s sip.conf I added:
>
> [sipp]
> context=test
> language=en
> type=friend
> host=dynamic
> nat=no
>
> To the extension.conf I added:
>
> [test]
> exten => 2000,1,Dial(SIP/89.222.222.222)
>
> So every caller being 2000, will be directed to the SIPP-Server...
>
>
>
> As you can see the SIPP-Client is sending, but it gets "unexpected
> messages":
>
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
> Call-rate(length) Port Total-time Total-calls Remote-host
> 10.0(100000 ms)/1.000s 5060 28.01 s 280 89.555.555.555:5060(UDP)
>
> 10 new calls during 1.000 s period 2 ms scheduler resolution
> 127 calls (limit 1000) Peak was 141 calls, after 14 s
> 0 Running, 128 Paused, 10 Woken up
> 0 out-of-call msg (discarded)
> 1 open sockets
> 0 Total RTP pckts sent 0.000 last period RTP rate (kB/s)
>
> Messages Retrans Timeout Unexpected-Msg
> INVITE ----------> 280 657 0
> 100 <---------- 0 0 0 153
> 180 <---------- 0 0 0 0
> 200 <---------- E-RTD1 0 0 0 0
>
> ACK ----------> 0 0
> [ NOP ]
> Pause [ 8000ms] 0 0
> [ NOP ]
> Pause [ 1000ms] 0 0
> BYE ----------> 0 0 0
> 200 <---------- 0 0 0 0
>
> ------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause
> traffic -----
>
>
>
>
> The SIPP-Server is receiving messages and sends answers back:
>
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
> Port Total-time Total-calls Transport
> 5060 35.02 s 209 UDP
>
> 10 new calls during 1.001 s period 2 ms scheduler resolution
> 164 calls Peak was 165 calls, after 30 s
> 0 Running, 164 Paused, 0 Woken up
> 1 open sockets
>
> Messages Retrans Timeout Unexpected-Msg
> ----------> INVITE 209 0 0
>
> <---------- 180 209 0
> <---------- 200 209 0 0
> ----------> ACK E-RTD1 209 0 0
>
> ----------> BYE 85 0 0
> <---------- 200 85 0
> [ 4000ms] Pause 85 0
> ------------------------------ Sipp Server Mode
> -------------------------------
>
>
>
>
> But as you can see on the SIPP-Client´s output, there are no
> succesfull
> calls. I logged everything and get the following failure types:
>
> sipp: The following events occured:
> 2007-08-15 16:38:16:570 1187188696.570840: Aborting call on unexpected
> message for Call-Id '[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>': while$
> Via: SIP/2.0/UDP
> 127.0.0.1:5060
> <http://127.0.0.1:5060>;branch=z9hG4bK-5633-96-0;received=89.110.157.78
> <http://89.110.157.78>
> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=96
> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as52a75494
> Call-ID: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> 2007-08-15 16:38:17:568 1187188697.568788: Aborting call on unexpected
> message for Call-Id '[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>': whil$
> Via: SIP/2.0/UDP
> 127.0.0.1:5060
> <http://127.0.0.1:5060>;branch=z9hG4bK-5633-106-0;received=89.110.157.78
> <http://89.110.157.78>
> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=106
> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as6eb78372
> Call-ID: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 02007-08-15 16:38:17:568 1187188697.568788: Aborting
> call on unexpected message for Call-Id '106-5633$
> Via: SIP/2.0/UDP
> 127.0.0.1:5060
> <http://127.0.0.1:5060>;branch=z9hG4bK-5633-106-0;received=
> 89.110.157.78 <http://89.110.157.78>
> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=106
> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as6eb78372
> Call-ID: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
>
>
>
>
> What is going wrong? What can I do to get correct sip calls running?
> It would be great if someone could help me.
>
> thanks
>
> Christian
>
>
>
>
>
>
>
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