Re: [Sipp-users] Suggestion for - IVR Scenario

sindelka Fri, 10 Mar 2017 09:51:05 -0800

Hi Robert,

1) the pcap you've attached does not match the error log, and neither ofthe two illustrates your original issue (404 response to BYE).

2) the pcap shows that the embedded UAC scenario cannot cope with whatyour SER (Freeswitch) send back to it. In the 200 OK, there are several|Record-Route| headers, so the UAC scenario has to mirror them as|Route| headers in the ACK and any subsequent requests it sends withinthat call, including the BYE - see the purpose of the |rrs| attribute of|recv| and the |[routes]| keyword in the SIPp manual. These are missingin the embedded UAC scenario, so the ACK doesn't make it to the actualUAS and thus the latter keeps re-transmitting the 200 until timeout;when the timeout expires, it attempts to actively terminate the dialogby sending a BYE which the UAC scenario doesn't expect to receive -instead, it intends to send its own one but later.

So if you want to move further, you have to dump the embedded UACscenario into a file, modify it the way suggested above, and try again.

3) from your original description, I'm quite sceptical regarding the"detect if there is audio coming back" part. SIPp does not have anembedded detection of RTP arrival, let alone a detector/analyzer of thatRTP's contents, so you'd have to glue one to it if you'd want to checkthat something else than silence is coming from the IVR.



Pavel


Dne 10.3.2017 v 17:41 Remsik,Robert napsal(a):

Sure!  Thank you for the help!  :)
For some reason this didn't post the several errors to the screen soI'm pasting what I've been seeing the in error logs
---- Errors ----
root@monitor:/home/zabbix/AutoCallMatrix/1 $ cat uac_30975_errors.log
sipp: The following events occured:
2017-03-1009:21:40:4201489162900.420444: Discarding message whichcan't be mapped to a known SIPp call:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.82.254.254:5061;branch=z9hG4bK-30930-1-0
Record-Route:<sip:129.82.254.202:5066;lr;sipXecs-CallDest=UNK%2CAA;sipXecs-rs=*auth~.*from~MzA5MzBTSVBwVGFnMDAx!083191711afa22cd1495fd22778d74c8>
Record-Route: <sip:129.82.254.200;transport=tcp;r2=on;lr;did=36c.3d9>
Record-Route: <sip:129.82.254.200;r2=on;lr;did=36c.3d9>
From: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
To: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
Call-Id: 1-30930@129.82.254.254
Cseq: 1 INVITE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 168
Remote-Party-Id: "IVR"<sip:IVR@129.82.254.200>;party=calling;privacy=off;screen=no
Date: Fri, 10 Mar 2017 16:21:41 GMT

v=0
o=FreeSWITCH 1489150845 1489150846 IN IP4 129.82.254.200
s=FreeSWITCH
c=IN IP4 129.82.254.200
t=0 0
m=audio 12048 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
.
2017-03-1009:21:44:4201489162904.420305: Discarding message whichcan't be mapped to a known SIPp call:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.82.254.254:5061;branch=z9hG4bK-30930-1-0
Record-Route:<sip:129.82.254.202:5066;lr;sipXecs-CallDest=UNK%2CAA;sipXecs-rs=*auth~.*from~MzA5MzBTSVBwVGFnMDAx!083191711afa22cd1495fd22778d74c8>
Record-Route: <sip:129.82.254.200;transport=tcp;r2=on;lr;did=36c.3d9>
Record-Route: <sip:129.82.254.200;r2=on;lr;did=36c.3d9>
From: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
To: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
Call-Id: 1-30930@129.82.254.254
Cseq: 1 INVITE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 168
Remote-Party-Id: "IVR"<sip:IVR@129.82.254.200>;party=calling;privacy=off;screen=no
Date: Fri, 10 Mar 2017 16:21:45 GMT

v=0
o=FreeSWITCH 1489150845 1489150846 IN IP4 129.82.254.200
s=FreeSWITCH
c=IN IP4 129.82.254.200
t=0 0
m=audio 12048 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
.
2017-03-1009:21:48:4201489162908.420240: Discarding message whichcan't be mapped to a known SIPp call:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.82.254.254:5061;branch=z9hG4bK-30930-1-0
Record-Route:<sip:129.82.254.202:5066;lr;sipXecs-CallDest=UNK%2CAA;sipXecs-rs=*auth~.*from~MzA5MzBTSVBwVGFnMDAx!083191711afa22cd1495fd22778d74c8>
Record-Route: <sip:129.82.254.200;transport=tcp;r2=on;lr;did=36c.3d9>
Record-Route: <sip:129.82.254.200;r2=on;lr;did=36c.3d9>
From: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
To: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
Call-Id: 1-30930@129.82.254.254
Cseq: 1 INVITE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 168
Remote-Party-Id: "IVR"<sip:IVR@129.82.254.200>;party=calling;privacy=off;screen=no
Date: Fri, 10 Mar 2017 16:21:49 GMT

v=0
o=FreeSWITCH 1489150845 1489150846 IN IP4 129.82.254.200
s=FreeSWITCH
c=IN IP4 129.82.254.200
t=0 0
m=audio 12048 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
.
2017-03-1009:21:52:4191489162912.419994: Discarding message whichcan't be mapped to a known SIPp call:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.82.254.254:5061;branch=z9hG4bK-30930-1-0
Record-Route:<sip:129.82.254.202:5066;lr;sipXecs-CallDest=UNK%2CAA;sipXecs-rs=*auth~.*from~MzA5MzBTSVBwVGFnMDAx!083191711afa22cd1495fd22778d74c8>
Record-Route: <sip:129.82.254.200;transport=tcp;r2=on;lr;did=36c.3d9>
Record-Route: <sip:129.82.254.200;r2=on;lr;did=36c.3d9>
From: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
To: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
Call-Id: 1-30930@129.82.254.254
Cseq: 1 INVITE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 168
Remote-Party-Id: "IVR"<sip:IVR@129.82.254.200>;party=calling;privacy=off;screen=no
Date: Fri, 10 Mar 2017 16:21:53 GMT

v=0
o=FreeSWITCH 1489150845 1489150846 IN IP4 129.82.254.200
s=FreeSWITCH
c=IN IP4 129.82.254.200
t=0 0
m=audio 12048 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
.
2017-03-1009:21:56:4201489162916.420085: Discarding message whichcan't be mapped to a known SIPp call:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.82.254.254:5061;branch=z9hG4bK-30930-1-0
Record-Route:<sip:129.82.254.202:5066;lr;sipXecs-CallDest=UNK%2CAA;sipXecs-rs=*auth~.*from~MzA5MzBTSVBwVGFnMDAx!083191711afa22cd1495fd22778d74c8>
Record-Route: <sip:129.82.254.200;transport=tcp;r2=on;lr;did=36c.3d9>
Record-Route: <sip:129.82.254.200;r2=on;lr;did=36c.3d9>
From: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
To: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
Call-Id: 1-30930@129.82.254.254
Cseq: 1 INVITE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 168
Remote-Party-Id: "IVR"<sip:IVR@129.82.254.200>;party=calling;privacy=off;screen=no
Date: Fri, 10 Mar 2017 16:21:57 GMT

v=0
o=FreeSWITCH 1489150845 1489150846 IN IP4 129.82.254.200
s=FreeSWITCH
c=IN IP4 129.82.254.200
t=0 0
m=audio 12048 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
.
2017-03-1009:22:00:4201489162920.420097: Discarding message whichcan't be mapped to a known SIPp call:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.82.254.254:5061;branch=z9hG4bK-30930-1-0
Record-Route:<sip:129.82.254.202:5066;lr;sipXecs-CallDest=UNK%2CAA;sipXecs-rs=*auth~.*from~MzA5MzBTSVBwVGFnMDAx!083191711afa22cd1495fd22778d74c8>
Record-Route: <sip:129.82.254.200;transport=tcp;r2=on;lr;did=36c.3d9>
Record-Route: <sip:129.82.254.200;r2=on;lr;did=36c.3d9>
From: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
To: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
Call-Id: 1-30930@129.82.254.254
Cseq: 1 INVITE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 168
Remote-Party-Id: "IVR"<sip:IVR@129.82.254.200>;party=calling;privacy=off;screen=no
Date: Fri, 10 Mar 2017 16:22:01 GMT

v=0
o=FreeSWITCH 1489150845 1489150846 IN IP4 129.82.254.200
s=FreeSWITCH
c=IN IP4 129.82.254.200
t=0 0
m=audio 12048 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
.
2017-03-1009:22:04:4211489162924.421232: Discarding message whichcan't be mapped to a known SIPp call:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 129.82.254.254:5061;branch=z9hG4bK-30930-1-0
Record-Route:<sip:129.82.254.202:5066;lr;sipXecs-CallDest=UNK%2CAA;sipXecs-rs=*auth~.*from~MzA5MzBTSVBwVGFnMDAx!083191711afa22cd1495fd22778d74c8>
Record-Route: <sip:129.82.254.200;transport=tcp;r2=on;lr;did=36c.3d9>
Record-Route: <sip:129.82.254.200;r2=on;lr;did=36c.3d9>
From: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
To: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
Call-Id: 1-30930@129.82.254.254
Cseq: 1 INVITE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 168
Remote-Party-Id: "IVR"<sip:IVR@129.82.254.200>;party=calling;privacy=off;screen=no
Date: Fri, 10 Mar 2017 16:22:05 GMT

v=0
o=FreeSWITCH 1489150845 1489150846 IN IP4 129.82.254.200
s=FreeSWITCH
c=IN IP4 129.82.254.200
t=0 0
m=audio 12048 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
.
2017-03-1009:22:04:9211489162924.921063: Discarding message whichcan't be mapped to a known SIPp call:
BYE sip:sipp@129.82.254.254:5061 SIP/2.0
Via: SIP/2.0/UDP129.82.254.200;branch=z9hG4bKf0dd.dcd64e02ac0542f5fe683cff18c426df.0;i=229Via: SIP/2.0/TCP129.82.254.202:5066;branch=z9hG4bK-XX-0589jM4`uQ9Y8Im2`gxJrQMhyAVia: SIP/2.0/UDP129.82.254.200:15060;rport=15060;branch=z9hG4bKNtFtQ3Q9HcXZK
Max-Forwards: 19
From: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
To: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
Call-Id: 1-30930@129.82.254.254
Cseq: 104245398 BYE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Reason: SIP;cause=408;text="ACK Timeout"
Content-Length: 0
Date: Fri, 10 Mar 2017 16:22:05 GMT

.
2017-03-1009:22:05:3851489162925.385709: Discarding message whichcan't be mapped to a known SIPp call:
BYE sip:sipp@129.82.254.254:5061 SIP/2.0
Via: SIP/2.0/UDP129.82.254.200;branch=z9hG4bKf0dd.dcd64e02ac0542f5fe683cff18c426df.0;i=229Via: SIP/2.0/TCP129.82.254.202:5066;branch=z9hG4bK-XX-0589jM4`uQ9Y8Im2`gxJrQMhyAVia: SIP/2.0/UDP129.82.254.200:15060;rport=15060;branch=z9hG4bKNtFtQ3Q9HcXZK
Max-Forwards: 19
From: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
To: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
Call-Id: 1-30930@129.82.254.254
Cseq: 104245398 BYE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Reason: SIP;cause=408;text="ACK Timeout"
Content-Length: 0
Date: Fri, 10 Mar 2017 16:22:05 GMT

.
2017-03-1009:22:06:3851489162926.385672: Discarding message whichcan't be mapped to a known SIPp call:
BYE sip:sipp@129.82.254.254:5061 SIP/2.0
Via: SIP/2.0/UDP129.82.254.200;branch=z9hG4bKf0dd.dcd64e02ac0542f5fe683cff18c426df.0;i=229Via: SIP/2.0/TCP129.82.254.202:5066;branch=z9hG4bK-XX-0589jM4`uQ9Y8Im2`gxJrQMhyAVia: SIP/2.0/UDP129.82.254.200:15060;rport=15060;branch=z9hG4bKNtFtQ3Q9HcXZK
Max-Forwards: 19
From: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
To: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
Call-Id: 1-30930@129.82.254.254
Cseq: 104245398 BYE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Reason: SIP;cause=408;text="ACK Timeout"
Content-Length: 0
Date: Fri, 10 Mar 2017 16:22:05 GMT

.
2017-03-1009:22:08:3851489162928.385672: Discarding message whichcan't be mapped to a known SIPp call:
BYE sip:sipp@129.82.254.254:5061 SIP/2.0
Via: SIP/2.0/UDP129.82.254.200;branch=z9hG4bKf0dd.dcd64e02ac0542f5fe683cff18c426df.0;i=229Via: SIP/2.0/TCP129.82.254.202:5066;branch=z9hG4bK-XX-0589jM4`uQ9Y8Im2`gxJrQMhyAVia: SIP/2.0/UDP129.82.254.200:15060;rport=15060;branch=z9hG4bKNtFtQ3Q9HcXZK
Max-Forwards: 19
From: 137 <sip:137@129.82.254.200:5060>;tag=N1QXacZ7Fr6mj
To: sipp <sip:sipp@129.82.254.254:5061>;tag=30930SIPpTag001
Call-Id: 1-30930@129.82.254.254
Cseq: 104245398 BYE
Contact: <sip:137@129.82.254.200:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Reason: SIP;cause=408;text="ACK Timeout"
Content-Length: 0
Date: Fri, 10 Mar 2017 16:22:05 GMT

.
2017-03-1009:22:11:4741489162931.474507: Aborting call on anunexpected BYE for call: 1-30975@129.82.254.254.
.:: Command ::.
sipp -sf /home/zabbix/AutoCallMatrix/1/uac.xml 129.82.254.200 -m 1 -au11012 -ap 24384 -s 137 -i 129.82.254.254 -timeout 90s -timeout_error-d 90000 -trace_screen -trace_err
------------------------------ Scenario Screen -------- [1-9]: ChangeScreen --
  Call-rate(length)   Port   Total-time  Total-calls  Remote-host
10.0(90000 ms)/1.000s 5061 32.30 s 1129.82.254.200:5060(UDP)
  Call limit reached (-m 1), 0.000 s period  0 ms scheduler resolution
  0 calls (limit 2700)                   Peak was 1 calls, after 0 s
  0 Running, 3 Paused, 0 Woken up
  0 dead call msg (discarded)            0 out-of-call msg (discarded)
  1 open sockets
Messages Retrans TimeoutUnexpected-Msg
      INVITE ---------->         1         0         0
         100 <----------         1         0         0       0
         180 <----------         0         0         0       0
         183 <----------         0         0         0       0
         200 <----------  E-RTD1 1         1         0       0
         ACK ---------->         1         1
       Pause [     1:30]         1     1
         BYE ---------->         0         0         0
         200 <----------         0         0         0       0
------------------------------ Test Terminated--------------------------------
----------------------------- Statistics Screen ------- [1-9]: ChangeScreen --
  Start Time             | 2017-03-1009:26:30:1601489163190.160205
  Last Reset Time        | 2017-03-1009:27:02:4691489163222.469667
  Current Time           | 2017-03-1009:27:02:4691489163222.469781
-------------------------+---------------------------+--------------------------
  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:00:000              | 00:00:32:309
  Call Rate              |    0.000 cps              |  0.031 cps
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |    0
  OutGoing call created  |        0                  |    1
  Total Call created     |                           |    1
  Current Call           |        0                  |
-------------------------+---------------------------+--------------------------
  Successful call        |        0                  |    0
  Failed call            |        0                  |    1
-------------------------+---------------------------+--------------------------
  Response Time 1        | 00:00:00:000              | 00:00:00:211
  Call Length            | 00:00:00:000              | 00:00:32:208
------------------------------ Test Terminated--------------------------------
2017-03-1009:27:02:4691489163222.469379: Aborting call on anunexpected BYE for call: 1-341@129.82.254.254.
Robert Remsik

ACNS

Desk Phone: 970 491 7120

robert.rem...@colostate.edu

------------------------------------------------------------------------
*From:* Sakharam Thorat <sakharam.tho...@outlook.com>
*Sent:* Thursday, March 9, 2017 6:27:54 PM
*To:* Remsik,Robert; sipp-users@lists.sourceforge.net
*Subject:* Re: Suggestion for - IVR Scenario

Can we have wireshark trace to debug further ?


Best Regards,
Sakharam Thorat.


------------------------------------------------------------------------
*From:* Remsik,Robert <robert.rem...@colostate.edu>
*Sent:* Friday, March 10, 2017 3:23 AM
*To:* sipp-users@lists.sourceforge.net
*Subject:* [Sipp-users] Suggestion for - IVR Scenario

Hello!
I'm trying to get a SIPP scenario that will have the client dial anumber (connected to an auto attendant), pick up, detect if there isaudio coming back, and then hang up after 10 seconds.
So far I've been using the basic 'uac.xml' scenario which gets it tocall, but when the client hangs up, the scenario expects a 200 ok butit's getting a 404 not found from the server. I'm unable to trackdown the origin of the 404 from the server side, so I'm wondering ifI'm using the wrong scenario file for sipp.
Is there a particular scenario file I should be using for this test-case?


Thank you in advance,

Robert




Robert Remsik

ACNS

Desk Phone: 970 491 7120

robert.rem...@colostate.edu



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account today to start using our lexical data to power your apps and
projects. Get started today and enter our developer competition.
http://sdm.link/oxford

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Re: [Sipp-users] Suggestion for - IVR Scenario

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