Hi Pavel, Thanks for replying, Please find the my scenario below,
UAS Side:
SIPp UAS IMS Server
| |
| REGISTER |
|------------------------------>|
| 401 Unauthorized |
|<------------------------------|
| REGISTER |
|------------------------------>|
| 200 OK |
|<------------------------------|
Post registration below the scenario,
IMS Server SIPp UAS
|(1) INVITE |
|------------------>|
|(2) 180 |
|<------------------|
|(3) 200 |
|<------------------|
|(4) ACK |
|------------------>|
| |
|(5) PAUSE |
| |
|(6) BYE |
|------------------>|
|(7) 200 |
|<------------------|
*UAC Side:*
SIPp UAC IMS Server
| |
| REGISTER |
|------------------------------>|
| 401 Unauthorized |
|<------------------------------|
| REGISTER |
|------------------------------>|
| 200 OK |
|<------------------------------|
| |
Post registration below the scenario,
SIPp UAC IMS server
|(1) INVITE |
|------------------>|
|(2) 100 (optional) |
|<------------------|
|(3) 180 (optional) |
|<------------------|
|(4) 200 |
|<------------------|
|(5) ACK |
|------------------>|
| |
|(6) RTP send (8s) |
|==================>|
| |
|(7) RFC2833 DIGIT 1|
|==================>|
| |
|(8) BYE |
|------------------>|
|(9) 200 |
|<------------------|
Please find the attached message & error logs, let me know any other
details required.
Thanks,
Mohanraj
On Wed, Jul 31, 2019 at 2:16 AM Šindelka Pavel <sinde...@ttc.cz> wrote:
> scenarios not shown, error message from the UAS not shown, packet capture
> from the UAS side not shown... hard to say what's wrong.
>
> P.
> Dne 30.7.2019 v 21:37 Mohanraj S napsal(a):
>
> Hi,
>
> I'm trying to make end to end call in Clearwater IMS environment using
> SIPp, but I'm getting Error “Discarding message which can't be mapped to a
> known SIPp call” on SIPp UAS endpoint for the incoming INVITE from SIPp
> Client(UAC).
> But when i use Jitsi as UAS i'm successful in making end to end call.
> Could someone help here.
>
> UAS Command:
> "sipp 34.68.201.203 -sf uas.xml -m 1 -t t1 -trace_msg -trace_err"
>
> UAS Output:
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
> Call-rate(length) Port Total-time Total-calls Remote-host
> 10.0(0 ms)/1.000s 5062 184.33 s 1 34.68.201.203:5060(TCP)
>
> Call limit reached (-m 1), 1.002 s period 1 ms scheduler resolution
> 1 calls (limit 120) Peak was 1 calls, after 0 s
> 0 Running, 2 Paused, 3 Woken up
> 0 dead call msg (discarded) 1 out-of-call msg (discarded)
> 4 open sockets
>
> Messages Retrans Timeout Unexpected-Msg
> REGISTER ----------> 1 0 0
> 100 <---------- 0 0 0 0
> 401 <---------- 1 0 0 0
> REGISTER ----------> 1 0 0
> 100 <---------- 0 0 0 0
> 200 <---------- 1 0 0 0
> INVITE <---------- 0 0 0 0
>
> 180 ----------> 0 0
> 200 ----------> 0 0 0
> ACK <---------- E-RTD1 0 0 0 0
>
> BYE <---------- 0 0 0 0
> 200 ----------> 0 0
> Pause [ 4000ms] 0 0
>
> ------- Waiting for active calls to end. Press [q] again to force exit.
> -------
>
> Last Error: Discarding message which can't be mapped to a known SIPp cal...
>
> UAC command:
> "sipp 34.68.201.203 -sf sippCall.xml -inf sippCall.csv -m 1 -t t1
> -trace_msg -trace_err"
>
> UAC Output:
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
> Call-rate(length) Port Total-time Total-calls Remote-host
> 10.0(0 ms)/1.000s 5063 34.27 s 1 34.68.201.203:5060(TCP)
>
> Call limit reached (-m 1), 0.000 s period 0 ms scheduler resolution
> 0 calls (limit 960) 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)
> 0 open sockets
>
> Messages Retrans Timeout Unexpected-Msg
> REGISTER ----------> 1 0 0
> 100 <---------- 0 0 0 0
> 401 <---------- 1 0 0 0
> REGISTER ----------> 1 0 0
> 100 <---------- 0 0 0 0
> 200 <---------- 1 0 0 0
> Pause [ 2000ms] 1 0
> INVITE ----------> 1 0 0
> 100 <---------- 1 0 0 0
> 180 <---------- 0 0 0 1
> 200 <---------- E-RTD1 0 0 0 0
>
> ACK ----------> 0 0
> [ NOP ]
> Pause [ 30.0s] 0 0
> BYE ----------> 0 0 0
> 200 <---------- 0 0 0 0
>
> ------------------------------ Test Terminated
> --------------------------------
>
> ----------------------------- Statistics Screen ------- [1-9]: Change
> Screen --
> Start Time | 2019-07-30 19:04:11.633714 1564513451.633714
> Last Reset Time | 2019-07-30 19:04:45.905139 1564513485.905139
> Current Time | 2019-07-30 19:04:45.905300 1564513485.905300
>
> -------------------------+---------------------------+--------------------------
> Counter Name | Periodic value | Cumulative value
>
> -------------------------+---------------------------+--------------------------
> Elapsed Time | 00:00:00:000000 | 00:00:34:271000
> Call Rate | 0.000 cps | 0.029 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 | 00:00:00:000000 | 00:00:00:000000
> Call Length | 00:00:00:000000 | 00:00:34:153000
> ------------------------------ Test Terminated
> --------------------------------
>
> 2019-07-30 19:04:45.891481 1564513485.891481: Aborting call on unexpected
> message for Call-Id '1-21223@172.18.0.4': while expecting '180' (index
> 10), received 'SIP/2.0 408 Request Timeout
> Via: SIP/2.0/TCP 172.18.0.4:5063
> ;rport=39243;received=10.16.0.1;branch=z9hG4bK-21223-1-8
> Record-Route:
> sip:sprout.default.svc.cluster.local:5054;transport=tcp;lr;billing-role=charge-term
> Record-Route:
> sip:sprout.default.svc.cluster.local:5054;transport=tcp;lr;billing-role=charge-orig
> Record-Route: sip:10.16.0.12:5058;transport=TCP;lr
> Record-Route: sip:sQtRDAKzv5@34.68.201.203:5060;transport=TCP;lr
> Call-ID: 1-21223@172.18.0.4
> From: sip:6505550...@default.svc.cluster.local;tag=11234
> To:
> sip:6505550...@default.svc.cluster.local;tag=z9hG4bKPj5cKdnkBC3Kd39-k4kWZi6LecMQAhnESt
> CSeq: 1 INVITE
> Content-Length: 0
>
>
> Regards,
>
> Mohanraj
>
> --
>
>
> *Ing. Pavel Šindelka *senior specialista
>
>
>
>
>
> TTC MARCONI s. r. o.
> Třebohostická 987/5, 100 00 Praha 10
> +420 234 051 712, +420 602 355 199
> sinde...@ttc.cz, www.ttc-marconi.com
>
>
<?xml version="1.0" encoding="ISO-8859-2" ?>
<!-- Use with CSV file struct like: 3000;192.168.1.106;[authentication username=3000 password=3000];
(user part of uri, server address, auth tag in each line)
-->
<scenario name="Register">
<User variables="my_id,bono_ip,peer_id,service" />
<nop hide="true">
<action>
<assignstr assign_to="my_id" value="[field0]" />
<!-- field1 is my_auth, but we can't store it in a variable -->
<assignstr assign_to="service" value="[field2]" />
<assignstr assign_to="bono_ip" value="[field3]" />
<assignstr assign_to="peer_id" value="[field4]" />
</action>
</nop>
<send retrans="500">
<![CDATA[
REGISTER sip:[$bono_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[$my_id]@[$service]>;tag=[call_number]
To: <sip:[$my_id]@[$service]>
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Contact: <sip:[$my_id]@[local_ip]:[local_port]>
Max-Forwards: 10
Expires: 120
User-Agent: SIPp/Win32
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE
Content-Length: 0
]]>
</send>
<!-- asterisk -->
<recv response="100" optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[$bono_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[$my_id]@[$service]>;tag=[call_number]
To: <sip:[$my_id]@[$service]>
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Contact: <sip:[$my_id]@[local_ip]:[local_port]>
[field1]
Max-Forwards: 10
Expires: 120
User-Agent: SIPp/Win32
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE
Content-Length: 0
]]>
</send>
<!-- asterisk -->
<recv response="100" optional="true">
</recv>
<recv response="200">
</recv>
<pause milliseconds="2000"/>
<send retrans="500">
<![CDATA[
INVITE sip:[$peer_id]@[$service]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
From: sip:[$my_id]@[$service]:[local_port];tag=[call_number]1234
To: sip:[$peer_id]@[$service]:[remote_port]
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[$my_id]@[local_ip]:[local_port];transport=[transport]
Route: <sip:[$service]:[remote_port];lr>
Max-Forwards: 70
Subject: Performance Test
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,PRACK
Content-Type: application/sdp
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Accession 4.0.0.0
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rrs="true" rtd="true" crlf="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[$peer_id]@[$service]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sip:[$my_id]@[$service]:[local_port];tag=[call_number]1234
To: sip:[$peer_id]@[$service]:[remote_port][peer_tag_param]
[routes]
Call-ID: [call_id]
CSeq: 1 ACK
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!-- Pause 8 seconds, which is approximately the duration of the -->
<!-- PCAP file -->
<pause milliseconds="30000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[$peer_id]@[$service]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
[routes]
From: sip:[$my_id]@[$service]:[local_port];tag=[call_number]1234
To: sip:[$peer_id]@[$service]:[remote_port][peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:[$my_id]@[local_ip]:[local_port];transport=[transport]
Max-Forwards: 10
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- 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>
UAC_Call_21223_messages.log
Description: Binary data
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder">
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurences are concatenated (CRLF seperated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send retrans="500">
<![CDATA[
REGISTER sip:34.68.201.203 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:6505550...@default.svc.cluster.local>;tag=[call_number]
To: <sip:6505550...@default.svc.cluster.local>
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Contact: <sip:6505550...@default.svc.cluster.local:[local_port]>
Max-Forwards: 10
Expires: 120
User-Agent: SIPp/Win32
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
]]>
</send>
<!-- asterisk -->
<recv response="100" optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:34.68.201.203 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:6505550...@default.svc.cluster.local>;tag=[call_number]
To: <sip:6505550...@default.svc.cluster.local>
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Contact: <sip:6505550287@[local_ip]:[local_port]>
[authentication username=6505550...@default.svc.cluster.local password=MQP3psFfX]
Max-Forwards: 10
Expires: 120
User-Agent: SIPp/Win32
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
]]>
</send>
<!-- asterisk -->
<recv response="100" optional="true">
</recv>
<recv response="200">
</recv>
<recv request="INVITE" crlf="true" rrs="true">
</recv>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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 request="ACK"
optional="true"
rtd="true"
crlf="true">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<pause milliseconds="4000"/>
<!-- 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>
UAC_Call_21223_errors.log
Description: Binary data
uas_21212_errors.log
Description: Binary data
uas_21212_messages.log
Description: Binary data
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