Hi Michael;
Xml in annex.
My csv file has the following format:
+351xxxxxxxxx;[authentication username=+351xxxxxxxxx password=xxxxxxx];user
domain;expire_time;called_number;
This xml Registers a user and then sends an INVITE without SDP (but I also used
it in scneario without Register).
BR
[cid:image001.png@01D0875B.E7EFA3C0]<http://www.telecom.pt/>
Alberto Manuel Rolim Valente
DEI - Direção de Engenharia Implementacao Rede e Plataformas
alberto.r.vale...@telecom.pt
[cid:image002.png@01D0875B.E7EFA3C0]<http://www.facebook.com/portugaltelecom>[cid:image003.png@01D0875B.E7EFA3C0]<http://twitter.com/portugaltelecom>[cid:image004.png@01D0875B.E7EFA3C0]<http://videos.sapo.pt/portugaltelecom>[cid:image005.png@01D0875B.E7EFA3C0]<http://www.youtube.com/portugaltelecom>[cid:image006.png@01D0875B.E7EFA3C0]<http://www.telecom.pt/InternetResource/PTSite/PT/RSS>
From: Azer, Michael (Michael) [mailto:ma...@avaya.com]
Sent: terça-feira, 5 de Maio de 2015 15:25
To: Alberto Valente; sipp-users@lists.sourceforge.net
Subject: RE: Ivite without SDP
Hi,
Can you send me your XML, because I think that the problem is not related to
ACK.
The Wireshark showing that after getting 200 message the SIPp is replying
immediately with ICMP port unreachable.
But, you are right, I missed this line and added it .
After adding this line, it is still not working.
Thanks,
Michael
From: Alberto Valente [mailto:alberto-r-vale...@telecom.pt]
Sent: Tuesday, May 05, 2015 4:15 PM
To: Azer, Michael (Michael) **CTR**;
sipp-users@lists.sourceforge.net<mailto:sipp-users@lists.sourceforge.net>
Subject: RE: Ivite without SDP
Hi;
I have a scenario similar to yours that Works. The only difference I can see is
that you're missing the Content-Type Header in the ACK reply (
Content-Type: application/sdp)
"My" ACK is as follows:
ACK sip:[field4]@[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]@[field2]>;tag=[call_number]
To: <sip:[field4]@[field2]>[peer_tag_param]
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
[last_Call-ID]
Cseq: [cseq] ACK
Content-Type: application/sdp
Content-Length: [len]
v=0
o=CiscoSystemsSIP-GW-UserAgent 2342 5515 IN IP[local_ip_type] [local_ip]
s=SIP Call
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 18 101
c=IN IP[media_ip_type] [media_ip]
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
BR
[cid:image001.png@01D0875B.E7EFA3C0]<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.telecom.pt_&d=AwMFAw&c=BFpWQw8bsuKpl1SgiZH64Q&r=Qlh7iU-afcg2NgiTv4ti9w&m=izY26WJsAszxpKV2gFCXN3SsvkECmZjxZT9Xa3k__Rc&s=CwaO14wXoX2KVsc4IalWhZ21O3XzvQQqUlrBSkGXHLg&e=>
Alberto Manuel Rolim Valente
alberto.r.vale...@telecom.pt<mailto:alberto.r.vale...@telecom.pt>
[cid:image002.png@01D0875B.E7EFA3C0]<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.facebook.com_portugaltelecom&d=AwMFAw&c=BFpWQw8bsuKpl1SgiZH64Q&r=Qlh7iU-afcg2NgiTv4ti9w&m=izY26WJsAszxpKV2gFCXN3SsvkECmZjxZT9Xa3k__Rc&s=CeOOhilQhaOeT2gDcQ-1qK8L4bmXv40Fuzas5jGJSnc&e=>[cid:image003.png@01D0875B.E7EFA3C0]<https://urldefense.proofpoint.com/v2/url?u=http-3A__twitter.com_portugaltelecom&d=AwMFAw&c=BFpWQw8bsuKpl1SgiZH64Q&r=Qlh7iU-afcg2NgiTv4ti9w&m=izY26WJsAszxpKV2gFCXN3SsvkECmZjxZT9Xa3k__Rc&s=vBoD-nipKZModL1ydYITZ3lLtbsr5k_ryQgawK8sFKc&e=>[cid:image004.png@01D0875B.E7EFA3C0]<https://urldefense.proofpoint.com/v2/url?u=http-3A__videos.sapo.pt_portugaltelecom&d=AwMFAw&c=BFpWQw8bsuKpl1SgiZH64Q&r=Qlh7iU-afcg2NgiTv4ti9w&m=izY26WJsAszxpKV2gFCXN3SsvkECmZjxZT9Xa3k__Rc&s=u7qajS7tSz_TrcguQ6Zw6FPk5oX0CudnJBTdHs59W5g&e=>[cid:image005.png@01D0875B.E7EFA3C0]<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.youtube.com_portugaltelecom&d=AwMFAw&c=BFpWQw8bsuKpl1SgiZH64Q&r=Qlh7iU-afcg2NgiTv4ti9w&m=izY26WJsAszxpKV2gFCXN3SsvkECmZjxZT9Xa3k__Rc&s=2CzoNdeS2cBSS299ZytqxXb3bWTCa7C7dIVaI2GLYYM&e=>[cid:image006.png@01D0875B.E7EFA3C0]<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.telecom.pt_InternetResource_PTSite_PT_RSS&d=AwMFAw&c=BFpWQw8bsuKpl1SgiZH64Q&r=Qlh7iU-afcg2NgiTv4ti9w&m=izY26WJsAszxpKV2gFCXN3SsvkECmZjxZT9Xa3k__Rc&s=Kyh4dQoOPfH6Djk8XMsRbE0oj8gPClQNF4albz15NqU&e=>
From: Azer, Michael (Michael) [mailto:ma...@avaya.com]
Sent: terça-feira, 5 de Maio de 2015 11:30
To: sipp-users@lists.sourceforge.net<mailto:sipp-users@lists.sourceforge.net>
Subject: [Sipp-users] Ivite without SDP
Hi,
I am trying to build the following script:
1. Sending Invite without SDP
2. Waiting to receive 100
3. Waiting to receive 180
4. Waiting to recive 200 +SDP
The problem is that SIPP is sending ICMP port unreachable after getting 200+
SDP (see attached sniffer file)
This is the XML which are using:
<send retrans="500">
- <![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From:[field0]<sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact:sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<recv response="100" optional="true" />
<recv response="180" />
<recv response="200" crlf="true" timeout="10000" />
- <!-- 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.
-->
- <!-- 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:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From:[field0]<sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact:sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: [len]
v=0
o=SIPp 187223291 6 IN IP4 [local_ip]
s=-
c=IN IP4 [local_ip]
b=AS:1920
t=0 0
m=audio [media_port] RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
]]>
Best Regards,
Michael Azer
[cid:image007.jpg@01D0875B.E7EFA3C0]
+972.3.767.9372 office | +972.549701544 mobile |
www.AVAYA.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.AVAYA.com&d=AwMFAw&c=BFpWQw8bsuKpl1SgiZH64Q&r=Qlh7iU-afcg2NgiTv4ti9w&m=izY26WJsAszxpKV2gFCXN3SsvkECmZjxZT9Xa3k__Rc&s=ueR7pZp8PTdeZoFQVy6WQ7Ou_ce5gy6Wv-Z-mCpGTYo&e=>
<?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="Cenário de INVITE / Re-INVITE sem SDP">
<!-- Começamos por registar um cliente para podermos efectuar a chamada -->
<send retrans="500">
<![CDATA[
REGISTER sip:[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
To: <sip:[field0]@[field2]:[remote_port]>
From: <sip:[field0]@[field2]:[remote_port]>;tag=1
Contact: "[field0]" <sip:[field0]@[local_ip]:[local_port]>;transport=[transport]
CSeq: [cseq] REGISTER
Call-ID: [call_id]
Expires: [field3]
]]>
</send>
<!-- Já está registado, esquece a autenticação -->
<recv response="200" optional="true" next="1" >
</recv>
<!-- Recebe TRYING -->
<recv response="100" optional="true">
</recv>
<!-- Recebe o 401 ao REGISTER enviado -->
<recv response="401" auth="true">
</recv>
<!-- Agora vamos enviar o REGISTER com os parâmetros de autenticação -->
<send retrans="500">
<![CDATA[
REGISTER sip:[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
To: <sip:[field0]@[field2]:[remote_port]>
From: <sip:[field0]@[field2]:[remote_port]>;tag=1
Contact: "[field0]" <sip:[field0]@[local_ip]:[local_port]>;transport=[transport]
[field1]
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Expires: [field3]
]]>
</send>
<!-- Recebe TRYING de volta! -->
<recv response="100" optional="true">
</recv>
<!-- Recebe 200 OK, Yes! -->
<recv response="200"
crlf="true">
</recv>
<label id="1"/>
<!-- Agora vamos ver se conseguimos fazer uma chamada. O número a chamar ! -->
<!-- é colocado no campo ficheiro CSV. O INVITE é enviado sem SDP ! -->
<send retrans="500">
<![CDATA[
INVITE sip:[field4]@[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[field0]"<sip:[field0]@[field2]>;tag=[call_number]
To: "[field4]"<sip:[field4]@[field2]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: "[field0]"<sip:[field0]@[local_ip]:[local_port]>
Max-Forwards: 70
Allow: INVITE,ACK,OPTIONS,BYE,PRACK,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Subject: Performance Test
]]>
</send>
<!-- Agora podemos receber várias mensagens -->
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<recv response="200" rtd="true">
</recv>
<!-- Enviamos ACK ao 200 OK (com SDP) -->
<send>
<![CDATA[
ACK sip:[field4]@[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]@[field2]>;tag=[call_number]
To: <sip:[field4]@[field2]>;[peer_tag_param]
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
[last_Call-ID]
Cseq: [cseq] ACK
Content-Type: application/sdp
Content-Length: [len]
v=0
o=CiscoSystemsSIP-GW-UserAgent 2342 5515 IN IP[local_ip_type] [local_ip]
s=SIP Call
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 18 101
c=IN IP[media_ip_type] [media_ip]
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]]>
</send>
<!-- Mantemos a chamada activa durante alguns segundos. Entretanto -->
<!-- enviamos umas tramas SDP -->
<nop>
<action>
<exec play_pcap_audio="pcap/fccn_ssrc_okay.pcap"/>
</action>
</nop>
<pause milliseconds="30000"/>
<!-- Agora, seguindo o comportamento do Montepio enviamos um INVITE -->
<!-- com SDP e media atribbute. Inactive -->
<send>
<![CDATA[
BYE sip:[field4]@[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[field0]"<sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: "+351[field4]" <sip:+351[field4]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: [cseq] BYE
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
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>
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