Hi all,
I have tested SIPp client to connect IMS, But im my scenario, it has worked
but client and server continue to transfer messages 200 OK SDP and ACK.
I don't find the reason .
Messages Retrans Timeout
Unexpected-Msg
REGISTER ----------> 1 0 0
401 <---------- 1 0 0 0
REGISTER ----------> 1 0 0
200 <---------- 1 0 0 0
Pause [ 500ms] 1 0
INVITE ----------> 1 0 0
100 <---------- 1 0 0 0
180 <---------- 1 0 0 0
403 <---------- 0 0 0 0
404 <---------- 0 0 0 0
408 <---------- 0 0 0 0
200 <---------- 1 9 0 0
ACK ----------> 1 9
Pause [ 500ms] 1 0
[ NOP ]
Pause [ 30.0s] 1 0
REGISTER ----------> 1 0 0
401 <---------- 1 0 0 0
REGISTER ----------> 1 0 0
200 <---------- 1 0 0 0
BYE <---------- 1 0 0 0
200 ----------> 1 0 0
This is my scenario :
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Session for conference">
<send retrans="500">
<![CDATA[
REGISTER sip:[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 20
From: "[field0]" <sip:[fiel...@[field2]>;tag=[call_number]
To: "[field0]" <sip:[fiel...@[field2]>
P-Access-Network-Info:
3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 7200
Content-Length: [len]
User-Agent: Sipp v1.1-TLS, version 20061124
Authorization: Digest username="[fiel...@[field2]", realm="[field2]"
Supported: path
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Route: [$1]
Max-Forwards: 20
From: "[field0]" <sip:[fiel...@[field2]>;tag=[call_number]
To: "[field0]" <sip:[fiel...@[field2]>
P-Access-Network-Info:
3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
Call-ID: [call_id]
CSeq: 2 REGISTER
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 7200
Content-Length: 0
User-Agent: Sipp v1.1-TLS, version 20061124
[field3]
Supported: path
]]>
</send>
<recv response="200">
<action>
<ereg regexp=".*" search_in="hdr" header="Service-Route:"
assign_to="1" />
</action>
</recv>
<pause milliseconds="500" crlf="true" />
<send retrans="500">
<![CDATA[
INVITE sip:sip-servlets-confere...@[remote_ip]:5080 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 20
Route: <sip:pcscf.open-ims.test:4060;lr>,[$1]
P-Preferred-Identity: <sip:[fiel...@[field2]>
Privacy: none
P-Access-Network-Info:
3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
From: <sip:[fiel...@[field2]>;tag=[call_number]
To: <sip:[servi...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 7200
User-Agent: Sipp v1.1-TLS, version 20061124
Allow: ACK, BYE, CANCEL, INVITE, REFER, OPTIONS, INFO, REGISTER,
NOTIFY, UPDATE, SUBSCRIBE, PRACK
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 3487063231 3487063231 IN IP[local_ip_type] [local_ip]
s=SJphone
c=IN IP[media_ip_type] [media_ip]
t=0 0
a=direction:active
m=audio [auto_media_port] RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="403" optional="true" next="1">
</recv>
<recv response="404" optional="true" next="1">
</recv>
<recv response="408" optional="true" next="1">
</recv>
<recv response="200" rrs="true">
</recv>
<send crlf="true" >
<![CDATA[
ACK [next_url] SIP/2.0
[last_Via:]
Max-Forwards: 20
[routes]
From: <sip:[fiel...@open-ims.test>;tag=[call_number]
To: <sip:[servi...@[remote_ip]:5080>
Call-ID: [call_id]
CSeq: [cseq] ACK
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<pause milliseconds="500" crlf="true" />
<nop>
<action>
<exec play_pcap_audio="8Khz.pcap"/>
</action>
</nop>
<pause milliseconds="30000" crlf="true" />
<send retrans="500">
<![CDATA[
REGISTER sip:[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 20
From: "[field0]" <sip:[fiel...@[field2]>;tag=[call_number]
To: "[field0]" <sip:[fiel...@[field2]>
P-Access-Network-Info:
3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
Call-ID: [call_id]
CSeq: 3 REGISTER
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 0
Content-Length: 0
User-Agent: Sipp v1.1-TLS, version 20061124
Authorization: Digest username="[fiel...@[field2]", realm="[field2]"
Supported: path
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Route: [$1]
Max-Forwards: 20
From: "[field0]" <sip:[fiel...@[field2]>;tag=[call_number]
To: "[field0]" <sip:[fiel...@[field2]>
P-Access-Network-Info:
3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
Call-ID: [call_id]
CSeq: 4 REGISTER
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 0
Content-Length: 0
User-Agent: Sipp v1.1-TLS, version 20061124
[field3]
Supported: path
]]>
</send>
<recv response="200" >
</recv>
<recv request="BYE">
</recv>
<send retrans="500" crlf="true">
<![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>
<label id="1"/>
<label id="2"/>
<!-- 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>
Thank in advanced for all helps!
B.R
T.Q.Tuan
------------------------------------------------------------------------------
What happens now with your Lotus Notes apps - do you make another costly
upgrade, or settle for being marooned without product support? Time to move
off Lotus Notes and onto the cloud with Force.com, apps are easier to build,
use, and manage than apps on traditional platforms. Sign up for the Lotus
Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
