Hi Tuan,
1) Can you take a capture and verify if the ACK has successfuly
reached the IMS
(which generates 200 OK)?
Thanks and Regards,
Reuben
On 12/7/10, tran quoc tuan <[email protected]> wrote:
> 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:[[email protected]>;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
>
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