|
Hi, I wrote a script which is supposed to simulate an endpoint which is the first callee in 3pcc flow 2 (attached). My problem is that the re-invite sent to me is recognized by sipp as an initial invite, and therefore as a new call. I want it to be handled in the same call, how do I do it?
<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. --> <recv request="INVITE" crlf="true"> </recv>
<!-- 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> <![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];lr;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];lr;transport=[transport]> Content-Type: application/sdp Content-Length: 136
v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- c=IN IP4 [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>
<!-- receive callback reinvite --> <recv request="INVITE" crlf="true"> </recv>
<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];lr;transport=[transport]> Content-Type: application/sdp Content-Length: 136
v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- c=IN IP4 [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"> <action> <ereg regexp="sip:.*:[0-9]{1,5}" search_in="hdr" header="Contact:" check_it="true" assign_to="1"/> </action> </recv>
<send retrans="500"> <![CDATA[
BYE [$1] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number] To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:[EMAIL PROTECTED]:[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>
Hagai.
|
------------------------------------------------------------------------- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
_______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
