From: Deng P. <de...@se...> - 2005-07-08 13:40:55
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I noticed the -p option, but it can only assign one local_port. Does that mean I need to write a shell script to start the 1000 uas with fixed local_port? I thought there should be a more elegant way to do this. =20 Deng Pan =20 -----Original Message----- From: Peter Higginson [mailto:pet...@ne...]=20 Sent: Friday, July 08, 2005 2:58 AM To: Deng Pan; sip...@li... Subject: RE: [Sipp-users] uas question =20 There is a command line option -p which you can use to set a fixed port at the server. =20 Peter Higginson Newport Networks Ltd, Direct line 01494 470694 http://www.newport-networks.com/ ________________________________ From: sip...@li... [mailto:sip...@li...] On Behalf Of Deng Pan Sent: 08 July 2005 04:56 To: sip...@li... Subject: [Sipp-users] uas question =20 Hi, I am using sipp as uac and uas to test a sip proxy server, and I have some problems with uas. What I want is to create 1000 uac and 1000 uas, which communicate through my sip proxy server, to emulate 1000 concurrent calls. Since the local_port of uas is randomly assigned, my proxy has no way to know its local_port (to route packets) unless it registers to the proxy. However, if the uas registers first in the scenario, it will not respond to the INVITE request. My uas.xml is as follows: <?xml version=3D"1.0" encoding=3D"ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name=3D"My UAS responder"> <send retrans=3D"500"> <![CDATA[ REGISTER sip:[field1]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=3D[call_number] To: [field1] <sip:[field1]@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 REGISTER Contact: sip:[field1]@[local_ip]:[local_port] Max-Forwards: 70 Expires: 6000 Content-Length: 0 ]]> </send> <recv response=3D"200" rtd=3D"true" optional=3D"true"> </recv> <recv request=3D"INVITE" crlf=3D"true"> </recv> <send> <![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=3D[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=3D[transport]> Content-Length: 0 ]]> </send> <send retrans=3D"500"> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=3D[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=3D[transport]> Content-Type: application/sdp Content-Length: 136 v=3D0 o=3Duser1 53655765 2353687637 IN IP4 127.0.0.1 s=3D- c=3DIN IP4 [media_ip] t=3D0 0 m=3Daudio [media_port] RTP/AVP 0 a=3Drtpmap:0 PCMU/8000 ]]> </send> <recv request=3D"ACK" optional=3D"true" rtd=3D"true" crlf=3D"true"> </recv> <recv request=3D"BYE"> </recv> <send> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=3D[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=3D[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=3D"4000"/> <!-- definition of the response time repartition table (unit is ms) --> <ResponseTimeRepartition value=3D"10, 20, 30, 40, 50, 100, 150, 200"/> <!-- definition of the call length repartition table (unit is ms) --> <CallLengthRepartition value=3D"10, 50, 100, 500, 1000, 5000, 10000"/> </scenario> |