Hi, Jeff, I am following your instruction in my scenario. When the INVITE from calling party arrived to called party, the called party always said "ICMP Destination Unreachable (Port unreachable), and then in the SIPp of called party, it claimed no INVITE received all the time. Do you have any experience of that ? Are your both instances working as UAC mode ? Regards, Jun
_____ From: Jeff Wright [mailto:jwri...@azteknetworks.net] Sent: Friday, December 04, 2009 10:42 PM To: Wen Jun; sipp-users@lists.sourceforge.net Subject: RE: [Sipp-users] Sipx with SIPp Jun, I have used sipX a lot in testing our SIP-based products here. Some of our tests use SIPp as a UA. Here are two scenario files that, together, establish a basic call between two instances of SIPp (of course, this assumes you have already registered SIPp on both sides). These work with sipX as a proxy (I am using sipX 3.8.1-011585). I run them side by side in tow different terminal windows, and execute them one right after the other, running the B side first, then the A side (this way the B side script is ready to accept the INVITE sent from the A side script). You will also need to have the CSV file (also included in this email) in order to populate the information in the [field] parameters in the scripts. Here are the actual scenario files: <?xml version="1.0" encoding="ISO-8859-1" ?> <scenario name="successful_call_single_proxy_sideA"> <send> <![CDATA[ INVITE sip:[fiel...@[field4] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: <sip:[fiel...@[field1]>;tag=from-[call_number] To: <sip:[fiel...@[field4]> Call-ID: [call_id] CSeq: [cseq] INVITE Contact: sip:[fiel...@[local_ip]:[local_port] User-Agent: SIPp/Linux Subject: Test 3.2.1.1.7 Content-Type: application/sdp Content-Length: [len] v=0 o=sipp1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" optional="true"> </recv> <recv response="180" optional="true"> </recv> <recv response="200"> </recv> <send> <![CDATA[ ACK sip:[fiel...@[field4] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] [last_From:] [last_To:] [last_Call-ID:] CSeq: [cseq] ACK Contact: sip:[fiel...@[local_ip]:[local_port] User-Agent: SIPp/Linux Content-Length: 0 ]]> </send> <pause milliseconds="1000"/> <send> <![CDATA[ BYE sip:[fiel...@[field4] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] [last_From:] [last_To:] [last_Call-ID:] CSeq: [cseq] BYE Contact: sip:[fiel...@[local_ip]:[local_port] User-Agent: SIPp/Linux Max-Forwards: 70 Content-Length: 0 ]]> </send> <recv response="200"> </recv> </scenario> <?xml version="1.0" encoding="ISO-8859-1" ?> <scenario name="successful_call_single_proxy_sideB"> <recv request="INVITE"> </recv> <send> <![CDATA[ SIP/2.0 180 UA2 Ringing [last_Via:] [last_From:] [last_To:];tag=to-[call_number] [last_Call-ID:] [last_CSeq:] Contact: sip:[fiel...@[local_ip]:[local_port] User-Agent: SIPp/Linux Content-Length: 0 ]]> </send> <pause milliseconds="500"/> <send> <![CDATA[ SIP/2.0 200 OK UA2 Answered [last_Via:] [last_From:] [last_To:];tag=to-[call_number] [last_Call-ID:] [last_CSeq:] Contact: sip:[fiel...@[local_ip]:[local_port] User-Agent: SIPp/Linux Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv request="ACK"> </recv> <recv request="BYE"> </recv> <send> <![CDATA[ SIP/2.0 200 OK UA2 Goodbye [last_Via:] [last_From:] [last_To:] [last_Call-ID:] [last_CSeq:] Contact: sip:[fiel...@[local_ip]:[local_port] User-Agent: SIPp/Linux Content-Length: 0 ]]> </send> </scenario> Here is the CSV file. You will have to modify the individual fields to match your domain, and the SIPp usernames and passwords as set up in sipX. The domain (e.g. test.azteknetworks.net) has to be the same as what you have it set up for in sipX. SEQUENTIAL sipp1;test.azteknetworks.net;[authentication username=sipp1 password=sipp1];sipp2;test.azteknetworks.net;[authentication username=sipp2 password=sipp2] Best of luck! Jeffrey Wright System Test Engineering Manager Aztek Networks, Inc. _____ From: Wen Jun [mailto:jun.wen.s...@gmail.com] Sent: Fri 12/4/2009 2:18 AM To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] Sipx with SIPp Hi, it might be a frequently asked question about how to test SIPX with SIPp. Does any have successful experience about that ? Appreciated if you can share some hints to me . I've well done the registration from SIPp UAC to SIPX but the call from SIPp UAC to SIPX was struck yet. Regards, Jun
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