Have anyone subj? I`m look to RFC and see one, but when i`m looking to sip trace I see another... And it works on SIP-clients, but not on my scenario (in attach) <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="UAC"> <!-- In client mode (sipp placing calls), the Call-ID MUST be --> <!-- generated by sipp. To do so, use [call_id] keyword. --> <send retrans="500" start_txn="invite"> <![CDATA[ INVITE sip:[field3]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number] To: [field3] <sip:[field3]@[field1]> Call-ID: [call_id] CSeq: [cseq] INVITE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=[field0] 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" response_txn="invite"/> <recv response="100" optional="true" response_txn="invite"/> <recv response="407" auth="true" response_txn="invite"/> <send ack_txn="invite"> <![CDATA[ ACK sip:[field3]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number] To: [field3] <sip:[field3]@[field1]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: [cseq] ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: [len] v=0 o=[field0] 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> <send retrans="500" start_txn="invite"> <![CDATA[ INVITE sip:[field3]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number] To: [service] <sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] INVITE Contact: sip:sipp@[local_ip]:[local_port] [field2] Max-Forwards: 70 Subject: Performance Test 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 response="100" optional="true" response_txn="invite"> </recv> <recv response="100" optional="true" response_txn="invite"> </recv> <recv response="183" optional="true" response_txn="invite"/> <!-- 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 response="200" rtd="true" response_txn="invite"> </recv> <!-- Packet lost can be simulated in any send/recv message by --> <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --> <send ack_txn="invite"> <![CDATA[ ACK sip:[field3]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number] To: [field3] <sip:[field3]@[field1]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: [cseq] ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <pause milliseconds="59000"/> <send retrans="500" start_txn="bye"> <![CDATA[ BYE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0 [last_Via:] [last_From:] [last_To:];tag=[pid][call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <recv response="200" crlf="true" response_txn="bye"/> <!-- <recv request="BYE"/> <send retrans="500"> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=[pid][call_number] [last_Call-ID:] [last_CSeq:] Max-Forwards: 70 Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Type: application/sdp Content-Length: 0 ]]> </send> --> <!-- 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> -- with best regards, Dmitriy Borisov