From: Johan De C. <jo...@de...> - 2017-10-11 14:22:58
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Hello, [image: Inline afbeelding 2] Using below script, when ACK is send out, the SDP is not understood (see image below). It might be that I am overlooking somehting trivial, but I can't spot it. Version used is 3.5 Best regards, <?xml version="1.0"?> <scenario name="3pcc"> <!-- In client mode (sipp placing calls), the Call-ID MUST be --> <!-- generated by sipp. To do so, use [call_id] keyword. --> <send retrans="500"> <![CDATA[ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 INVITE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Content-Type: application/sdp Content-Length: [len] ]]> </send> <recv response="100" optional="true"> </recv> <recv response="180" optional="true"> </recv> <recv response="183" optional="true"> </recv> <!-- 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" crlf="true"> </recv> <!-- Packet lost can be simulated in any send/recv message by --> <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --> <send> <![CDATA[ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Content-Length: [len] v=0 o=aastra400 695279534 695279535 IN IP4 192.168.1.25 s=- c=IN IP4 192.168.1.25 t=0 0 m=audio [media_port] RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sendrecv a=ptime:20 ]]> </send> <!-- Play a pre-recorded PCAP file (RTP stream) --> <nop> <action> <exec play_pcap_audio="/usr/src/sippy_cupscripts/simple_call2.pcap"/> </action> </nop> <!-- Pause 8 seconds, which is approximately the duration of the --> <!-- PCAP file --> <pause milliseconds="8000"/> <!-- The 'crlf' option inserts a blank line in the statistics report. --> <send retrans="500"> <![CDATA[ BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:sipp@[local_ip]:[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> |