you can throw this in the dustbin.
ACK was missing Content-Type: application/sdp
As I said : trivial ... :-)
2017-10-11 15:54 GMT+02:00 Johan De Clercq <jo...@democon.be>:
> 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>
>
>
>
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