Hey Guys,
I am Using sipp to simulate a Radiostation in an IP-Radio Environment.
As such a radio is compliant to the Eurocae standard it has to act like the
following:
---recv---> INVITE //Basestation connects to the radiochannel
<---send--- 200 OK
---recv---> ACK
<--transmition-->
---recv---> BYE //Basestation disconnects from the
radiochannel
<---send--- 200 OK
In order to do this i have written an .xml for simulating the (UAS)
radiostation (Script is in appendance).
For testing i am feeding my server with the standard UAC procedure. In
final state i will connect my UAS to the corporate Radio Basestation
Equipment my company developes.
Problem is that the UAC throws the error (in subject) as soon as my UAS
replies with its first 200.
Hope you guys can help me with this matter.
<?xml version="1.0" encoding="ISO-8859-1"?>
<scenario name="Radio UAS responder with Audio Replay">
<!-- Recieve an INVITE from OWP -->
<recv request="INVITE"
clrf="true">
</recv>
<!-- No Ringing 180 and Testing 100 of need -->
<!--Responsing with 200 OK -->
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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>
<!-- Recieve ACK from OWP -->
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<!-- Transmit Audiostream (UDP) from .pcap File -->
<nop>
<action>
<exec play_pcap_audio="pcap/sample_call.pcap"/>
</action>
</nop>
<!-- Wait 8s since audiostream takes 7s to replay -->
<pause milliseconds="8000"/>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[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="4000"/>
<!-- 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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