forked from Mirrors/freeswitch
da925bddf8
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@13790 d0543943-73ff-0310-b7d9-9358b9ac24b2
128 lines
3.4 KiB
XML
128 lines
3.4 KiB
XML
<?xml version="1.0" encoding="ISO-8859-1" ?>
|
|
<!DOCTYPE scenario SYSTEM "sipp.dtd">
|
|
|
|
|
|
<scenario name="MRCP Recognizer Resource UAS">
|
|
<!-- 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 request="INVITE" crlf="true">
|
|
</recv>
|
|
|
|
<!-- The '[last_*]' keyword is replaced automatically by the -->
|
|
<!-- specified header if it was present in the last message received -->
|
|
<!-- (except if it was a retransmission). If the header was not -->
|
|
<!-- present or if no message has been received, the '[last_*]' -->
|
|
<!-- keyword is discarded, and all bytes until the end of the line -->
|
|
<!-- are also discarded. -->
|
|
<!-- -->
|
|
<!-- If the specified header was present several times in the -->
|
|
<!-- message, all occurences are concatenated (CRLF seperated) -->
|
|
<!-- to be used in place of the '[last_*]' keyword. -->
|
|
|
|
<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 IP4 [local_ip]
|
|
s=-
|
|
c=IN IP4 [media_ip]
|
|
t=0 0
|
|
m=application 1050 TCP/MRCPv2 1
|
|
a=setup:passive
|
|
a=connection:new
|
|
a=channel:dca48cf082dd584b@speechrecog
|
|
a=cmid:1
|
|
m=audio [media_port] RTP/AVP 0 8
|
|
a=recvonly
|
|
a=mid:1
|
|
|
|
]]>
|
|
</send>
|
|
|
|
<recv request="ACK"
|
|
optional="true"
|
|
rtd="true"
|
|
crlf="true">
|
|
</recv>
|
|
|
|
<recv request="INVITE" crlf="true">
|
|
</recv>
|
|
|
|
<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 IP4 [local_ip]
|
|
s=-
|
|
c=IN IP4 [media_ip]
|
|
t=0 0
|
|
m=application 0 TCP/MRCPv2 1
|
|
a=setup:passive
|
|
a=connection:existing
|
|
a=channel:dca48cf082dd584b@speechrecog
|
|
a=cmid:1
|
|
m=audio 0 RTP/AVP 0 8
|
|
a=recvonly
|
|
a=mid:1
|
|
|
|
]]>
|
|
</send>
|
|
|
|
<recv request="ACK"
|
|
optional="true"
|
|
rtd="true"
|
|
crlf="true">
|
|
</recv>
|
|
|
|
<recv request="BYE">
|
|
</recv>
|
|
|
|
<send>
|
|
<![CDATA[
|
|
|
|
SIP/2.0 200 OK
|
|
[last_Via:]
|
|
[last_From:]
|
|
[last_To:]
|
|
[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>
|