I figured out the payload had to match the pcap file. However when I send the
call to my desk phone I hear the tones in the continuous pcap file but sound
very short maybe choppy. I am trying to play the tones separately however I
only here on tone. I am trying to play a series of digits. As before an
account number pause 5 seconds and then a 4 digit pin. The xml file is below.
As stated before i think the payloads in the pcap file and in my invite are
matching and I am able to hear some digit... does anyone have a scenario that
works on an avaya system?
<?xml version="1.0" encoding="us-ascii"?>
<scenario name="Hangup_1">
<send>
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Supported: 100rel, replaces
Max-Forwards: 70
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS,BYE, UPDATE, REFER, NOTIFY
Contact: <sip:sipp@[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=SIP Media Capabilities
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
a=maxptime:20
a=sendrecv
]]>
</send>
<recv response="100" crlf="true" />
<recv response="183" crlf="true" />
<send>
<![CDATA[
PRACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 PRACK
Max-Forwards: 70
RAck: 1 1 INVITE
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true" />
<recv response="200" crlf="true" />
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: <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
Subject: Performance Test
Content-Length: 0
]]>
</send>
<pause milliseconds="15000" />
<nop>
<action>
<exec
play_pcap_audio="pcap\dtmf_2833_5.pcap" />
</action>
</nop>
<pause milliseconds="750" />
<nop>
<action>
<exec
play_pcap_audio="pcap\dtmf_2833_1.pcap" />
</action>
</nop>
<pause milliseconds="750" />
<nop>
<action>
<exec
play_pcap_audio="pcap\dtmf_2833_1.pcap" />
</action>
</nop>
<pause milliseconds="750" />
<nop>
<action>
<exec
play_pcap_audio="pcap\dtmf_2833_1.pcap" />
</action>
</nop>
<pause milliseconds="750" />
<nop>
<action>
<exec
play_pcap_audio="pcap\dtmf_2833_1.pcap" />
</action>
</nop>
<pause milliseconds="40000" />
<send>
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 BYE
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true" />
<label id="1" />
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150,
200" />
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000,
10000" />
</scenario>
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