We are using SIPp to send load to an Avaya IVR platform which will have the
Qfinity/Autonomy Platform Added to it. It will have The AES and OBSERVE
servers to where the call comes in and activates the CTI Screen POP and Records
the call between the Agent and the Caller. We need to send voice energy to the
platform. According to SIPp this is possible by sending the PCAP rtp/Audio
stream file once the session has been established. Below is the partial
Script. We are sending G729 to our platform. We are asking for Professional
Services and/or a Subject Matter Expert experienced in Sending RTP streams or
wave files via SIPp to an IVR platform.
MayamaTakeshi or anyone:
You say you use do not use RTP for Load Testing but Use SIPp, This is what we
are using and trying to develop the script with the play_pcap_audio file within
our scripting. You had mentioned "eliminate the payloads in the list that
don't correspond to the payload present in the pcap file". That being said,
are you suggesting removing the "a=rtpmap:2 G726-32/8000" line if we are
sending G729 to our platform? Does there need to be a pause before and after
the NOP command Line?
We are able to get the call established we see the RTP packets being sent but
we do not hear the audio to the called party (should we?). (I am calling my
desk set).
Anyone with Experience and or Professional Services who could answer this
question is greatly appreciated.
Thanks
John Corcione
H&R Block Sr. Software/Load & Performance Test Engineer
Message: 1
Date: Tue, 22 Feb 2011 10:06:51 -0600
From: "Corcione, John" <john.corci...@hrblock.com>
Subject: [Sipp-users] Discarded message which can't be mapped & Can't
Hear Audio to Party Cont'd
To: "sipp-users@lists.sourceforge.net"
<sipp-users@lists.sourceforge.net>
Message-ID:
<dd2220c1bc39c04795acc71a9d93467b0530880...@bcmsg01.hrbinc.hrblock.net>
Content-Type: text/plain; charset="us-ascii"
This is the script which sends the PCAP file to our IVR platform - is my media
type correct? Is the codec correct? Should I hear any audio to the called
party? I am trying to determine why I would not hear audio to the called party
playing the PCAP file. Please let me know if I am asking the wrong questions.
I have sent several emails to the list and no one is answering.
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=SIP Media Capabilities
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 18 0 2 96
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap96 telephone-event/8000
a=fmtp:96 0-15
a=fmtp:18 annexb=no
a=fmtp:96 0-15
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>
<nop>
<action>
<exec play_pcap_audio="pcap\MMdemoMainmenuAMR.pcap"/>
</action>
</nop>
<pause milliseconds="360000" />
<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
------------------------------------------------------------------------------
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