On Wed, Feb 23, 2011 at 11:48 PM, Corcione, John <[email protected]>wrote:
> > 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? I am actually talking about this line: m=audio [media_port] RTP/AVP 18 0 2 96 In the above ,you are offering payloads 18, 0 2 and 96. So since you are using 18 in the file, just send it as m=audio [media_port] RTP/AVP 18 96 Does there need to be a pause before and after the NOP command Line? > No idea. > > 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?). Yes. You failed to check these: - are the packets reaching the server? - which payload is being selected by the server? > (I am calling my desk set). > You could try to use an actual sip softphone from the same machine you are running sipp to rule out network issues. > > 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" <[email protected]> > Subject: [Sipp-users] Discarded message which can't be mapped & Can't > Hear Audio to Party Cont'd > To: "[email protected]" > <[email protected]> > 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 > > > > ------------------------------------------------------------------------------ > Free Software Download: Index, Search & Analyze Logs and other IT data in > Real-Time with Splunk. Collect, index and harness all the fast moving IT > data > generated by your applications, servers and devices whether physical, > virtual > or in the cloud. Deliver compliance at lower cost and gain new business > insights. http://p.sf.net/sfu/splunk-dev2dev > _______________________________________________ > Sipp-users mailing list > [email protected] > https://lists.sourceforge.net/lists/listinfo/sipp-users >
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