Thanks for your test artifacts David. After encountering the same problem with 
your files too, I realized the problem was at the UAS end.

If I use the following "sipp -sn uas -nr -rtp_echo" at the UAS end the SDP 
answer to my offer comes back with the c-line as 127.0.0.1. I need to use 
"./sipp -sn uas -nr -rtp_echo -i X.Y.Z.W" at the UAS for appropriate behavior.

Thanks again and Regards,
Amit

Subject: RE: [Sipp-users] Having trouble with play_pcap_audio
Date: Thu, 29 Apr 2010 16:45:07 -0700
From: david.wh...@watchguard.com
To: sipp-users@lists.sourceforge.net
CC: voip_e...@hotmail.com








RE: [Sipp-users] Having trouble with play_pcap_audio






indeed, working examples are nice.



here's what I use, and it seems to work fine



attached are uac_pcap.xml and rtp.cap.  then run:



sudo sipp -sf uac_pcap.xml-m 1 -l 1 -mp 6999 -i 10.1.1.12 10.1.1.100



where 10.1.1.12 is the IP address on the uac

and 10.1.1.100 is the IP addr of the uas.



-----Original Message-----

From: Voip Engineer [mailto:voip_e...@hotmail.com]

Sent: Thu 4/29/2010 4:31 PM

To: sipp-users@lists.sourceforge.net

Subject: [Sipp-users] Having trouble with play_pcap_audio





Hello sipp forum members,

 I am trying to use the PCAP (media) play feature of sipp. Following is my sipp 
version built from source with the pcap feature compiled in

SIPp v2.0-PCAP, version 20071128, built Apr 14 2010, 15:48:12.



 I am using the available "UAC with media" scenario from 
http://sipp.sourceforge.net/doc/uac_pcap.xml. With the following usage I see 
that the uac runs fine to completion of one call:

"sudo sipp -sf uac.xml -l 1 -m 1 -r 1 -nr -mp 7300 10.10.X.Y"



I also see the following output in the scenario screen:



 246 Total RTP pckts sent               0.000 last period RTP rate (kB/s)



 Problem is that I don't see the rtp packets at the UAS end nor can I see them 
in a packet capture I run on the UAC host using tcpdump with (100%)appropriate 
filtering options.



 Can someone please give me some guidance or tips on how to debug this.



Thanks and Best Regards,

Voip Engg.











                                         

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