Hi,
I have used latest sipp on LINUX  as a valuable tool for generating bulk 
calls.Thanks for the great tool and the hardwork put by unselfish 
hardworking engineers.Our company makes SIP based voice gateways and 
important way to ensure it's quality is to make bulk calls and assess the 
quality metrics (like MOS,PESQ,PSQM,hit and clips etc). We use the 
commercial tools from Spirent and Empirix to do that. But as the features in 
our gateway grow , it's not economically feasible to buy multiple licenses 
of these expensive equipments so the need for Sipp as an alternative viable 
solution.I have started implemeting sipp to add to our automation process; 
These are some of my road blocks.
1. sending the different media files like G711(A/U), G723.1,G.729 etc and 
DTMF tones and playing it back using RTP echo option. Done.But I am not able 
to introduce real time network scenarios like Jitter,packet loss rate etc in 
general add all the RTCP parameters. Is there a source code for it ?Also I 
am not able to stop the test on any type of unexpected error.
2.Has anyone done any voice quality measurement (not just listening) on the 
received packets?
3.Is there a limit on max calls made/received from client/server when 
playing pcap files?
4.Has anybody tested STUN,TURN and ICE NAT traversal features with SIPp as a 
test tool?
5.I was thinking of evaluating WinSIP(from touch stone) as another 
alternative; but no commitment from our management. Has anyone used that and 
opinions on that?

Thanks very much,
Kad

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