Kad, thanks for the kudos. For your points: 1. Creating delay, jitter and other IP impairments could be done on Linux using "NISTnet". You might find some other interesting information in this slide set: http://www.fostel.org/2006/slides/fostel2006-jacques-test-tools.pdf 2. I'd like to know too. A possibility could be to integrate a true RTP stack to SIPp (I have started looking at pjmedia) 3. Yes! the pcap play feature is currently starting one thread per call, which limits a lot the number of simultaneous calls. I was able to reach 1200 simultaneous calls this way (calls where lasting 9s) 4. Don't know. There are probably many things missing in SIPp for that. 5. Don't know Olivier.
kade reva wrote: > Actaully I forgot to add major problem. I get only one way audio (seen > through ethereal RTP traces) like client sends a pcap file and server also > sends a pcap file it but when I run the stream analysis on captured trace, I > get only forward direction. Has anybody got 2 way audio?What are the tricks > ? > > > >> From: "kade reva" <[EMAIL PROTECTED]> >> To: [email protected] >> Subject: [Sipp-users] SIPp as a alternative to commercial bulk >> callgenerators >> Date: Tue, 24 Oct 2006 15:36:15 -0400 >> >> 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 >> >> -- Olivier HP OpenCall Software http://www.hp.com/go/opencall/ ------------------------------------------------------------------------- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
