Matthew Boehm
2005-Jun-28 20:51 UTC
[Asterisk-Users] Anyone using SipP to produce RTP load?
Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729 conversion would occur. Nothing. I was able to pump 10 simul calls that went this path: sipp -> asterisk -> pri -> telco ->pri ->asterisk ..and still no 729 usage or any other discernable load on the server. Can anyone offer suggestion on how to really simulate calls (using sipp or other tester) to asterisk to verify its ability to process X calls? I know someone out there has done this, but forget who it was. Thanks, Matthew
That would probably be me. You could use a lot of different things to do the testing, one would be the tcl script in your asterisk/contrib/scripts directory, some more can be found in the beginning of this presentation: http://astertest.com/astricon_performance.ppt We started some callgenerator for asterisk a very long time ago. ( I have to admit its far from ready and contains many bugs). A howto for this tool can be found at : http://www.asteriskguru.com/tutorials/astertest.html If you want to use sipp, be sure to use playback on your asterisk server and not app_milliwatt, meetme or echo. (those applications will not send any rtp if nothing was received). SIPP only does echoing Zoa ----------- Asteriskguru tutorials: http://www.asteriskguru.com/tutorials/ Visit ClueCon - the asterisk developpers conference: http://www.cluecon.com/ - Dates: August 3, 4 and 5, 2005 Best Western Chicago West Matthew Boehm wrote:>Hey gang, > I've been able to use sipp to produce some call volume on our asterisk >server. The server has no problems handling 50 simul calls. But then again, >no RTP is being done. I tried to use the rtp echo ability of sipp but that >doesn't seem to work right. > I also setup a fake number in asterisk that when called by sipp, would dial >another number via PRI, hoping that some 729 conversion would occur. >Nothing. I was able to pump 10 simul calls that went this path: > > sipp -> asterisk -> pri -> telco ->pri ->asterisk > >..and still no 729 usage or any other discernable load on the server. > >Can anyone offer suggestion on how to really simulate calls (using sipp or >other tester) to asterisk to verify its ability to process X calls? > >I know someone out there has done this, but forget who it was. > >Thanks, >Matthew > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 254 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050629/b4b13b22/signature.pgp
On 29 Jun 2005, at 04:51, Matthew Boehm wrote:> Hey gang, > I've been able to use sipp to produce some call volume on our > asterisk > server. The server has no problems handling 50 simul calls. But > then again, > no RTP is being done. I tried to use the rtp echo ability of sipp > but that > doesn't seem to work right. > I also setup a fake number in asterisk that when called by sipp, > would dial > another number via PRI, hoping that some 729 conversion would occur. > Nothing. I was able to pump 10 simul calls that went this path: > > sipp -> asterisk -> pri -> telco ->pri ->asterisk > > ..and still no 729 usage or any other discernable load on the server. > > Can anyone offer suggestion on how to really simulate calls (using > sipp or > other tester) to asterisk to verify its ability to process X calls? > > I know someone out there has done this, but forget who it was. >I think you mean Signate. I saw a presentation at Astrcon . They call the milliwatt generator to fill the RTP stream. They were getting 122 passthrough ulaw calls on a 'stock' pc. If I remember right the benchmark scripts and methodology are available. If you are looking to benchmark that 4 way 500Mhz box of yours I'd be _very_ interested in the results with varying numbers of CPUs. Signate were saying that the limiting factor (with ulaw passthrough) is the PC architecture (bus and interrupt structure) not the CPU. I've done a _tiny_ experiment myself. I found that a single 729->alaw- >PRI call uses less than 10% of the CPU on a 1Ghz nemiah Tim.> > Thanks, > Matthew >
On 29 Jun 2005, at 04:51, Matthew Boehm wrote:> Hey gang, > I've been able to use sipp to produce some call volume on our > asterisk > server. The server has no problems handling 50 simul calls. But > then again, > no RTP is being done. I tried to use the rtp echo ability of sipp > but that > doesn't seem to work right. > I also setup a fake number in asterisk that when called by sipp, > would dial > another number via PRI, hoping that some 729 conversion would occur. > Nothing. I was able to pump 10 simul calls that went this path: > > sipp -> asterisk -> pri -> telco ->pri ->asterisk > > ..and still no 729 usage or any other discernable load on the server. > > Can anyone offer suggestion on how to really simulate calls (using > sipp or > other tester) to asterisk to verify its ability to process X calls? > > I know someone out there has done this, but forget who it was. > > >I think you mean Signate. I saw a presentation at Astrcon . They call the milliwatt generator to fill the RTP stream. They were getting 122 passthrough ulaw calls on a 'stock' pc. If I remember right the benchmark scripts and methodology are available. If you are looking to benchmark that 4 way 500Mhz box of yours I'd be _very_ interested in the results with varying numbers of CPUs. Signate were saying that the limiting factor (with ulaw passthrough) is the PC architecture (bus and interrupt structure) not the CPU. I've done a _tiny_ experiment myself. I found that a single 729->alaw- >PRI call uses less than 10% of the CPU on a 1Ghz nemiah Tim.> > Thanks, > Matthew > > >
On 06/29/05 11:51 Matthew Boehm said the following:> Hey gang, > I've been able to use sipp to produce some call volume on our asterisk > server. The server has no problems handling 50 simul calls. But then again, > no RTP is being done. I tried to use the rtp echo ability of sipp but thati've used the following sipp command line, sipp -d 30000 -r 5 -t un -sn uac -l 50 -m 100 -s 20 -mp 10000 <asterisk ip> which will generate 100 calls of 30 seconds each, limiting it to 50 simultaneous calls at a time to extension 20 on asterisk. extensions 20 was a simple exten => 20,1,Answer() exten => 20,2,Playback(demo-instruct) exten => 20,3,Goto(1) this had asterisk send back the Playback output on RTP port 10000 to sipp. if you wanted to test ulaw<-->g729 conversions between two asterisk servers, have the above exten lines in the second asterisk server with the exten lines in the first just being exten => 20, 1, Dial(IAX2/asterisk2/20) -- Regards, /\_/\ "All dogs go to heaven." dinesh@alphaque.com (0 0) http://www.alphaque.com/ +==========================----oOO--(_)--OOo----==========================+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=========================================================================+