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e9d22f802e
It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
111 lines
3.8 KiB
Plaintext
111 lines
3.8 KiB
Plaintext
Measuring the SIP channel driver's Performance
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==============================================
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This file documents the methods I used to measure
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the performance of the SIP channel driver, in
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terms of maximum simultaneous calls and how quickly
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it could handle incoming calls.
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Knowing these limitations can be valuable to those
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implementing PBX's in 'large' environments. Will your
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installation handle expected call volume?
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Quoting these numbers can be totally useless for other
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installations. Minor changes like the amount of RAM
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in a system, the speed of the ethernet, the amount of
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cache in the CPU, the CPU clock speed, whether or not
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you log CDR's, etc. can affect the numbers greatly.
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In my set up, I had a dedicated test machine running Asterisk,
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and another machine which ran sipp, connected together with
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ethernet.
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The version of sipp that I used was sipp-2.0.1; however,
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I have reason to believe that other versions would work
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just as well.
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On the asterisk machine, I included the following in my
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extensions.ael file:
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context test11
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{
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s => {
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Answer();
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while (1) {
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Background(demo-instruct);
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}
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Hangup();
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}
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_X. => {
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Answer();
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while (1) {
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Background(demo-instruct);
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}
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Hangup();
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}
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}
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Basically, incoming SIP calls are answered, and
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the demo-instruct sound file is played endlessly
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to the caller. This test depends on the calling
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party to hang up, thus allowing sipp to determine
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the length of a call.
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The sip.conf file has this entry:
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[asterisk02]
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type=friend
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context=test11
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host=192.168.134.240 ;; the address of the host you will be running sipp on
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user=sipp
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directmedia=no
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disallow=all
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allow=ulaw
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Note that it's pretty simplistic; no authentication beyond the host ip,
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and it uses ulaw, which is pretty efficient, low-cpu-intensive codec.
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To measure the impact of incoming call traffic on the Asterisk
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machine, I run vmstat. It gives me an idea of the cpu usage by
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Asterisk. The most common failure mode of Asterisk at high call volumes,
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is that the CPU reaches 100% utilization, and then cannot keep up with
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the workload, resulting in timeouts and other failures, which swiftly
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compound and cascade, until gross failure ensues. Watch the CPU Idle %
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numbers.
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I learned to split the testing into two modes: one for just call call processing
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power, in the which we had relatively few simultaneous calls in place,
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and another where we allow the the number of simultaneous calls to quickly
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reach a set maximum, and then rerun sipp, looking for the maximum.
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Call processing power is measured with extremely short duration calls:
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./sipp -sn uac 192.168.134.252 -s 12 -d 100 -l 256
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The above tells sipp to call your asterisk test machine (192.168.134.252)
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at extension 12, each call lasts just .1 second, with a limit of 256 simultaneous
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calls. The simultaneous calls will be the rate/sec of incoming calls times the call length,
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so 1 simultaneous call at 10 calls/sec, and 45 at 450 calls/sec. Setting the limit
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to 256 implies you do not intend to test above 2560 calls/sec.
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Sipp starts at 10 calls/sec, and you can slowly increase the speed by hitting '*' or '+'.
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Watch your cpu utilization on the asterisk server. When you approach 100%, you have found
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your limit.
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Simultaneous calls can be measured with very long duration calls:
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./sipp -sn uac 192.168.134.252 -s 12 -d 100000 -l 270
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This will place 100 sec duration calls to Asterisk. The number of simultaneous
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calls will increase until the maximum of 270 is reached. If Asterisk survives
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this number and is not at 100% cpu utilization, you can stop sipp and run it again
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with a higher -l argument.
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By changing one Asterisk parameter at a time, you can get a feel for how much that change
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will affect performance.
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