[svn-commits] murf: branch murf/bug11210 r103686 - /team/murf/bug11210/doc/

Thu Feb 14 14:25:12 CST 2008

Author: murf
Date: Thu Feb 14 14:25:11 2008
New Revision: 103686

URL: http://svn.digium.com/view/asterisk?view=rev&rev=103686
Log:
Adding some documentation for testing sip channel performance using sipp.

Added:
    team/murf/bug11210/doc/chan_sip-perf-testing.txt   (with props)

Added: team/murf/bug11210/doc/chan_sip-perf-testing.txt
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/doc/chan_sip-perf-testing.txt?view=auto&rev=103686
==============================================================================

--- team/murf/bug11210/doc/chan_sip-perf-testing.txt (added)
+++ team/murf/bug11210/doc/chan_sip-perf-testing.txt Thu Feb 14 14:25:11 2008
@@ -1,0 +1,110 @@
+Measuring the SIP channel driver's Performance
+==============================================
+
+This file documents the methods I used to measure
+the performance of the SIP channel driver, in 
+terms of maximum simultaneous calls and how quickly
+it could handle incoming calls.
+
+Knowing these limitations can be valuable to those
+implementing PBX's in 'large' environments. Will your
+installation handle expected call volume?
+
+Quoting these numbers can be totally useless for other
+installations. Minor changes like the amount of RAM
+in a system, the speed of the ethernet, the amount of
+cache in the CPU, the CPU clock speed, whether or not
+you log CDR's, etc. can affect the numbers greatly.
+
+In my set up, I had a dedicated test machine running Asterisk,
+and another machine which ran sipp, connected together with
+ethernet.
+
+The version of sipp that I used was sipp-2.0.1; however, 
+I have reason to believe that other versions would work 
+just as well.
+
+On the asterisk machine, I included the following in my
+extensions.ael file:
+
+context test11
+{
+        s => {
+                Answer();
+                while (1) {
+                        Background(demo-instruct);
+                }
+                Hangup();
+        }
+        _X. => {
+                Answer();
+                while (1) {
+                        Background(demo-instruct);
+                }
+                Hangup();
+        }
+}
+
+Basically, incoming SIP calls are answered, and
+the demo-instruct sound file is played endlessly
+to the caller. This test depends on the calling
+party to hang up, thus allowing sipp to determine
+the length of a call.
+
+The sip.conf file has this entry:
+
+[asterisk02]
+type=friend
+context=test11
+host=192.168.134.240 ;; the address of the host you will be running sipp on
+user=sipp
+canreinvite=no
+disallow=all
+allow=ulaw
+
+Note that it's pretty simplistic; no authentication beyond the host ip, 
+and it uses ulaw, which is pretty efficient, low-cpu-intensive codec.
+
+
+To measure the impact of incoming call traffic on the Asterisk
+machine, I run vmstat. It gives me an idea of the cpu usage by 
+Asterisk. The most common failure mode of Asterisk at high call volumes,
+is that the CPU reaches 100% utilization, and then cannot keep up with
+the workload, resulting in timeouts and other failures, which swiftly 
+compound and cascade, until gross failure ensues. Watch the CPU Idle % 
+numbers.
+
+I learned to split the testing into two modes: one for just call call processing
+power, in the which we had relatively few simultaneous calls in place,
+and another where we allow the the number of simultaneous calls to quickly 
+reach a set maximum, and then rerun sipp, looking for the maximum.
+
+Call processing power is measured with extremely short duration calls:
+
+    ./sipp -sn uac 192.168.134.252 -s 12 -d 100 -l 256
+
+The above tells sipp to call your asterisk test machine (192.168.134.252)
+at extension 12, each call lasts just .1 second, with a limit of 256 simultaneous 
+calls. The simultaneous calls will be the rate/sec of incoming calls times the call length,
+so 1 simultaneous call at 10 calls/sec, and 45 at 450 calls/sec. Setting the limit
+to 256 implies you do not intend to test above 2560 calls/sec.
+
+Sipp starts at 10 calls/sec, and you can slowly increase the speed by hitting '*' or '+'.
+Watch your cpu utilization on the asterisk server. When you approach 100%, you have found 
+your limit.
+
+
+Simultaneous calls can be measured with very long duration calls:
+
+./sipp -sn uac 192.168.134.252 -s 12 -d 100000 -l 270
+
+This will place 100 sec duration calls to Asterisk. The number of simultaneous
+calls will increase until the maximum of 270 is reached. If Asterisk survives
+this number and is not at 100% cpu utilization, you can stop sipp and run it again
+with a higher -l argument.
+
+
+By changing one Asterisk parameter at a time, you can get a feel for how much that change
+will affect performance. 
+
+

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