Asterisk Documentation 1.6.1 chan_sip-perf-testing.txt

-=NOTE: These pages are automatically updated once per
day from the Asterisk subversion repository when the repository changes revisions. Any
changes made to this page will be automatically overwritten with the
latest version from http://svn.digium.com/view/asterisk/branches/.

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. 




-=NOTE: These pages are automatically updated once per
day from the Asterisk subversion repository when the repository changes revisions. Any
changes made to this page will be automatically overwritten with the
latest version from http://svn.digium.com/view/asterisk/branches/.

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. 




Created by: josiahbryan, Last modification: Tue 25 of May, 2010 (08:41 UTC)
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