Ever since the entry of Docker, everyone is busy porting their applications to Docker Containers. Now with the tools like Mesos, CoreOS etc we can easily achieve scalability also. @Plivo we always dedicate ourselves to play around such new technologies. In my previous blog posts, i’ve explained how to containerize the Freeswitch, how to perform some basic load test using simple dialplans etc. My previous load tests required a bunch of basic Freeswitch servers to originate calls to flood the calls to the FreeSwitch container. So this time i’m going to use a simple method, which everyone can use even from their laptops.
Enter SIPp. SIPp is a free Open Source test tool / traffic generator for the SIP protocol. But the main issue for beginer like me is in generating a proper XML for SIPp that can match to my exact production scenarios. After googling, i came across a super simple ruby wrapper over SIPp called sippy_cup. SIPpy_cup is a simple ruby wrapper over SIPp. We just need to create a simple yaml file and sippy_cup parses this yml file and generates the XML equivalent which will be then used to generate calls. sippy_cup can also be used to generate only the XML file for SIPp.
Setting up sippy_cup is very simple. There are only two dependencies
1) ruby (2.1.2 recomended)
2) SIPp
Another important dependency is our local internet bandwidth. Flooding too many calls will definitely result in network bottlenecks, which i faced when i generated 1k calls from my laptop. Now let’s install SIPp.
sudo apt-get install pcaputils libpcap-dev libncurses5-dev
wget 'http://sourceforge.net/projects/sipp/files/sipp/3.2/sipp.svn.tar.gz/download'
tar zxvf sipp.svn.tar.gz
# compile sipp
make
# compile sipp with pcapplay support
make pcapplay
Once we have installed SIPp and ruby, we can install sippy_cup via ruby gems.
gem install sippy_cup
Configuring sippy_cup
First we need to create yml file for our call flow. There is a good documentation available on the Readme on various options that can be used to create the yml to suit to our call flow. My call flow is pretty simple, i’ve a DialPlan in my Docker FS, which will play an mp3 file. So below is a simple yml config for this call flow
source: <local_machine_ip>
destination: <docker_fs_ip>:<fs_port>
max_concurrent: <no_of_concurrent_calls>
calls_per_second: <calls_per_second>
number_of_calls: <total_no_of_calls>
to_user: <to_number> # => should match the FS Dialplan
steps: # call flow steps
- invite # Initial Call INVITE
- wait_for_answer # Waiting for Answer, handles 100, 180/183 and finally 200 OK
- ack_answer # ACK for the 200 OK
- sleep 1000 # Sleeps for 1000 seconds
- send_bye # Sends BYE signal to FS
Now let’s run sippy_cup using our config yml
sippy_cup -r test.yml
Below is the output of a sample load test. Total 20 calls with 10 concurrent calls
INVITE ----------> 20 1 0
100 <---------- 20 0 0 0
180 <---------- 0 0 0 0
183 <---------- 0 0 0 0
200 <---------- E-RTD1 20 0 0 0
ACK ----------> 20 0
[ NOP ]
Pause [ 30.0s] 20 0
BYE ----------> 20 0
------------------------------ Test Terminated --------------------------------
----------------------------- Statistics Screen ------- [1-9]: Change Screen --
Start Time | 2014-10-22 19:12:40.494470 1414030360.494470
Last Reset Time | 2014-10-22 19:13:45.355358 1414030425.355358
Current Time | 2014-10-22 19:13:45.355609 1414030425.355609
-------------------------+---------------------------+--------------------------
Counter Name | Periodic value | Cumulative value
-------------------------+---------------------------+--------------------------
Elapsed Time | 00:00:00:000000 | 00:01:04:861000
Call Rate | 0.000 cps | 0.308 cps
-------------------------+---------------------------+--------------------------
Incoming call created | 0 | 0
OutGoing call created | 0 | 20
Total Call created | | 20
Current Call | 0 |
-------------------------+---------------------------+--------------------------
Successful call | 0 | 20
Failed call | 0 | 0
-------------------------+---------------------------+--------------------------
Response Time 1 | 00:00:00:000000 | 00:00:01:252000
Call Length | 00:00:00:000000 | 00:00:31:255000
------------------------------ Test Terminated --------------------------------
I, [2014-10-22T19:13:45.357508 #17234] INFO -- : Test completed successfully!
I tried to perform a large scale load test by making 1k calls with 250 concurrent calls. My local internet was flooding with network traffic as there was real Media packets coming from the servers, though it bottlenecked my internet, but still i was able to make 994 successfull calls. I suggest to do such heavy load test on machines wich has good network throughput. Below are the output for this test.
------------------------------ Scenario Screen -------- [1-9]: Change Screen --
Call-rate(length) Port Total-time Total-calls Remote-host
5.0(0 ms)/1.000s 8836 585.61 s 1000 54.235.170.44:5060(UDP)
Call limit reached (-m 1000), 0.507 s period 1 ms scheduler resolution
6 calls (limit 250) Peak was 176 calls, after 150 s
0 Running, 8 Paused, 1 Woken up
604 dead call msg (discarded) 0 out-of-call msg (discarded)
3 open sockets
1490603 Total RTP pckts sent 0.000 last period RTP rate (kB/s)
Messages Retrans Timeout Unexpected-Msg
INVITE ----------> 1000 332 0
100 <---------- 954 53 0 0
180 <---------- 0 0 0 0
183 <---------- 0 0 0 0
200 <------2014-10-22 19:19:23.202714 1414030763.202714: Dead call 990-17510@192.168.1.146 (successful),
received 'SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.146:8836;received=208.66.27.62;branch=z9hG4bK-17510-990-8
From: "sipp" <sip:sipp@192.168.1.146>;tag=990
To: <sip:14158872327@54.235.170.44:5060>;tag=9p6t351mvXZXg
Call-ID: 990-17510@192.168.1.146
CSeq: 2 BYE
User-Agent: Plivo
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Conte---- E-RTD1 994 126 0 0
ACK ----------> 994 126
[ NOP ]
Pause [ 30.0s] 994 0
BYE ----------> 994 0
------------------------------ Test Terminated --------------------------------
----------------------------- Statistics Screen ------- [1-9]: Change Screen --
Start Time | 2014-10-22 19:15:29.941276 1414030529.941276
Last Reset Time | 2014-10-22 19:25:15.056475 1414031115.056475
Current Time | 2014-10-22 19:25:15.564038 1414031115.564038
-------------------------+---------------------------+--------------------------
Counter Name | Periodic value | Cumulative value
-------------------------+---------------------------+--------------------------
Elapsed Time | 00:00:00:507000 | 00:09:45:622000
Call Rate | 0.000 cps | 1.708 cps
-------------------------+---------------------------+--------------------------
Incoming call created | 0 | 0
OutGoing call created | 0 | 1000
Total Call created | | 1000
Current Call | 6 |
-------------------------+---------------------------+--------------------------
Successful call | 0 | 994
Failed call | 0 | 0
-------------------------+---------------------------+--------------------------
Response Time 1 | 00:00:00:000000 | 00:00:01:670000
Call Length | 00:00:00:000000 | 00:00:31:673000
------------------------------ Test Terminated --------------------------------
Sippy_cup is definitely a good tool for all beginers who finds really hard time to work around with SIPp XML’s. I’m really excited to see how Docker is going to contirbute to VOIP world.
beingasysadmin 