Sippy Cup

Load testing voice systems, and voice applications in particular, is tricky. While several commercial tools exist, there is really only one tool in the Open Source world that is good at efficiently generating SIP load: SIPp. While SIPp does a good job of generating load, it is somewhat clumsy to use, due to a verbose XML format for scenarios, a confusing set of command line parameters, and worst of all, a lack of tools to create media needed to interact with voice applications.

The last problem is especially tricky: Imagine you want to load test an IVR. Testing requires:

• calling a test number
• waiting a certain amount of time
• sending some DTMF
• waiting some more
• sending more DTMF
• etc....

To test this with SIPp you need a PCAP file that contains the properly timed DTMF interactions. Since there is no tool to create this media, it is usually necessary to call into the system and record the PCAP, isolate the RTP from the captured packets with something like Wireshark, then connect the pcap file into the SIPp scenario. This process is time consuming and error prone, meaning that testing isn't done as often as it should.

SippyCup aims to help solve these problems.

Sippy Cup is a tool to generate SIPp load test profiles and the corresponding media in PCAP format. The goal is to take an input document that describes a load test in a very simple way (call this number, wait this many seconds, send this digit, wait a few more seconds, etc). The ideas are taken from LoadBot, but the goal is for a more performant load generating tool with no dependency on Asterisk.

SippyCup relies on the following to generate scenarios and the associated media PCAP files:

• Ruby 2.3.0 or later
• SIPp latest master branch - Download from https://github.com/sipp/sipp - NOTE: Version SIPp version 3.4 may work, but will be missing certain new Sippy Cup features, such as rate scaling
• "root" user access via sudo: needed to run SIPp so it can bind to raw network sockets

If you do not have Ruby 2.3.3 available (check using ruby --version), we recommend installing Ruby with RVM

Once Ruby is installed, install SippyCup:

gem install sippy_cup

Now you can start creating scenario files like in the examples below.

You use bundle command (from the "bundler" package) to install from the source directly. First, clone the repository into a working directory.

Install bundle via gem:

gem install bundler --no-ri --no-rdoc

Then build the sippy_cup application with bundle.

bundle install

Using bundle will then install the gem dependencies and allow you to run sippy_cup from your working directory.

---
source: 192.0.2.15
destination: 192.0.2.200
max_concurrent: 10
calls_per_second: 5
number_of_calls: 20
steps:
  - invite
  - wait_for_answer
  - ack_answer
  - sleep 3
  - send_digits '3125551234'
  - sleep 5
  - send_digits '#'
  - wait_for_hangup

Both source and destination above may be optionally supplied with a port number, eg. 192.0.2.200:5061

Next, execute the scenario:

$ sippy_cup -r my_test_scenario.yml
I, [2013-09-30T14:48:08.388106 #9883]  INFO -- : Preparing to run SIPp command: sudo sipp -i 192.0.2.15 -p 8836 -sf /var/folders/n4/dpzsp6_95tb3c4sp12xj5wdr0000gn/T/scenario20130930-9883-1crejcw -l 10 -m 20 -r 5 -s 1 192.0.2.200
Password:

...snip...

I, [2013-09-30T14:48:16.728712 #9883]  INFO -- : Test completed successfully.

More examples are available in the source repository.

require 'sippy_cup'

scenario = SippyCup::Scenario.new 'Sippy Cup', source: '192.168.5.5:10001', destination: '10.10.0.3:19995' do |s|
  s.invite
  s.wait_for_answer
  s.ack_answer

  s.sleep 3
  s.send_digits '3125551234'
  s.sleep 5
  s.send_digits '#'

  s.wait_for_hangup
end

# Create the scenario XML and PCAP media. File will be named after the scenario name, in our case:
# * sippy_cup.xml
# * sippy_cup.pcap
scenario.compile!

The above code can be executed as a standalone Ruby script and the resulting scenario file run with SIPp.

Each command below can take SIPp attributes as optional arguments. For a full list of available steps with arguments explained, see the API documentation.

• sleep <seconds> Wait a specified number of seconds
• invite Send a SIP INVITE to the specified target
• receive_invite Wait for an INVITE to be received
• register <username> [password] Register the specified user to the target with an optional password
• send_trying Send a 100 Trying provisional response
• receive_trying Expect to receive a 100 Trying response from the target
• send_ringing Send a 180 Ringing provisional response
• receive_ringing Expect to receive a 180 Ringing response from the target
• receive_progress Expect to receive a 183 Progress response from the target
• send_answer Send a 200 Ok response to an INVITE (answer the call)
• receive_answer Expect to receive a 200 OK (answering the call) response from the target
• answer Convenient shortcut for send_answer; receive_ack
• wait_for_answer Convenient shortcut for receive_trying; receive_ringing; receive_progress; receive_answer, with all but the answer marked as optional
• ack_answer Send an ACK in response to a 200 OK
• receive_ack Expect to receive an ACK
• send_digits <string> Send a DTMF string. May send one or many digits, including 0-9, *, #, and A-D
• receive_ok Expect to receive a 200 OK
• receive_message [regex] Expect to receive a SIP MESSAGE, optionally matching a regex
• send_bye Send a BYE (hangup request)
• receive_bye Expect to receive a BYE from the target
• ack_bye Send a 200 OK response to a BYE
• wait_for_hangup Convenient shortcut for receive_bye; ack_bye
• hangup Convenient shortcut for send_bye; receive_ok
• call_length_repartition Creates a histogram table of individual call lengths in milliseconds between min length and max length, at the specified interval
• response_time_repartition Creates a histogram table of individual SIP request response times in milliseconds between min length and max length, at the specified interval

Don't want your scenario to end up in the same directory as your script? Need the filename to be different than the scenario name? No problem!

For the sippy_cup manifest, use filename:

---
filename: /path/to/somewhere

Or, in Ruby:

s = SippyCup::Scenario.new 'SippyCup', source: '192.168.5.5:10001', destination: '10.10.0.3:19995', filename: '/path/to/somewhere' do
  # scenario definitions here...
end
s.compile!

This will create the files somewhere.xml and somewhere.pcap in the /path/to/ directory.

Each parameter has an impact on the test, and may either be changed once the XML file is generated or specified in the options hash for SippyCup::Scenario.new. In addition to the default parameters, some additional parameters can be set:

stats_file

Path to a file where call statistics will be stored in a CSV format, defaults to not storing stats

stats_interval

Frequency (in seconds) of statistics collections. Defaults to 10. Has no effect unless :stats_file is also specified

from_user

SIP user from which traffic should appear. Default: sipp

to

SIP user / address to send requests to. Defaults to SIPp's default: `s@[destination]` (as in `s@127.0.0.1`). Can specify either a user (`foouser`) or a full address (`foouser@there.com`), the latter being useful for testing multi-tenant systems where the `To` domain is not the same as the hostname of the system.

transport

Specify the SIP transport. Valid options are `udp` (default) or `tcp`. Default: `udp`

full_sipp_output

By default, SippyCup will show SIPp's command line output while running a scenario. Set this parameter to `false` to hide full command line output. Default: `true`

summary_report_file

Write a summary of the SIPp run to the specified file. This summary is the output from the SIPp `-trace_screen` command. Default: unused

errors_report_file

Record SIPp's errors to the specified file. This report is the output from the SIPp `-trace_err` command. Default: unused

options

A string of SIPp command line options included with the SIPp run. Default: none

media_port

By default, SIPp assigns RTP ports dynamically. However, if there is a need for a static RTP port (say, for data collection purposes), it can be done by supplying a port number here. Default: SIPp's default of 6000

dtmf_mode

Specify the mechanism by which DTMF is signaled. Valid options are `rfc2833` for within the RTP media, or `info` for SIP INFO. Default: rfc2833

scenario_variables

If you're using sippy_cup to run a SIPp XML file, there may be CSV fields in the scenario ([field0], [field1], etc.). Specify a path to a CSV file containing the required information using this option. (File is semicolon delimeted, information can be found [here](http://sipp.sourceforge.net/doc/reference.html#inffile).) Default: unused

number_of_calls

The total number of calls permitted for the entire test. When this limit is reached, the test is over. Defaults to none - test will run forever until manually stopped

number_of_calls

The total number of calls permitted for the entire test. When this limit is reached, the test is over. Defaults to nil.

concurrent_max

The maximum number of calls permitted to be active at any given time. When this limit is reached, SIPp will slow down or stop sending new calls until there it falls below the limit. Defaults to SIPp's default: (3 * call_duration (seconds) * calls_per_second)

calls_per_second

The rate at which new calls should be created. Note that SIPp will automatically adjust this downward to stay at or beneath the maximum number of concurrent calls (`concurrent_max`). Defaults to SIP's default of 10

calls_per_second_incr

When used with `calls_per_second_max`, tells SIPp the amount by which `calls_per_second` should be incremented. CPS rate is adjusted each `calls_per_second_interval`. Default: 1.

calls_per_second_interval

When used with `calls_per_second_max`, tells SIPp the time interval (in seconds) by which calls-per-second should be incremented. Default: Unset; SIPp's default (60s). NOTE: Requires a development build of SIPp; see SIPp/sipp#107

calls_per_second_max

The maximum rate of calls-per-second. Default: unused (`calls_per_second` will not change)

advertise_address

The IP address to advertise in SIP and SDP if different from the bind IP. Default: `source` IP address

With Sippy Cup, you can add additional attributes to each step of the scenario:

# This limits the amount of time the server has to reply to an invite (3 seconds)
s.receive_answer timeout: 3000

# You can override the default 'optional' parameters
s.receive_ringing optional: false
s.receive_answer optional: true

# Let's combine multiple attributes...
s.receive_answer timeout: 3000, crlf: true

For more information on possible attributes, visit the SIPp Documentation.

Copyright (C) 2013-2015 Mojo Lingo LLC

Sippy Cup is released under the MIT license. Please see the LICENSE file for details.

Sippy Cup was created by Ben Klang and Will Drexler with support from Mojo Lingo and their clients.

"Sippy Cup" name suggested by Jamey Owens