Voip Open Linear Testing Suite

Related tags

Testingvolts
Overview

VOLTS

Voip Open Linear Tester Suite

Functional tests for VoIP systems based on voip_patrol and docker

10'000 ft. view

System is designed to run simple call scenarios, that you usually do with your desk phones. So, call some destination and control call arrival on another phone(s).
This tool would not configure your PBX to provide call flows, it's up to you. It will just make and receive calls, nothing more. It also will not do transfers at the moment. Sorry

Suite consists of 3 parts, that are running sequentially

  1. Preparation - at this part we're transforming templates to real scenarios using Jinja2 template engine
  2. Running voip_patrol against list of scenarios sequentially. One scenario at a time
  3. Report - at this part we're analyzing results of previous step reading and interpreting file obtained at step 2. Printing results in a desired way. Table by default.

Building

Suite is designed to run locally from your Linux PC or Mac. And of course, docker should be installed. It's up to you.
To build, just run ./build.sh. It would build 3 docker images and tag em accordingly.

Running

After building, just run

./run.sh

Simple, isn't it? This will run all scenarios found in vp_scenarios folder. To run single scenario, run

./run.sh <scenario_name>.xml

or

./run.sh vp_scenarios/<scenario_name>.xml

After running of the suite you can always find a voip_patrol presented results in tmp/output folder.

But simply run something blindly is boring, so before this best to do some

Configuration

We suppose to configure 2 parts here. First, and most complex are

Scenarios

VOLTS scenarios are voip_patrol scenarios, that are just being templatized with Jinja2 style. Mostly done not to repeat some passwords, usernames, domains, etc.
Values for templates are taken from vp_scenarios/config.yaml
One thing to mention here, that vars from global section transforms to c. and from accounts to a. in templates for shorter notation.
Also all settings from global section are inherited to accounts section automatically, unless they are defined there explicitly.
To get most of it, please refer to voip_patrol config, but here just some more basic examples.

config.yaml

global:
  domain:     '
   
    '
   
  transport:  'tls'
  srtp:       'dtls,sdes,force'
  play_file:  '/voice_ref_files/8000_12s.wav'
accounts:
  '88881':
    username: '88881'
    password: 'SuperSecretPass1'
  '88882':
    username: '88882'
    password: 'SuperSecretPass1'
  '90001':
    username: '90001'
    password: 'SuperSecretPass2'

Make a successful register

"> 
<config>
    <actions>
        <action type="register" label="Register {{ a.88881.label }}"
            transport="{{ a.88881.transport }}"
            account="{{ a.88881.label }}"
            username="{{ a.88881.username }}"
            password="{{ a.88881.password }}"
            registrar="{{ c.domain }}"
            realm="{{ a.88881.domain }}"
            expected_cause_code="200"
        />
        <action type="wait" complete="true" ms="2000"/>
    actions>
config>

Expect fail on register

"> 
<config>
    <actions>
        <action type="register" label="Register {{ a.88881.label }}"
            transport="{{ a.88881.transport }}"
            account="{{ a.88881.label }}"
            username="{{ a.88881.username }}"
            password="{{ a.88881.password }}"
            registrar="{{ c.domain }}"
            realm="{{ a.88881.domain }}"
            expected_cause_code="407"
        />
        <action type="wait" complete="true" ms="2000"/>
    actions>
config>

Register with 1 account and make a call

from 90001 to 88881. Max wait time to answer - 15 sec, duration of connected call - 10 sec.
Point, we don't register account 90001 here, as we're not receiving a calls on it, just need to provide credentials on INVITE.

"> 
<config>
    <actions>
        <action type="codec" disable="all"/>
        <action type="codec" enable="pcma" priority="250"/>
        <action type="codec" enable="pcmu" priority="249"/>
        <action type="codec" enable="opus" priority="248"/>
        <action type="register" label="Register {{ a.88881.label }}"
            transport="{{ a.88881.transport }}"
            account="{{ a.88881.label }}"
            username="{{ a.88881.username }}"
            password="{{ a.88881.password }}"
            registrar="{{ c.domain }}"
            realm="{{ c.domain }}"
            expected_cause_code="200"
            srtp="{{ a.88881.srtp }}"
        />
        <action type="wait" complete="true" ms="2000"/>
        <action type="accept" label="Receive call on {{ a.88881.label }}"
            call_count="1"
            account="{{ a.88881.label }}"
            hangup="10"
            code="200" reason="OK"
            transport="{{ a.88881.transport }}"
            srtp="{{ a.88881.srtp }}"
            play="{{ c.play_file }}"
        />
        <action type="call" label="Call {{ a.90001.label }} -> {{ a.88881.label }}"
            transport="tls"
            expected_cause_code="200"
            caller="{{ a.90001.label }}@{{ c.domain }}"
            callee="{{ a.88881.label }}@{{ c.domain }}"
            from="sip:{{ a.90001.label }}@{{ c.domain }}"
            to_uri="{{ a.88881.label }}@{{ c.domain }}"
            max_duration="20" hangup="10"
            username="{{ a.90001.username }}"
            password="{{ a.90001.password }}"
            realm="{{ c.domain }}"
            rtp_stats="true"
            max_ring_duration="15"
            srtp="{{ a.90001.srtp }}"
            play="{{ c.play_file }}"
        />
        <action type="wait" complete="true" ms="30000"/>
    actions>
config>

Register with 2 accounts

and call from third one, not answer on 1st and make sure we receive call on second. So, your PBX should be configured to make a Forward-No-Answer from 88881 to 88882

"> 
<config>
    <actions>
        <action type="codec" disable="all"/>
        <action type="codec" enable="pcma" priority="250"/>
        <action type="codec" enable="pcmu" priority="249"/>
        <action type="codec" enable="opus" priority="248"/>
        <action type="register" label="Register {{ a.88881.label }}"
            transport="{{ a.88881.transport }}"
            account="{{ a.88881.label }}"
            username="{{ a.88881.username }}"
            password="{{ a.88881.password }}"
            registrar="{{ c.domain }}"
            realm="{{ c.domain }}"
            expected_cause_code="200"
            srtp="{{ a.88881.srtp }}"
        />
        <action type="register" label="Register {{ a.88882.label }}"
            transport="{{ a.88882.transport }}"
            account="{{ a.88882.label }}"
            username="{{ a.88882.username }}"
            password="{{ a.88882.password }}"
            registrar="{{ c.domain }}"
            realm="{{ c.domain }}"
            expected_cause_code="200"
            srtp="{{ a.88882.srtp }}"
        />
        <action type="wait" complete="true" ms="2000"/>
        <action type="call" label="Call from 90001 to 88881->88882"
            transport="{{ a.90001.transport }}"
            expected_cause_code="200"
            caller="{{ a.90001.label }}@{{ c.domain }}"
            callee="88881@{{ c.domain }}"
            from="sip:{{ a.90001.label }}@{{ c.domain }}"
            to_uri="88881@{{ c.domain }}"
            max_duration="20" hangup="10"
            username="{{ a.90001.username }}"
            password="{{ a.90001.password }}"
            realm="{{ c.domain }}"
            rtp_stats="true"
            max_ring_duration="60"
            srtp="{{ a.90001.srtp }}"
            play="{{ c.play_file }}"
        />
        <action type="accept" label="Receive call on {{ a.88881.label }}"
            account="{{ a.88881.label }}"
            call_count="1"
            hangup="10"
            ring_duration="30"
            cancel="force"
            transport="{{ a.88881.transport }}"
            srtp="{{ a.88881.srtp }}"
        />
        <action type="accept" label="Receive call on {{ a.88882.label }}"
            account="{{ a.88882.label }}"
            call_count="1"
            hangup="10"
            code="200" reason="OK"
            transport="{{ a.88882.transport }}"
            srtp="{{ a.88882.srtp }}"
            play="{{ c.play_file }}"
        />
        <action type="wait" complete="true" ms="20000"/>
    actions>
config>

run.sh script

Not that much to configure here, mostly you'll be interested in setting environement variables at the start of the script

Variable name Description
REPORT_TYPE Actually, report type, that would be provided at the end.
table - print results in table, only failed tests are pritend.
json - print results in JSON format, only failed tests are pritend.
table_full, json_full - prints results in table or JSON respectively, but print full info on tests passed
VP_LOG_LEVEL voip_patrol log level on the console

Results

As a results, you would have tables like

+-------------------------+----------------+--------+-----------------+
|                Scenario |           Test | Status |            Text |
+-------------------------+----------------+--------+-----------------+
| 09-queue-forward-member |                |   PASS | Scenario passed |
|                         | Register 90002 |   PASS |     Test passed |
|                         |  Call to 91502 |   PASS |     Test passed |
+-------------------------+----------------+--------+-----------------+
Tests passed OK!

Not really much to describe here, just read info on the console

Owner
Igor Olhovskiy
VoIP engineer and developer. Asterisk/Freeswitch/Kamailio/OpenSIPS and all related to this stuff.
Igor Olhovskiy
GitHub Repository
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