VOLTS

VoIP Open Linear Tester Suite

Functional tests for VoIP systems based on voip_patrol, sipp, sox, chromaprint, opensips, and docker.

Overview

The system is designed to run simple call scenarios, that you usually do with your desk phones. Scenarios are run one by one from scenarios folder in alphabetical order, which could be considered a limitation, but also allows you to reuse the same accounts in a different set of tests. This stands for Linear in the name 😉

So, call some destination(s) with one (or more) device(s) and control call arrival on another phone(s). But wait, there is more:

Database Integration: VOLTS can integrate with your MySQL and/or PostgreSQL databases to write some data there before the test and remove it after
Media Recording & Analysis: Record and play media during calls and perform media checks of these files via SoX and Chromaprint
TDD Support: Use it in Test-Driven Development approach when adding functionalities to your existing PBX system. Test-Fail-Fix.

System Architecture

The suite consists of 8 parts, running sequentially:

Preparation - Transform templates to real scenarios using Jinja2 template engine with jinja2_time extension
WebSocket-TLS Proxy - Start proxy to provide WSS transport for voip_patrol scenarios
Database (Pre) - Run database scripts to put test data
VoIP Testing - Run voip_patrol or sipp scenario
Database (Post) - Remove test data from databases
Media Check - Analyze obtained media files if necessary
Proxy Teardown - Stop WebSocket-TLS proxy
Report Generation - Analyze results and print them in desired format

Steps 3-6 run sequentially against scenario files prepared in step 1, one at a time. Again, it’s for Linear.

Getting Started

Building

Suite is designed to run locally from your Linux PC or Mac. Docker should be installed.

Notes on using podman: VOLTS can run using podman-docker package. One obstacle by default - the volumes permissions inside a container. To address this issue please refer to this article.

To build, just run:

./build.sh

Script will build 6 docker images and tag them accordingly.

In case if voip_patrol or sipp is updated, you need to rebuild these containers:

./build.sh -r

Running

After building, run all scenarios:

./run.sh

Simple, isn’t it? This will run all scenarios found in scenarios folder one by one.

To run a single scenario:

./run.sh <scenario_name>
# or
./run.sh scenarios/<scenario_name>

To get a set of tests running using tag keyword:

./run.sh tag=set1,set2

Command Options

Option	Description
`-h, --help`	Show help message
`-l, --log-level N`	Set log level (0=silent, 1=normal, 2=verbose, 3=debug)
`-r, --report TYPE`	Set report type (table\|json\|table_full\|json_full)
`-t, --timeout N`	Set maximum single test time in seconds
`-v, --verbose`	Enable verbose output (equivalent to -l 2)
`-d, --debug`	Enable debug output (equivalent to -l 3)
`--tls-port N`	Set OpenSIPS TLS port
`--wss-port N`	Set OpenSIPS WSS port
`--heps-port N`	Set HEP source port
`--hepd-port N`	Set HEP destination port

Special Commands

Command	Description
`stop`	Stop tests and delete all containers
`sngrep`	Launch SIP packet capture tool
`dbclean`	Clean up test data from databases

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

Environment Variables

You can also configure behavior via environment variables (command-line options take precedence):

Variable	Description
`REPORT_TYPE`	Report type: `table`, `json`, `table_full`, `json_full`. Overridden by `-r/--report`
`LOG_LEVEL`	Log level (0-3). Overridden by `-l/--log-level`
`MAX_SINGLE_TEST_TIME`	Maximum time for a single test in seconds. Overridden by `-t/--timeout`
`OPENSIPS_TLS_PORT`	OpenSIPS TLS port (default: 6051). Overridden by `--tls-port`
`OPENSIPS_WSS_PORT`	OpenSIPS WSS port (default: 9443). Overridden by `--wss-port`
`OPENSIPS_HEPS_PORT`	HEP source port (default: 8887). Overridden by `--heps-port`
`OPENSIPS_HEPD_PORT`	HEP destination port (default: 8888). Overridden by `--hepd-port`

Configuration

Scenarios

VOLTS scenarios are combined voip_patrol/sipp, database, and media_check scenarios, templatized with Jinja2 style. This is done to avoid repeating passwords, usernames, domains, etc.

Due to using jinja2-time extension, it’s possible to use dynamic time/date values in your scenarios, for example testing some time-based rules on your PBX.

Global Config

Values for templates are taken from scenarios/config.yaml. Variables from global section transform to c. (for config) and from accounts to a. in templates for shorter notation.

There is a special name scenario_name that transforms to a scenario file name stripped .xml extension.

All settings from global section are inherited to the accounts section automatically unless defined there explicitly.

There is also an env variable that exposes system environment variables to templates. Use it when you need to pass runtime values without modifying config.yaml:

{{ env.MY_VAR }}                        <!-- empty string if not set -->
{{ env.MY_VAR | default('fallback') }}  <!-- explicit fallback value -->
{{ env.DOMAIN | default(c.domain) }}    <!-- fall back to config.yaml value -->

Example config.yaml:

global:
  domain:     '<SOME_DOMAIN>'
  transport:  'tls'
  srtp:       'dtls,sdes,force'
  play_file:  '/voice_ref_files/8000_12s.wav'
databases:
  'sipproxydb':
    type:       'mysql'
    user:       'sipproxydbrw'
    password:   'sipproxydbrwpass'
    base:       'sipproxydb'
    host:       'mysipproxydb.local'
accounts:
  '88881':
    username:       '88881'
    auth_username:  '88881-1'
    password:       'SuperSecretPass1'
  '88882':
    username:       '88882'
    auth_username:  '88882-67345'
    password:       'SuperSecretPass2'

VoIP Patrol Configuration

Basic Registration Example

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

WebSocket Transport Notice

As voip_patrol itself doesn’t support WebSocket transport, OpenSIPS is used as a TLS-WSS SIP proxy. Using it is simple: specify transport="wss" in your voip_patrol section.

You can debug proxy traffic using sngrep:

./run.sh sngrep
# or
docker exec -it volts_opensips sngrep -L udp:127.0.0.1:8888
# or using a local sngrep installation
sudo sngrep -L udp:127.0.0.1:8888 port 8888

Where 8888 is the OPENSIPS_HEPD_PORT (configurable with --hepd-port option).

Note: you need to use the turn section of your voip_patrol configuration to ensure media passes through. See the WSS example below.

SIPP Configuration

You can add existing SIPP scenarios mainly unchanged. They run with:

sipp <target> -sf <scenario.xml> -m 1 -mp <random_port> -i <container_ip>

Example SIPP OPTIONS test:

<config tag="sipp">
    <section type="sipp">
        <actions>
            <action transport="{{ c.transport }}" target="{{ c.domain }}">
                <scenario name="Options">
                    <send>
                        <![CDATA[
                OPTIONS sip:check_server_health@{{ c.domain }} SIP/2.0
                Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
                Max-Forwards: 70
                To: <sip:check_server_health@{{ c.domain }}>
                From: sipp <sip:check_server_health@[local_ip]:[local_port]>;tag=[call_number]
                Call-ID: [call_id]
                CSeq: 1 OPTIONS
                Contact: <sip:check_server_health@[local_ip]:[local_port]>
                Accept: application/sdp
                Content-Length: 0
                ]]>
                    </send>
                    <recv response="200" timeout="200"/>
                </scenario>
            </action>
        </actions>
    </section>
</config>

SIPP Attributes

Attribute	Description
`transport`	Transport SIPP will use: `udp` (default), `tcp`, or `tls`
`socket_mode`	`single` (default) or `multi` sockets
`target`	Target address (port 5061 appended for TLS if not specified)
`call_rate`	Call rate in calls per second (default: 10)
`max_calls`	Maximum number of calls (default: 1)
`max_concurrent_calls`	Maximum simultaneous calls (default: 10)
`total_timeout`	Test timeout in seconds (default: 600)

Database

Database configuration is done in XML, section database. There are 2 stages:

Stage	Description
`pre`	Launched before running `voip_patrol`. Usually to put accounts data, routing, etc.
`post`	Running after `voip_patrol`. For cleanup data inserted in `pre` stage.

Table Operations

Attribute	Description
`name`	Table name
`type`	Operation: `insert`, `replace`, or `delete`
`continue_on_error`	Ignore errors and continue (optional)
`cleanup_after_test`	Auto-cleanup on post stage for insert operations (optional)

Example with Database:

<config>
    <section type="database">
        <actions>
            <action database="sippproxydb" stage="pre">
                <table name="subscriber" type="insert" cleanup_after_test="true">
                    <field name="username" value="{{ a.88881.username }}"/>
                    <field name="domain" value="{{ c.domain }}"/>
                    <field name="password" value="{{ a.88881.password }}"/>
                </table>
            </action>
        </actions>
    </section>
    <section type="voip_patrol">
        <!-- VoIP Patrol configuration here -->
    </section>
</config>

Media Check

Analyze call recordings with various media tools.

Media Check Attributes

Attribute	Description
`type`	Media check type: `sox`, `sox_st`, or `fpcalc`
`file`	Path to file (must use `/output/` prefix)
`delete_after`	Delete file after check: `yes`/`no`/`keep_failed` (default)
`print_debug`	Print debug info: `yes`/`no` (default)
`sox_filter`	Used if `type` is `sox`/`sox_st`. Semicolon-separated expressions for SoX validation
`fingerprint`	Used if `type` is `fpcalc`. Raw fingerprint from `fpcalc` tool
`length`	Used if `type` is `fpcalc`. Expected length in seconds. Supports `<min>-<max>` format
`likeness`	Used if `type` is `fpcalc`. Minimum similarity score (default: `0.9`)

SoX Media Check

SoX collects parameters using sox --i <file>, sox <file> -n stat, and sox <file> -n stats.

Filter example:

sox_filter="length s -ge 10; length s -le 11; crest factor -lt 10"

This checks that:

Length is between 10-11 seconds
Crest factor is less than 10

Uses bash-style comparison operators (-eq, -lt, -gt, -le, -ge, -ne) instead of <, > symbols.

You can also use sox_st as a type to trim silence at the start and end of the recording before analysis — this allows slightly better file fingerprinting.

Chromaprint Media Check

The fpcalc type calculates the “likeness” (similarity) of a recorded audio file against a provided reference fingerprint using the fpcalc utility. More info about fingerprinting can be found here.

Note: make sure files have at least 3-4 seconds of audio. 10-15 seconds is recommended.

Likeness score reference:

Likeness score	Meaning
0.95 – 1.00	Near identical (codec change, minor distortion)
0.85 – 0.95	Same content, more significant degradation
0.70 – 0.85	Possibly related, uncertain
< 0.70	Likely different content

Example with Media Check:

<config>
    <section type="voip_patrol">
        <actions>
            <!-- Call configuration with record="/output/{{ scenario_name }}.wav" -->
        </actions>
    </section>
    <section type="media_check">
        <actions>
            <action type="sox"
                sox_filter="length s -ge 10; length s -le 11"
                file="/output/{{ scenario_name }}.wav"
            />
        </actions>
    </section>
</config>

Examples

Basic Tests

Simple Registration Test

<config>
    <section type="voip_patrol">
        <actions>
            <action type="register" label="Register {{ a.88881.label }}"
                transport="{{ a.88881.transport }}"
                <!-- Account parameter is more used in receiving calls on this account later -->
                account="{{ a.88881.label }}"
                <!-- username would be a part of AOR - <sip:username@realm> -->
                username="{{ a.88881.username }}"
                <!-- auth_username would be used in WWW-Authorize procedure -->
                auth_username="{{ a.88881.auth_username }}"
                password="{{ a.88881.password }}"
                registrar="{{ c.domain }}"
                realm="{{ a.88881.domain }}"
                <!-- We are expecting to get 200 code here, so REGISTER is successful -->
                expected_cause_code="200"
            />
            <!-- Just wait 2 sec for all timeouts -->
            <action type="wait" complete="true" ms="2000"/>
        </actions>
    </section>
</config>

Basic Call Scenario

<config>
    <section type="voip_patrol">
        <actions>
            <!-- As we're using call functionality here - define the list of codecs -->
            <action type="codec" disable="all"/>
            <action type="codec" enable="pcma" priority="250"/>
            <action type="codec" enable="pcmu" priority="249"/>
            
            <action type="register" label="Register {{ a.88881.label }}"
                transport="{{ a.88881.transport }}"
                account="{{ a.88881.label }}"
                username="{{ a.88881.username }}"
                auth_username="{{ a.88881.auth_username }}"
                password="{{ a.88881.password }}"
                registrar="{{ c.domain }}"
                realm="{{ c.domain }}"
                expected_cause_code="200"
                <!-- Make sure we are using SRTP on a call received. This is done here as accounts are created before accept(answer) action -->
                srtp="{{ a.88881.srtp }}"
            />
            <action type="wait" complete="true" ms="2000"/>
            
            <action type="accept" label="Receive call on {{ a.88881.label }}"
                <!-- This is not a load test - so only 1 call is expected -->
                call_count="1"
                <!-- Make sure we have received a call on a previously registered account -->
                match_account="{{ a.88881.label }}"
                <!-- Hangup in 10 seconds after answer -->
                hangup="10"
                <!-- Send back "200 OK" -->
                code="200" reason="OK"
                transport="{{ a.88881.transport }}"
                <!-- Make sure we are using SRTP -->
                srtp="{{ a.88881.srtp }}"
                <!-- Play a file back to gather RTCP stats in the report -->
                play="{{ c.play_file }}"
            />
            
            <action type="call" label="Call {{ a.90001.label }} -> {{ a.88881.label }}"
                transport="tls"
                <!-- We are waiting for an answer -->
                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"
                <!-- We are specifying all auth data here for INVITE -->
                auth_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>
    </section>
</config>

Advanced Scenarios

Call with Database Integration

<!-- Register with 90012 and receive a call from 90011 -->
<config>
    <section type="database">
        <actions>
        <!-- add subscribers to sip proxy -->
            <!-- "sippproxydb" here is referring to an entity in "databases" from config.yaml. -->
            <action database="sippproxydb" stage="pre">
                <table name="subscriber" type="insert" cleanup_after_test="true">
                    <field name="username" value="{{ a.90011.username }}"/>
                    <field name="domain" value="{{ c.domain }}"/>
                    <field name="ha1" value="{{ a.90011.ha1 }}"/>
                    <!-- here password due to ha1 is useless, so we can put some data based on jinja2_time.TimeExtension -->
                    <field name="password" value="{% now 'local' %}"/>
                </table>
                <table name="subscriber" type="insert" cleanup_after_test="true">
                    <field name="username" value="{{ a.90012.username }}"/>
                    <field name="domain" value="{{ c.domain }}"/>
                    <field name="ha1" value="{{ a.90012.ha1 }}"/>
                    <field name="password" value="{% now 'local' + 'days=1', '%D' %}"/>
                </table>
            </action>
        </actions>
    </section>
    <section type="voip_patrol">
        <!-- VoIP testing actions -->
    </section>
</config>

WSS Call with Media Check

<!-- Call echo service and make sure receive an answer with media -->
<config>
    <section type="voip_patrol">
        <actions>
            <action type="codec" disable="all"/>
            <action type="codec" enable="pcma" priority="250"/>
            <action type="codec" enable="pcmu" priority="249"/>
            <action type="turn" enabled="true" server="{{ c.stun_address }}" stun_only="true"/>
            <action type="wait" ms="5000"/>
            
            <action type="call" label="Call to 11111 (echo) WSS"
                transport="wss"
                <!-- We are expecting answer here -->
                expected_cause_code="200"
                caller="{{ a.88881.label }}@{{ c.domain }}"
                callee="11111@{{ a.88881.domain_wss }}"
                from="sip:{{ a.88881.label }}@{{ c.domain }}"
                to_uri="11111@{{ c.domain }}"
                max_duration="20" hangup="10"
                auth_username="{{ a.88881.username }}"
                password="{{ a.88881.password }}"
                realm="{{ c.domain }}"
                play="{{ c.play_file }}"
                rtp_stats="true"
                srtp="{{ a.88881.dtls }}"
                <!-- We need to record file on answer. To analyze it below now it MUST be with "/output/" path prefix -->
                record="/output/{{ scenario_name }}.wav"
            />
            <action type="wait" complete="true" ms="30000"/>
        </actions>
    </section>
    <section type="media_check">
        <actions>
            <action type="sox"
                <!-- We are testing that the outcome of the recorded file is between 9 and 11 seconds and checking amplitude -->
                sox_filter="length s -ge 9; length s -le 11; maximum amplitude -ge 0.9"
                <!-- File name is the same as in the "record" attribute in the "call" action above. Now it MUST be with "/output/" path prefix -->
                file="/output/{{ scenario_name }}.wav"
                delete_after="yes"
            />
        </actions>
    </section>
</config>

Chromaprint Media Check Example

<!-- Call echo service and verify audio fingerprint similarity -->
<config>
    <section type="voip_patrol">
        <actions>
            <action type="codec" disable="all"/>
            <action type="codec" enable="pcma" priority="250"/>
            <action type="codec" enable="pcmu" priority="249"/>
            <action type="call" label="Call to 11111 (echo)"
                transport="{{ a.88881.transport }}"
                expected_cause_code="200"
                caller="{{ a.88881.label }}@{{ c.domain }}"
                callee="11111@{{ a.88881.domain }}"
                from="sip:{{ a.88881.label }}@{{ c.domain }}"
                to_uri="11111@{{ c.domain }}"
                max_duration="20" hangup="10"
                auth_username="{{ a.88881.username }}"
                password="{{ a.88881.password }}"
                realm="{{ c.domain }}"
                play="{{ c.play_file }}"
                rtp_stats="true"
                srtp="{{ a.88881.srtp }}"
                record="/output/{{ scenario_name }}.wav"
            />
            <action type="wait" complete="true" ms="30000"/>
        </actions>
    </section>
    <section type="media_check">
        <actions>
            <action type="fpcalc"
                length="9.5-10.5"
                fingerprint="3089777192, 3089826344, 3089901096, 3089902136, 3111922232, 3120310648, 2577140987, 2577204442"
                file="/output/{{ scenario_name }}.wav"
                likeness="0.85"
            />
        </actions>
    </section>
</config>

SIPP Tests

Load Testing with SIPP

<config>
    <section type="sipp">
        <actions>
            <action transport="{{ c.transport }}"
                    target="{{ c.domain }}"
                    max_calls="1000"
                    call_rate="10"
                    max_concurrent_calls="10"
                    socket_mode="single">
                <!-- sipp {{ c.domain }}:5061 -sf scen.xml -m 1000 -r 10 -l 10 -t ln -->
                <scenario name="UAC REGISTER - UnREGISTER with Auth">
                    <send retrans="500">
                        <![CDATA[
                            REGISTER sip:{{ c.domain }}:[remote_port] SIP/2.0
                            Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
                            From: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@{{ c.domain }}>;tag=[pid]VOLTsTag00[call_number]
                            To: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@{{ c.domain }}:[remote_port]>
                            Call-ID: {% now 'utc', '%H%M%S' %}///[call_id]
                            CSeq: 1 REGISTER
                            Contact: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@[local_ip]:[local_port];transport=[transport]>
                            Max-Forwards: 70
                            User-Agent: VOLTS/1.0
                            Expires: 60
                            Content-Length: 0

                        ]]>
                    </send>
                    <recv response="401" auth="true"></recv>
                    <send retrans="500">
                        <![CDATA[
                            REGISTER sip:{{ c.domain }}:[remote_port] SIP/2.0
                            Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
                            From: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@{{ c.domain }}>;tag=[pid]VOLTsTag00[call_number]
                            To: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@{{ c.domain }}:[remote_port]>
                            Call-ID: {% now 'utc', '%H%M%S' %}///[call_id]
                            CSeq: 2 REGISTER
                            Contact: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@[local_ip]:[local_port];transport=[transport]>
                            Max-Forwards: 70
                            User-Agent: VOLTS/1.0
                            Expires: 60
                            [authentication username={{ a.88881.username }} password={{ a.88881.password }}]
                            Content-Length: 0

                        ]]>
                    </send>
                    <recv response="200" crlf="true"></recv>
                    
                    <!-- Unregister -->
                    <pause milliseconds="5000"/>
                    <send retrans="500">
                        <![CDATA[
                            REGISTER sip:{{ c.domain }}:[remote_port] SIP/2.0
                            Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
                            From: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@{{ c.domain }}>;tag=[pid]VOLTsTag00[call_number]
                            To: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@{{ c.domain }}:[remote_port]>
                            Call-ID: {% now 'utc', '%H%M%S' %}///[call_id]
                            CSeq: 3 REGISTER
                            Contact: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@[local_ip]:[local_port];transport=[transport]>
                            Max-Forwards: 70
                            User-Agent: VOLTS/1.0
                            Expires: 0
                            Content-Length: 0

                        ]]>
                    </send>
                    <recv response="401" auth="true"></recv>
                    <send retrans="500">
                        <![CDATA[
                            REGISTER sip:{{ c.domain }}:[remote_port] SIP/2.0
                            Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
                            From: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@{{ c.domain }}>;tag=[pid]VOLTsTag00[call_number]
                            To: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@{{ c.domain }}:[remote_port]>
                            Call-ID: {% now 'utc', '%H%M%S' %}///[call_id]
                            CSeq: 4 REGISTER
                            Contact: "VOLTS UAC TESTER" <sip:{{ a.88881.label }}@[local_ip]:[local_port];transport=[transport]>
                            Max-Forwards: 70
                            User-Agent: VOLTS/1.0
                            Expires: 0
                            [authentication username={{ a.88881.username }} password={{ a.88881.password }}]
                            Content-Length: 0

                        ]]>
                    </send>
                    <recv response="200"></recv>
                </scenario>
            </action>
        </actions>
    </section>
</config>

Tagged Test Execution

You can specify a tag on each test scenario:

<config tag='set1'>
    <section type="voip_patrol">
        <!-- Test configuration -->
    </section>
</config>

Multiple tags per scenario are supported:

<config tag='set1,set3'>
    <section type="voip_patrol">
        <!-- Test configuration -->
    </section>
</config>

Run tagged tests:

./run.sh tag=set1
./run.sh tag=set1,set2,test_set3

Results

After running tests, you’ll get a table like this:

+---------------------------------------+-----------------------------------------------------------+------+----------+-------+--------+------------------+
|                              Scenario |                                               VoIP Patrol | SIPP | Database | Media | Status |             Text |
+---------------------------------------+-----------------------------------------------------------+------+----------+-------+--------+------------------+
|                           01-register |                                                      PASS |  N/A |      N/A |   N/A |   PASS |  Scenario passed |
|                                       |                                      Register 88881       |      |          |       |   PASS | Main test passed |
|                          02-call-echo |                                                      PASS |  N/A |      N/A |   N/A |   PASS |  Scenario passed |
|                                       |                                      Call to 11111 (echo) |      |          |       |   PASS | Main test passed |
|            51-call-echo-media-control |                                                      PASS |  N/A |      N/A |  PASS |   PASS |  Scenario passed |
|                                       |                                      Call to 11111 (echo) |      |          |       |   PASS | Main test passed |
| 52-delayed-call-forward-unconditional |                                                      PASS |  N/A |     PASS |   N/A |   PASS |  Scenario passed |
|                                       |                                      Register 90012       |      |          |       |   PASS | Main test passed |
|                                       |                                      Register 90013       |      |          |       |   PASS | Main test passed |
|                                       |                      Receive call on 90012 and not answer |      |          |       |   PASS |    Call canceled |
|                                       |   Call from 90011 to 90012 (delay forward 25 sec) ->90013 |      |          |       |   PASS | Main test passed |
|                                       |                       Receive call on 90013       finally |      |          |       |   PASS | Main test passed |
|                53-server-check-health |                                                       N/A | PASS |      N/A |   N/A |   PASS | SIPP test passed |
+---------------------------------------+-----------------------------------------------------------+------+----------+-------+--------+------------------+

If any scenarios fail, you’ll see:

Scenarios ['49-teams-follow-forward', '50-team-no-answer-forward'] are failed!

This means your system needs tuning or there are issues with the tests. Check the console output and logs in tmp/output folder for detailed information.

For more detailed examples and advanced configurations, refer to the scenarios folder in the repository.