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This commit is contained in:
Dave Horton
2020-01-07 10:34:03 -05:00
commit 523e2a308b
61 changed files with 7876 additions and 0 deletions
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27
test/create-test-db.js Normal file
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const test = require('tape').test ;
const exec = require('child_process').exec ;
const pwd = process.env.TRAVIS ? '' : '-p$MYSQL_ROOT_PASSWORD';
test('creating jambones_test database', (t) => {
exec(`mysql -h localhost -u root ${pwd} < ${__dirname}/db/create_test_db.sql`, (err, stdout, stderr) => {
if (err) return t.end(err);
t.pass('database successfully created');
t.end();
});
});
test('creating schema', (t) => {
exec(`mysql -h localhost -u root ${pwd} -D jambones_test < ${__dirname}/db/jambones-sql.sql`, (err, stdout, stderr) => {
if (err) return t.end(err);
t.pass('schema successfully created');
t.end();
});
});
test('populating test case data', (t) => {
exec(`mysql -h localhost -u root ${pwd} -D jambones_test < ${__dirname}/db/populate-test-data.sql`, (err, stdout, stderr) => {
if (err) return t.end(err);
t.pass('test data set created');
t.end();
});
});

16
test/data/bad/bad-enum.json Normal file
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{
"dial": {
"action": "http://example.com",
"callerId": "+1312888899",
"target": [
{
"type": "foo",
"number": "sip:1617333456@sip.trunk1.com",
"auth": {
"user": "foo",
"password": "bar"
}
}
]
}
}

4
test/data/bad/bad-payload.json Normal file
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{
"key1": "value",
"key2": "value"
}

1
test/data/bad/bad-payload2.json Normal file
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[1, 2]

5
test/data/bad/invalid-type.json Normal file
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{
"sip:decline": {
"status": "hello"
}
}

15
test/data/bad/missing-required-property.json Normal file
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{
"dial": {
"action": "http://example.com",
"callerId": "+1312888899",
"target": [
{
"type": "sip",
"number": "sip:1617333456@sip.trunk1.com",
"auth": {
"password": "bar"
}
}
]
}
}

3
test/data/bad/unknown-instruction.json Normal file
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{
"foo": "bar"
}

6
test/data/bad/unknown-property.json Normal file
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{
"sip:decline": {
"status": 480,
"foo": "bar"
}
}

22
test/data/good/dial-listen.json Normal file
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{
"dial": {
"action": "http://example.com",
"callerId": "+1312888899",
"target": [
{
"type": "phone",
"number": "+15083084809"
}
],
"listen": {
"wsUrl": "wss://myrecorder.example.com:4433",
"mixType" : "stereo",
"sampleRate": 8000,
"passDtmf": true,
"source": "parent",
"metadata": {
"clientId": "12udih"
}
}
}
}

12
test/data/good/dial-phone.json Normal file
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{
"dial": {
"action": "http://example.com",
"callerId": "+1312888899",
"target": [
{
"type": "phone",
"number": "+15083084809"
}
]
}
}

16
test/data/good/dial-sip.json Normal file
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{
"dial": {
"action": "http://example.com",
"callerId": "+1312888899",
"target": [
{
"type": "sip",
"number": "sip:1617333456@sip.trunk1.com",
"auth": {
"user": "foo",
"password": "bar"
}
}
]
}
}

20
test/data/good/dial-transcribe.json Normal file
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{
"dial": {
"action": "http://example.com",
"callerId": "+1312888899",
"target": [
{
"type": "phone",
"number": "+15083084809"
}
],
"transcribe": {
"action": "http://example.com/transcribe",
"language" : "en-US",
"source" : "both",
"interim": true,
"vendor": "google",
"jsonKey": "--json service key--"
}
}
}

12
test/data/good/dial-user.json Normal file
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{
"dial": {
"action": "http://example.com",
"callerId": "+1312888899",
"target": [
{
"type": "user",
"name": "spike@sip.example.com"
}
]
}
}

9
test/data/good/sip-decline.json Normal file
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{
"sip:decline": {
"status": 480,
"reason": "Gone Fishin",
"headers": {
"Retry-After": 300
}
}
}

3
test/db/create_test_db.sql Normal file
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create database jambones_test;
create user jambones_test@localhost IDENTIFIED WITH mysql_native_password by 'jambones_test';
grant all on jambones_test.* to jambones_test@localhost;

272
test/db/jambones-sql.sql Normal file
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/* SQLEditor (MySQL (2))*/
DROP TABLE IF EXISTS `call_routes`;
DROP TABLE IF EXISTS `conference_participants`;
DROP TABLE IF EXISTS `queue_members`;
DROP TABLE IF EXISTS `calls`;
DROP TABLE IF EXISTS `phone_numbers`;
DROP TABLE IF EXISTS `applications`;
DROP TABLE IF EXISTS `conferences`;
DROP TABLE IF EXISTS `queues`;
DROP TABLE IF EXISTS `subscriptions`;
DROP TABLE IF EXISTS `registrations`;
DROP TABLE IF EXISTS `api_keys`;
DROP TABLE IF EXISTS `accounts`;
DROP TABLE IF EXISTS `service_providers`;
DROP TABLE IF EXISTS `sip_gateways`;
DROP TABLE IF EXISTS `voip_carriers`;
CREATE TABLE IF NOT EXISTS `applications`
(
`application_sid` CHAR(36) NOT NULL UNIQUE ,
`name` VARCHAR(255) NOT NULL,
`account_sid` CHAR(36) NOT NULL,
`call_hook` VARCHAR(255) NOT NULL,
`call_status_hook` VARCHAR(255) NOT NULL,
PRIMARY KEY (`application_sid`)
) ENGINE=InnoDB COMMENT='A defined set of behaviors to be applied to phone calls with';
CREATE TABLE IF NOT EXISTS `call_routes`
(
`call_route_sid` CHAR(36) NOT NULL UNIQUE ,
`order` INTEGER NOT NULL,
`account_sid` CHAR(36) NOT NULL,
`regex` VARCHAR(255) NOT NULL,
`application_sid` CHAR(36) NOT NULL,
PRIMARY KEY (`call_route_sid`)
) ENGINE=InnoDB;
CREATE TABLE IF NOT EXISTS `conferences`
(
`id` INTEGER(10) UNSIGNED NOT NULL AUTO_INCREMENT UNIQUE ,
`conference_sid` CHAR(36) NOT NULL UNIQUE ,
`name` VARCHAR(255),
PRIMARY KEY (`id`)
) ENGINE=InnoDB COMMENT='An audio conference';
CREATE TABLE IF NOT EXISTS `conference_participants`
(
`conference_participant_sid` CHAR(36) NOT NULL UNIQUE ,
`call_sid` CHAR(36),
`conference_sid` CHAR(36) NOT NULL,
PRIMARY KEY (`conference_participant_sid`)
) ENGINE=InnoDB COMMENT='A relationship between a call and a conference that it is co';
CREATE TABLE IF NOT EXISTS `queues`
(
`id` INTEGER(10) UNSIGNED NOT NULL AUTO_INCREMENT UNIQUE ,
`queue_sid` CHAR(36) NOT NULL UNIQUE ,
`name` VARCHAR(255),
PRIMARY KEY (`id`)
) ENGINE=InnoDB COMMENT='A set of behaviors to be applied to parked calls';
CREATE TABLE IF NOT EXISTS `registrations`
(
`registration_sid` CHAR(36) NOT NULL UNIQUE ,
`username` VARCHAR(255) NOT NULL,
`domain` VARCHAR(255) NOT NULL,
`sip_contact` VARCHAR(255) NOT NULL,
`sip_user_agent` VARCHAR(255),
PRIMARY KEY (`registration_sid`)
) ENGINE=InnoDB COMMENT='An active sip registration';
CREATE TABLE IF NOT EXISTS `queue_members`
(
`queue_member_sid` CHAR(36) NOT NULL UNIQUE ,
`call_sid` CHAR(36),
`queue_sid` CHAR(36) NOT NULL,
`position` INTEGER,
PRIMARY KEY (`queue_member_sid`)
) ENGINE=InnoDB COMMENT='A relationship between a call and a queue that it is waiting';
CREATE TABLE IF NOT EXISTS `calls`
(
`call_sid` CHAR(36) NOT NULL UNIQUE ,
`parent_call_sid` CHAR(36),
`application_sid` CHAR(36),
`status_url` VARCHAR(255),
`time_start` DATETIME NOT NULL,
`time_alerting` DATETIME,
`time_answered` DATETIME,
`time_ended` DATETIME,
`direction` ENUM('inbound','outbound'),
`phone_number_sid` CHAR(36),
`inbound_user_sid` CHAR(36),
`outbound_user_sid` CHAR(36),
`calling_number` VARCHAR(255),
`called_number` VARCHAR(255),
`caller_name` VARCHAR(255),
`status` VARCHAR(255) NOT NULL COMMENT 'Possible values are queued, ringing, in-progress, completed, failed, busy and no-answer',
`sip_uri` VARCHAR(255) NOT NULL,
`sip_call_id` VARCHAR(255) NOT NULL,
`sip_cseq` INTEGER NOT NULL,
`sip_from_tag` VARCHAR(255) NOT NULL,
`sip_via_branch` VARCHAR(255) NOT NULL,
`sip_contact` VARCHAR(255),
`sip_final_status` INTEGER UNSIGNED,
`sdp_offer` VARCHAR(4096),
`sdp_answer` VARCHAR(4096),
`source_address` VARCHAR(255) NOT NULL,
`source_port` INTEGER UNSIGNED NOT NULL,
`dest_address` VARCHAR(255),
`dest_port` INTEGER UNSIGNED,
`url` VARCHAR(255),
PRIMARY KEY (`call_sid`)
) ENGINE=InnoDB COMMENT='A phone call';
CREATE TABLE IF NOT EXISTS `service_providers`
(
`service_provider_sid` CHAR(36) NOT NULL UNIQUE ,
`name` VARCHAR(255) NOT NULL UNIQUE ,
`description` VARCHAR(255),
`root_domain` VARCHAR(255) UNIQUE ,
`registration_hook` VARCHAR(255),
`hook_basic_auth_user` VARCHAR(255),
`hook_basic_auth_password` VARCHAR(255),
PRIMARY KEY (`service_provider_sid`)
) ENGINE=InnoDB COMMENT='An organization that provides communication services to its ';
CREATE TABLE IF NOT EXISTS `api_keys`
(
`api_key_sid` CHAR(36) NOT NULL UNIQUE ,
`token` CHAR(36) NOT NULL UNIQUE ,
`account_sid` CHAR(36),
`service_provider_sid` CHAR(36),
PRIMARY KEY (`api_key_sid`)
) ENGINE=InnoDB COMMENT='An authorization token that is used to access the REST api';
CREATE TABLE IF NOT EXISTS `accounts`
(
`account_sid` CHAR(36) NOT NULL UNIQUE ,
`name` VARCHAR(255) NOT NULL,
`sip_realm` VARCHAR(255) UNIQUE ,
`service_provider_sid` CHAR(36) NOT NULL,
`registration_hook` VARCHAR(255),
`hook_basic_auth_user` VARCHAR(255),
`hook_basic_auth_password` VARCHAR(255),
`is_active` BOOLEAN NOT NULL DEFAULT true,
PRIMARY KEY (`account_sid`)
) ENGINE=InnoDB COMMENT='A single end-user of the platform';
CREATE TABLE IF NOT EXISTS `subscriptions`
(
`id` INTEGER(10) UNSIGNED NOT NULL AUTO_INCREMENT UNIQUE ,
`subscription_sid` CHAR(36) NOT NULL UNIQUE ,
`registration_sid` CHAR(36) NOT NULL,
`event` VARCHAR(255),
PRIMARY KEY (`id`)
) ENGINE=InnoDB COMMENT='An active sip subscription';
CREATE TABLE IF NOT EXISTS `voip_carriers`
(
`voip_carrier_sid` CHAR(36) NOT NULL UNIQUE ,
`name` VARCHAR(255) NOT NULL UNIQUE ,
`description` VARCHAR(255),
PRIMARY KEY (`voip_carrier_sid`)
) ENGINE=InnoDB COMMENT='An external organization that can provide sip trunking and D';
CREATE TABLE IF NOT EXISTS `phone_numbers`
(
`phone_number_sid` CHAR(36) UNIQUE ,
`number` VARCHAR(255) NOT NULL UNIQUE ,
`voip_carrier_sid` CHAR(36) NOT NULL,
`account_sid` CHAR(36),
`application_sid` CHAR(36),
PRIMARY KEY (`phone_number_sid`)
) ENGINE=InnoDB COMMENT='A phone number that has been assigned to an account';
CREATE TABLE IF NOT EXISTS `sip_gateways`
(
`sip_gateway_sid` CHAR(36),
`ipv4` VARCHAR(32) NOT NULL,
`port` INTEGER NOT NULL DEFAULT 5060,
`inbound` BOOLEAN NOT NULL,
`outbound` BOOLEAN NOT NULL,
`voip_carrier_sid` CHAR(36) NOT NULL,
`is_active` BOOLEAN NOT NULL DEFAULT true,
PRIMARY KEY (`sip_gateway_sid`)
);
CREATE UNIQUE INDEX `applications_idx_name` ON `applications` (`account_sid`,`name`);
CREATE INDEX `applications_application_sid_idx` ON `applications` (`application_sid`);
CREATE INDEX `applications_name_idx` ON `applications` (`name`);
CREATE INDEX `applications_account_sid_idx` ON `applications` (`account_sid`);
ALTER TABLE `applications` ADD FOREIGN KEY account_sid_idxfk (`account_sid`) REFERENCES `accounts` (`account_sid`);
CREATE INDEX `call_routes_call_route_sid_idx` ON `call_routes` (`call_route_sid`);
ALTER TABLE `call_routes` ADD FOREIGN KEY account_sid_idxfk_1 (`account_sid`) REFERENCES `accounts` (`account_sid`);
ALTER TABLE `call_routes` ADD FOREIGN KEY application_sid_idxfk (`application_sid`) REFERENCES `applications` (`application_sid`);
CREATE INDEX `conferences_conference_sid_idx` ON `conferences` (`conference_sid`);
CREATE INDEX `conference_participants_conference_participant_sid_idx` ON `conference_participants` (`conference_participant_sid`);
ALTER TABLE `conference_participants` ADD FOREIGN KEY call_sid_idxfk (`call_sid`) REFERENCES `calls` (`call_sid`);
ALTER TABLE `conference_participants` ADD FOREIGN KEY conference_sid_idxfk (`conference_sid`) REFERENCES `conferences` (`conference_sid`);
CREATE INDEX `queues_queue_sid_idx` ON `queues` (`queue_sid`);
CREATE INDEX `registrations_registration_sid_idx` ON `registrations` (`registration_sid`);
CREATE INDEX `queue_members_queue_member_sid_idx` ON `queue_members` (`queue_member_sid`);
ALTER TABLE `queue_members` ADD FOREIGN KEY call_sid_idxfk_1 (`call_sid`) REFERENCES `calls` (`call_sid`);
ALTER TABLE `queue_members` ADD FOREIGN KEY queue_sid_idxfk (`queue_sid`) REFERENCES `queues` (`queue_sid`);
CREATE INDEX `calls_call_sid_idx` ON `calls` (`call_sid`);
ALTER TABLE `calls` ADD FOREIGN KEY parent_call_sid_idxfk (`parent_call_sid`) REFERENCES `calls` (`call_sid`);
ALTER TABLE `calls` ADD FOREIGN KEY application_sid_idxfk_1 (`application_sid`) REFERENCES `applications` (`application_sid`);
CREATE INDEX `calls_phone_number_sid_idx` ON `calls` (`phone_number_sid`);
ALTER TABLE `calls` ADD FOREIGN KEY phone_number_sid_idxfk (`phone_number_sid`) REFERENCES `phone_numbers` (`phone_number_sid`);
ALTER TABLE `calls` ADD FOREIGN KEY inbound_user_sid_idxfk (`inbound_user_sid`) REFERENCES `registrations` (`registration_sid`);
ALTER TABLE `calls` ADD FOREIGN KEY outbound_user_sid_idxfk (`outbound_user_sid`) REFERENCES `registrations` (`registration_sid`);
CREATE INDEX `service_providers_service_provider_sid_idx` ON `service_providers` (`service_provider_sid`);
CREATE INDEX `service_providers_name_idx` ON `service_providers` (`name`);
CREATE INDEX `service_providers_root_domain_idx` ON `service_providers` (`root_domain`);
CREATE INDEX `api_keys_api_key_sid_idx` ON `api_keys` (`api_key_sid`);
CREATE INDEX `api_keys_account_sid_idx` ON `api_keys` (`account_sid`);
ALTER TABLE `api_keys` ADD FOREIGN KEY account_sid_idxfk_2 (`account_sid`) REFERENCES `accounts` (`account_sid`);
CREATE INDEX `api_keys_service_provider_sid_idx` ON `api_keys` (`service_provider_sid`);
ALTER TABLE `api_keys` ADD FOREIGN KEY service_provider_sid_idxfk (`service_provider_sid`) REFERENCES `service_providers` (`service_provider_sid`);
CREATE INDEX `accounts_account_sid_idx` ON `accounts` (`account_sid`);
CREATE INDEX `accounts_name_idx` ON `accounts` (`name`);
CREATE INDEX `accounts_sip_realm_idx` ON `accounts` (`sip_realm`);
CREATE INDEX `accounts_service_provider_sid_idx` ON `accounts` (`service_provider_sid`);
ALTER TABLE `accounts` ADD FOREIGN KEY service_provider_sid_idxfk_1 (`service_provider_sid`) REFERENCES `service_providers` (`service_provider_sid`);
ALTER TABLE `subscriptions` ADD FOREIGN KEY registration_sid_idxfk (`registration_sid`) REFERENCES `registrations` (`registration_sid`);
CREATE INDEX `voip_carriers_voip_carrier_sid_idx` ON `voip_carriers` (`voip_carrier_sid`);
CREATE INDEX `voip_carriers_name_idx` ON `voip_carriers` (`name`);
CREATE INDEX `phone_numbers_phone_number_sid_idx` ON `phone_numbers` (`phone_number_sid`);
CREATE INDEX `phone_numbers_voip_carrier_sid_idx` ON `phone_numbers` (`voip_carrier_sid`);
ALTER TABLE `phone_numbers` ADD FOREIGN KEY voip_carrier_sid_idxfk (`voip_carrier_sid`) REFERENCES `voip_carriers` (`voip_carrier_sid`);
ALTER TABLE `phone_numbers` ADD FOREIGN KEY account_sid_idxfk_3 (`account_sid`) REFERENCES `accounts` (`account_sid`);
ALTER TABLE `phone_numbers` ADD FOREIGN KEY application_sid_idxfk_2 (`application_sid`) REFERENCES `applications` (`application_sid`);
CREATE UNIQUE INDEX `sip_gateways_sip_gateway_idx_hostport` ON `sip_gateways` (`ipv4`,`port`);
ALTER TABLE `sip_gateways` ADD FOREIGN KEY voip_carrier_sid_idxfk_1 (`voip_carrier_sid`) REFERENCES `voip_carriers` (`voip_carrier_sid`);

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test/db/populate-test-data.sql Normal file
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insert into service_providers (service_provider_sid, name, root_domain, registration_hook, hook_basic_auth_user, hook_basic_auth_password)
values ('3f35518f-5a0d-4c2e-90a5-2407bb3b36f0', 'SP A', 'jambonz.org', 'http://127.0.0.1:4000/auth', 'foo', 'bar');
insert into accounts(account_sid, service_provider_sid, name, sip_realm, registration_hook, hook_basic_auth_user, hook_basic_auth_password)
values ('ed649e33-e771-403a-8c99-1780eabbc803', '3f35518f-5a0d-4c2e-90a5-2407bb3b36f0', 'test account', 'sip.example.com', 'http://127.0.0.1:4000/auth', 'foo', 'bar');
insert into voip_carriers (voip_carrier_sid, name) values ('287c1452-620d-4195-9f19-c9814ef90d78', 'westco');
insert into sip_gateways (sip_gateway_sid, voip_carrier_sid, ipv4, inbound, outbound)
values ('124a5339-c62c-4075-9e19-f4de70a96597', '287c1452-620d-4195-9f19-c9814ef90d78', '172.38.0.20', true, true);
insert into sip_gateways (sip_gateway_sid, voip_carrier_sid, ipv4, port, inbound, outbound)
values ('efbc4830-57cd-4c78-a56f-d64fdf210fe8', '287c1452-620d-4195-9f19-c9814ef90d78', '3.3.3.3', 5062, false, true);

3
test/db/remove_test_db.sql Normal file
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DROP DATABASE jambones_test;
REVOKE ALL PRIVILEGES, GRANT OPTION FROM 'jambones_test'@'localhost';
DROP USER 'jambones_test'@'localhost';

55
test/docker-compose-testbed.yaml Normal file
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version: '3'
networks:
sbc-inbound:
driver: bridge
ipam:
config:
- subnet: 172.38.0.0/16
services:
sbc:
image: drachtio/drachtio-server:latest
command: drachtio --contact "sip:*;transport=udp" --loglevel debug --sofia-loglevel 9
ports:
- "9060:9022/tcp"
networks:
sbc-inbound:
ipv4_address: 172.38.0.10
appserver:
image: drachtio/sipp:latest
command: sipp -sf /tmp/uas.xml
volumes:
- ./scenarios:/tmp
tty: true
networks:
sbc-inbound:
ipv4_address: 172.38.0.11
auth-server:
image: jambonz/customer-auth-server:latest
command: npm start
ports:
- "4000:4000/tcp"
env_file: docker.env
networks:
sbc-inbound:
ipv4_address: 172.38.0.12
redis:
image: redis:5-alpine
ports:
- "16379:6379/tcp"
networks:
sbc-inbound:
ipv4_address: 172.38.0.13
rtpengine:
image: drachtio/rtpengine:latest
ports:
- "12222:22222/udp"
networks:
sbc-inbound:
ipv4_address: 172.38.0.14

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test/docker.env Normal file
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# tip: to minify json into a single string as below, see https://codebeautify.org/jsonminifier
NODE_CONFIG={"server":{"port":4000,"path":"/auth","auth":{"users":{"foo":"bar"}}},"credentials":{"jambonz.org":[{"username":"john","password":"1234"},{"username":"jane","password":"5678"}]}}

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test/docker_start.js Normal file
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const test = require('tape').test ;
const exec = require('child_process').exec ;
const async = require('async');
test('starting docker network..', (t) => {
exec(`docker-compose -f ${__dirname}/docker-compose-testbed.yaml up -d`, (err, stdout, stderr) => {
t.end(err);
});
});

12
test/docker_stop.js Normal file
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const test = require('tape').test ;
const exec = require('child_process').exec ;
test('stopping docker network..', (t) => {
t.timeoutAfter(10000);
exec(`docker-compose -f ${__dirname}/docker-compose-testbed.yaml down`, (err, stdout, stderr) => {
//console.log(`stderr: ${stderr}`);
process.exit(0);
});
t.end() ;
});

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test/index.js Normal file
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require('./unit-tests');
/*
require('./docker_start');
require('./create-test-db');
require('./sip-tests');
require('./remove-test-db');
require('./docker_stop');
*/

11
test/remove-test-db.js Normal file
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const test = require('tape').test ;
const exec = require('child_process').exec ;
const pwd = process.env.TRAVIS ? '' : '-p$MYSQL_ROOT_PASSWORD';
test('dropping jambones_test database', (t) => {
exec(`mysql -h localhost -u root ${pwd} < ${__dirname}/db/remove_test_db.sql`, (err, stdout, stderr) => {
if (err) return t.end(err);
t.pass('database successfully dropped');
t.end();
});
});

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test/scenarios/pcap/g711a.pcap Normal file
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test/scenarios/uac-device-invalid-password.xml Normal file
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-device-unknown-user
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send>
<![CDATA[
ACK sip:[service]@jambonz.org SIP/2.0
[last_Via]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Subject: uac-device-unknown-user
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
[authentication username=john password=2222]
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-device-unknown-user
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="403">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@jambonz.org SIP/2.0
[last_Via]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Subject: uac-device-unknown-user
Content-Length: 0
]]>
</send>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

76
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@@ -0,0 +1,76 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@jambones.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-device-unknown-user
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="403">
</recv>
<send>
<![CDATA[
ACK sip:[service]@jambones.org SIP/2.0
[last_Via]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Subject: uac-device-unknown-user
Content-Length: 0
]]>
</send>
</scenario>

132
test/scenarios/uac-device-unknown-user.xml Normal file
View File

@@ -0,0 +1,132 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-device-unknown-user
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send>
<![CDATA[
ACK sip:[service]@jambonz.org SIP/2.0
[last_Via]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Subject: uac-device-unknown-user
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
[authentication username=jay password=1234]
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-device-unknown-user
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="403">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@jambonz.org SIP/2.0
[last_Via]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Subject: uac-device-unknown-user
Content-Length: 0
]]>
</send>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

44
test/scenarios/uac-info-expect-480.xml Normal file
View File

@@ -0,0 +1,44 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INFO sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INFO
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="480" rtd="true"/>
</scenario>

44
test/scenarios/uac-message-expect-480.xml Normal file
View File

@@ -0,0 +1,44 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
MESSAGE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 MESSAGE
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="480" rtd="true"/>
</scenario>

44
test/scenarios/uac-options-expect-480.xml Normal file
View File

@@ -0,0 +1,44 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
OPTIONS sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 OPTIONS
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="480" rtd="true"/>
</scenario>

119
test/scenarios/uac-pcap-carrier-success.xml Normal file
View File

@@ -0,0 +1,119 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-pcap-carrier-success
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true" crlf="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Max-Forwards: 70
Subject: uac-pcap-carrier-success
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!-- Pause briefly -->
<pause milliseconds="3000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Subject: uac-pcap-carrier-success
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

187
test/scenarios/uac-pcap-device-success-in-dialog-request.xml Normal file
View File

@@ -0,0 +1,187 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-pcap-device-success
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send>
<![CDATA[
ACK sip:[service]@jambonz.org SIP/2.0
[last_Via]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Subject: uac-pcap-device-success
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
[authentication username=john password=1234]
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-pcap-device-success
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200" rtd="true" crlf="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Subject: uac-pcap-device-success
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!-- Pause briefly -->
<pause milliseconds="1000"/>
<send retrans="500">
<![CDATA[
INFO sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 INFO
Max-Forwards: 70
Event: X-custom-Event
Subject: uac-pcap-device-success
Content-Type: application/text
Content-Length: [len]
foobar
]]>
</send>
<recv response="200" crlf="true">
</recv>
<send retrans="500">
<![CDATA[
BYE sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 4 BYE
Max-Forwards: 70
Subject: uac-pcap-device-success
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

165
test/scenarios/uac-pcap-device-success.xml Normal file
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-pcap-device-success
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send>
<![CDATA[
ACK sip:[service]@jambonz.org SIP/2.0
[last_Via]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Subject: uac-pcap-device-success
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
[authentication username=john password=1234]
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-pcap-device-success
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200" rtd="true" crlf="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Subject: uac-pcap-device-success
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!-- Pause briefly -->
<pause milliseconds="3000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[service]@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 BYE
Max-Forwards: 70
Subject: uac-pcap-device-success
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

45
test/scenarios/uac-publish-expect-480.xml Normal file
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
PUBLISH sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 PUBLISH
Contact: <sip:sipp@[local_ip]:[local_port]>;expires=3600
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="480" rtd="true"/>
</scenario>

45
test/scenarios/uac-register-expect-480.xml Normal file
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
REGISTER sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: <sip:sipp@[local_ip]:[local_port]>;expires=3600
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="480" rtd="true"/>
</scenario>

45
test/scenarios/uac-subscribe-expect-480.xml Normal file
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
SUBSCRIBE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 SUBSCRIBE
Contact: <sip:sipp@[local_ip]:[local_port]>;expires=3600
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="480" rtd="true"/>
</scenario>

118
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

130
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder">
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv request="INVITE" crlf="true">
<action>
<ereg regexp=".*" search_in="hdr" header="Subject:" assign_to="1" />
</action>
</recv>
<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurrences are concatenated (CRLF separated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
[last_Record-Route:]
Subject:[$1]
Content-Length: 0
]]>
</send>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
[last_Record-Route:]
Subject:[$1]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<recv request="INFO" optional="true" next="1">
</recv>
<recv request="BYE">
</recv>
<send next="2">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<label id="1"/>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<label id="2"/>
</scenario>

68
test/sip-tests.js Normal file
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const test = require('blue-tape');
const { output, sippUac } = require('./sipp')('test_sbc-inbound');
const debug = require('debug')('drachtio:sbc-inbound');
const clearModule = require('clear-module');
process.on('unhandledRejection', (reason, p) => {
console.log('Unhandled Rejection at: Promise', p, 'reason:', reason);
});
function connect(connectable) {
return new Promise((resolve, reject) => {
connectable.on('connect', () => {
return resolve();
});
});
}
test('incoming call tests', (t) => {
const {srf} = require('../app');
connect(srf)
.then(() => {
return sippUac('uac-pcap-carrier-success.xml', '172.38.0.20');
})
.then(() => {
return t.pass('incoming call from carrier completed successfully');
})
.then(() => {
return sippUac('uac-pcap-device-success.xml', '172.38.0.30');
})
.then(() => {
return t.pass('incoming call from authenticated device completed successfully');
})
.then(() => {
return sippUac('uac-device-unknown-user.xml', '172.38.0.30');
})
.then(() => {
return t.pass('unknown user is rejected with a 403');
})
.then(() => {
return sippUac('uac-device-unknown-realm.xml', '172.38.0.30');
})
.then(() => {
return t.pass('unknown realm is rejected with a 403');
})
.then(() => {
return sippUac('uac-device-invalid-password.xml', '172.38.0.30');
})
.then(() => {
return t.pass('invalid password for valid user is rejected with a 403');
})
.then(() => {
return sippUac('uac-pcap-device-success-in-dialog-request.xml', '172.38.0.30');
})
.then(() => {
return t.pass('handles in-dialog requests');
})
.then(() => {
srf.disconnect();
t.end();
return;
})
.catch((err) => {
console.log(`error received: ${err}`);
if (srf) srf.disconnect();
t.error(err);
});
});

68
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const { spawn } = require('child_process');
const debug = require('debug')('test:sipp');
let network;
const obj = {};
let output = '';
let idx = 1;
function clearOutput() {
output = '';
}
function addOutput(str) {
for (let i = 0; i < str.length; i++) {
if (str.charCodeAt(i) < 128) output += str.charAt(i);
}
}
module.exports = (networkName) => {
network = networkName ;
return obj;
};
obj.output = () => {
return output;
};
obj.sippUac = (file, bindAddress) => {
const cmd = 'docker';
const args = [
'run', '-ti', '--rm', '--net', `${network}`,
'-v', `${__dirname}/scenarios:/tmp/scenarios`,
'drachtio/sipp', 'sipp', '-sf', `/tmp/scenarios/${file}`,
'-m', '1',
'-sleep', '250ms',
'-nostdin',
'-cid_str', `%u-%p@%s-${idx++}`,
'sbc'
];
if (bindAddress) args.splice(5, 0, '--ip', bindAddress);
//console.log(args.join(' '));
clearOutput();
return new Promise((resolve, reject) => {
const child_process = spawn(cmd, args, {stdio: ['inherit', 'pipe', 'pipe']});
child_process.on('exit', (code, signal) => {
if (code === 0) {
return resolve();
}
console.log(`sipp exited with non-zero code ${code} signal ${signal}`);
reject(code);
});
child_process.on('error', (error) => {
console.log(`error spawing child process for docker: ${args}`);
});
child_process.stdout.on('data', (data) => {
//debug(`stdout: ${data}`);
addOutput(data.toString());
});
child_process.stdout.on('data', (data) => {
//debug(`stdout: ${data}`);
addOutput(data.toString());
});
});
};

49
test/unit-tests.js Normal file
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const test = require('tape');
const debug = require('debug')('drachtio:jambonz:test');
const makeTask = require('../lib/tasks/make_task');
const noop = () => {};
const logger = {error: noop, info: noop, debug: noop};
process.on('unhandledRejection', (reason, p) => {
console.log('Unhandled Rejection at: Promise', p, 'reason:', reason);
});
test('app payload parsing tests', (t) => {
let task = makeTask(logger, require('./data/good/sip-decline'));
t.ok(task.name === 'sip:decline', 'parsed sip:decline');
t.throws(errInvalidInstruction, /invalid instruction/, 'throws error for invalid instruction');
t.throws(errUnknownProperty, /unknown property/, 'throws error for invalid instruction');
t.throws(errMissingProperty, /missing value/, 'throws error for missing required property');
t.throws(errInvalidType, /invalid data type/, 'throws error for invalid data type');
t.throws(errBadEnum, /must be one of/, 'throws error for invalid enum');
t.throws(errBadPayload, /invalid instruction payload/, 'throws error for invalid payload with multiple keys');
t.throws(errBadPayload2, /invalid instruction payload/, 'throws error for invalid payload that is not an object');
task = makeTask(logger, require('./data/good/dial-phone'));
t.ok(task.name === 'dial', 'parsed dial phone');
task = makeTask(logger, require('./data/good/dial-sip'));
t.ok(task.name === 'dial', 'parsed dial sip');
task = makeTask(logger, require('./data/good/dial-user'));
t.ok(task.name === 'dial', 'parsed dial user');
task = makeTask(logger, require('./data/good/dial-transcribe'));
t.ok(task.name === 'dial', 'parsed dial w/ transcribe');
task = makeTask(logger, require('./data/good/dial-listen'));
t.ok(task.name === 'dial', 'parsed dial w/ listen');
t.end();
});
const errInvalidInstruction = () => makeTask(logger, require('./data/bad/unknown-instruction'));
const errUnknownProperty = () => makeTask(logger, require('./data/bad/unknown-property'));
const errMissingProperty = () => makeTask(logger, require('./data/bad/missing-required-property'));
const errInvalidType = () => makeTask(logger, require('./data/bad/invalid-type'));
const errBadEnum = () => makeTask(logger, require('./data/bad/bad-enum'));
const errBadPayload = () => makeTask(logger, require('./data/bad/bad-payload'));
const errBadPayload2 = () => makeTask(logger, require('./data/bad/bad-payload2'));