ben.carrasco/sbc-outbound
1
0
Fork 0
mirror of https://github.com/jambonz/sbc-outbound.git synced 2025-12-19 04:27:45 +00:00
Code Issues Packages Projects Releases Wiki Activity

initial checkin

Browse Source
This commit is contained in:
Dave Horton
2019-08-25 10:58:29 -04:00
commit e2896aba9a
44 changed files with 7386 additions and 0 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
.eslintignore Normal file
View File

@@ -0,0 +1 @@
test/*

126
.eslintrc.json Normal file
View File

@@ -0,0 +1,126 @@
{
"env": {
"node": true,
"es6": true
},
"parserOptions": {
"ecmaFeatures": {
"jsx": false,
"modules": false
},
"ecmaVersion": 2017
},
"plugins": ["promise"],
"rules": {
"promise/always-return": "error",
"promise/no-return-wrap": "error",
"promise/param-names": "error",
"promise/catch-or-return": "error",
"promise/no-native": "off",
"promise/no-nesting": "warn",
"promise/no-promise-in-callback": "warn",
"promise/no-callback-in-promise": "warn",
"promise/no-return-in-finally": "warn",
// Possible Errors
// http://eslint.org/docs/rules/#possible-errors
"comma-dangle": [2, "only-multiline"],
"no-control-regex": 2,
"no-debugger": 2,
"no-dupe-args": 2,
"no-dupe-keys": 2,
"no-duplicate-case": 2,
"no-empty-character-class": 2,
"no-ex-assign": 2,
"no-extra-boolean-cast" : 2,
"no-extra-parens": [2, "functions"],
"no-extra-semi": 2,
"no-func-assign": 2,
"no-invalid-regexp": 2,
"no-irregular-whitespace": 2,
"no-negated-in-lhs": 2,
"no-obj-calls": 2,
"no-proto": 2,
"no-unexpected-multiline": 2,
"no-unreachable": 2,
"use-isnan": 2,
"valid-typeof": 2,
// Best Practices
// http://eslint.org/docs/rules/#best-practices
"no-fallthrough": 2,
"no-octal": 2,
"no-redeclare": 2,
"no-self-assign": 2,
"no-unused-labels": 2,
// Strict Mode
// http://eslint.org/docs/rules/#strict-mode
"strict": [2, "never"],
// Variables
// http://eslint.org/docs/rules/#variables
"no-delete-var": 2,
"no-undef": 2,
"no-unused-vars": [2, {"args": "none"}],
// Node.js and CommonJS
// http://eslint.org/docs/rules/#nodejs-and-commonjs
"no-mixed-requires": 2,
"no-new-require": 2,
"no-path-concat": 2,
"no-restricted-modules": [2, "sys", "_linklist"],
// Stylistic Issues
// http://eslint.org/docs/rules/#stylistic-issues
"comma-spacing": 2,
"eol-last": 2,
"indent": [2, 2, {"SwitchCase": 1}],
"keyword-spacing": 2,
"max-len": [2, 120, 2],
"new-parens": 2,
"no-mixed-spaces-and-tabs": 2,
"no-multiple-empty-lines": [2, {"max": 2}],
"no-trailing-spaces": [2, {"skipBlankLines": false }],
"quotes": [2, "single", "avoid-escape"],
"semi": 2,
"space-before-blocks": [2, "always"],
"space-before-function-paren": [2, "never"],
"space-in-parens": [2, "never"],
"space-infix-ops": 2,
"space-unary-ops": 2,
// ECMAScript 6
// http://eslint.org/docs/rules/#ecmascript-6
"arrow-parens": [2, "always"],
"arrow-spacing": [2, {"before": true, "after": true}],
"constructor-super": 2,
"no-class-assign": 2,
"no-confusing-arrow": 2,
"no-const-assign": 2,
"no-dupe-class-members": 2,
"no-new-symbol": 2,
"no-this-before-super": 2,
"prefer-const": 2
},
"globals": {
"DTRACE_HTTP_CLIENT_REQUEST" : false,
"LTTNG_HTTP_CLIENT_REQUEST" : false,
"COUNTER_HTTP_CLIENT_REQUEST" : false,
"DTRACE_HTTP_CLIENT_RESPONSE" : false,
"LTTNG_HTTP_CLIENT_RESPONSE" : false,
"COUNTER_HTTP_CLIENT_RESPONSE" : false,
"DTRACE_HTTP_SERVER_REQUEST" : false,
"LTTNG_HTTP_SERVER_REQUEST" : false,
"COUNTER_HTTP_SERVER_REQUEST" : false,
"DTRACE_HTTP_SERVER_RESPONSE" : false,
"LTTNG_HTTP_SERVER_RESPONSE" : false,
"COUNTER_HTTP_SERVER_RESPONSE" : false,
"DTRACE_NET_STREAM_END" : false,
"LTTNG_NET_STREAM_END" : false,
"COUNTER_NET_SERVER_CONNECTION_CLOSE" : false,
"DTRACE_NET_SERVER_CONNECTION" : false,
"LTTNG_NET_SERVER_CONNECTION" : false,
"COUNTER_NET_SERVER_CONNECTION" : false
}
}

38
.gitignore vendored Normal file
View File

@@ -0,0 +1,38 @@
# Logs
logs
*.log
# Runtime data
pids
*.pid
*.seed
# github pages site
_site
#transient test cases
examples/nosave.*.js
# Directory for instrumented libs generated by jscoverage/JSCover
lib-cov
# Coverage directory used by tools like istanbul
coverage
.nyc_output/
# Grunt intermediate storage (http://gruntjs.com/creating-plugins#storing-task-files)
.grunt
# node-waf configuration
.lock-wscript
# Compiled binary addons (http://nodejs.org/api/addons.html)
build/Release
# Dependency directory
# https://www.npmjs.org/doc/misc/npm-faq.html#should-i-check-my-node_modules-folder-into-git
node_modules
.DS_Store
examples/*

9
.travis.yml Normal file
View File

@@ -0,0 +1,9 @@
sudo: required
language: node_js
node_js:
- "lts/*"
services:
- docker
script:
- npm test

21
LICENSE Normal file
View File

@@ -0,0 +1,21 @@
MIT License
Copyright (c) 2019 jambonz
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.

81
README.md Normal file
View File

@@ -0,0 +1,81 @@
# sbc-outbound [![Build Status](https://secure.travis-ci.org/jambonz/sbc-outbound.png)](http://travis-ci.org/jambonz/sbc-outbound)
This application provides a part of the SBC (Session Border Controller) functionality of jambonz. It handles outbound INVITE requests from the cpaas application server that is going to carrier sip trunks or registered sip users/devices, including webrtc applications.
## Configuration
Configuration is provided via the [npmjs config](https://www.npmjs.com/package/config) package. Much of the configuration is standard "where is drachtio?, where is rtpengine?, where is redis?" type of thing. This application also needs information about the sip trunk to send PSTN calls out on. In the initial release, only a single outbound SIP trunk is supported.
The following elements make up the configuration for the application:
##### outbound sip trunk
```
"trunks": {
"outbound": {
"name": "carrier1",
"host": "sip:172.39.0.20"
}
},
```
The sip uri specifies the sip trunk uri to send PSTN calls to. All outbound PSTN calls from the cpaas application server are expected to be in E.164 format with a leading plus (+) sign. Outbound sip uris that do not begin with a '+' are assumed to be calls to registered devices/users.
##### drachtio server location
```
"drachtio": {
"port": 3001,
"secret": "cymru"
},
```
the `drachtio` object specifies the port to listen on for tcp connections from drachtio servers as well as the shared secret that is used to authenticate to the server.
> Note: either inbound or [outbound connections](https://drachtio.org/docs#outbound-connections) may be used, depending on the configuration supplied. In production, it is the intent to use outbound connections for easier centralization and clustering of application logic, while inbound connections are used for the automated test suite.
##### rtpengine location
```
"rtpengine": {
"host": "127.0.0.1",
"port": 22222
},
```
the `rtpengine` object specifies the location of the rtpengine, which will typically be running on the same server as drachtio.
##### redis location
```
"redis": {
"port": 6379,
"host": "127.0.0.1"
},
```
the `redis` object specifies the ip/dns and port that redis is listening on.
##### application log level
```
"logging": {
"level": "info"
}
```
##### transcoding options
The transcoding options for rtpengine are found in the configuration file, however these should not need to be modified.
```
"transcoding": {
"rtpCharacteristics" : {
"transport protocol": "RTP/AVP",
"DTLS": "off",
"SDES": "off",
"ICE": "remove",
"rtcp-mux": ["demux"]
},
"srtpCharacteristics": {
"transport-protocol": "UDP/TLS/RTP/SAVPF",
"ICE": "force",
"SDES": "off",
"flags": ["generate mid", "SDES-no"],
"rtcp-mux": ["require"]
}
}
```
## Tests
The automated test suite requires a docker installation.
```
npm test
```

35
app.js Normal file
View File

@@ -0,0 +1,35 @@
const Srf = require('drachtio-srf');
const srf = new Srf();
const config = require('config');
const logger = require('pino')(config.get('logging'));
const {route, setLogger} = require('./lib/middleware');
const debug = require('debug')('jambonz:sbc-outbound');
// disable logging in test mode
if (process.env.NODE_ENV === 'test') {
const noop = () => {};
logger.info = logger.debug = noop;
logger.child = () => {return {info: noop, error: noop, debug: noop};};
}
// config dictates whether to use outbound or inbound connections
if (config.has('drachtio.host')) {
srf.connect(config.get('drachtio'));
srf.on('connect', (err, hp) => {
debug(`connected to drachtio at ${hp}`);
logger.info(`connected to drachtio listening on ${hp}`);
});
}
else {
srf.listen(config.get('drachtio'));
}
if (process.env.NODE_ENV === 'test') {
srf.on('error', (err) => {
logger.error(err, 'Error connecting to drachtio');
});
}
srf.use('invite', [setLogger(logger), route(config.get('redis'))]);
srf.invite(require('./lib/invite'));
module.exports = {srf};

39
config/default.json Normal file
View File

@@ -0,0 +1,39 @@
{
"drachtio": {
"port": 3003,
"secret": "cymru"
},
"rtpengine": {
"host": "127.0.0.1",
"port": 22222
},
"redis": {
"port": 6379,
"host": "127.0.0.1"
},
"logging": {
"level": "info"
},
"trunks": {
"outbound": {
"name": "carrier1",
"host": "sip:10.123.22.3"
}
},
"transcoding": {
"rtpCharacteristics" : {
"transport protocol": "RTP/AVP",
"DTLS": "off",
"SDES": "off",
"ICE": "remove",
"rtcp-mux": ["demux"]
},
"srtpCharacteristics": {
"transport-protocol": "UDP/TLS/RTP/SAVPF",
"ICE": "force",
"SDES": "off",
"flags": ["generate mid", "SDES-no"],
"rtcp-mux": ["require"]
}
}
}

40
config/local-test.json Normal file
View File

@@ -0,0 +1,40 @@
{
"drachtio": {
"host": "127.0.0.1",
"port": 9060,
"secret": "cymru"
},
"rtpengine": {
"host": "127.0.0.1",
"port": 12222
},
"redis": {
"port": 16379,
"host": "127.0.0.1"
},
"logging": {
"level": "info"
},
"trunks": {
"outbound": {
"name": "carrier1",
"host": "sip:172.39.0.20"
}
},
"transcoding": {
"rtpCharacteristics" : {
"transport protocol": "RTP/AVP",
"DTLS": "off",
"SDES": "off",
"ICE": "remove",
"rtcp-mux": ["demux"]
},
"srtpCharacteristics": {
"transport-protocol": "UDP/TLS/RTP/SAVPF",
"ICE": "force",
"SDES": "off",
"flags": ["generate mid", "SDES-no"],
"rtcp-mux": ["require"]
}
}
}

94
lib/invite.js Normal file
View File

@@ -0,0 +1,94 @@
const config = require('config');
const Client = require('rtpengine-client').Client ;
const rtpengine = new Client();
const offer = rtpengine.offer.bind(rtpengine, config.get('rtpengine'));
const answer = rtpengine.answer.bind(rtpengine, config.get('rtpengine'));
const del = rtpengine.delete.bind(rtpengine, config.get('rtpengine'));
const {makeRtpEngineOpts} = require('./utils');
const {forwardInDialogRequests} = require('drachtio-fn-b2b-sugar');
const debug = require('debug')('jambonz:sbc-outbound');
module.exports = handler;
async function handler(req, res) {
const logger = req.locals.logger;
const srf = req.srf;
const useWss = req.locals.registration && req.locals.registration.protocol === 'wss';
const rtpEngineOpts = makeRtpEngineOpts(req, false, useWss);
const rtpEngineResource = {destroy: del.bind(rtpengine, rtpEngineOpts.common)};
try {
const response = await offer(rtpEngineOpts.offer);
if ('ok' !== response.result) {
res.send(480);
throw new Error(`failed allocating rtpengine endpoint: ${JSON.stringify(response)}`);
}
let uri, proxy;
if (req.locals.registration) {
proxy = req.locals.registration.contact;
uri = req.uri;
}
else {
uri = config.get('trunks.outbound.host');
}
const {uas, uac} = await srf.createB2BUA(req, res, uri, {
proxy,
proxyRequestHeaders: ['User-Agent', 'Subject'],
localSdpB: response.sdp,
localSdpA: (sdp, res) => {
const opts = Object.assign({sdp, 'to-tag': res.getParsedHeader('To').params.tag},
rtpEngineOpts.answer);
return answer(opts)
.then((response) => {
if ('ok' !== response.result) throw new Error('error allocating rtpengine');
return response.sdp;
});
}
});
logger.info('call connected');
debug('call connected');
setHandlers(logger, uas, uac, rtpEngineOpts, rtpEngineResource);
} catch (err) {
logger.error(err, 'Error connecting call');
rtpEngineResource.destroy();
}
}
function setHandlers(logger, uas, uac, rtpEngineOpts, rtpEngineResource) {
[uas, uac].forEach((dlg) => {
//hangup
dlg.on('destroy', () => {
logger.info('call ended');
rtpEngineResource.destroy();
});
//re-invite
dlg.on('modify', onReinvite.bind(dlg, logger, rtpEngineOpts));
});
// default forwarding of other request types
forwardInDialogRequests(uas);
}
async function onReinvite(logger, rtpEngineOpts, req, res) {
try {
let response = await offer(Object.assign({sdp: req.body}, rtpEngineOpts.offer));
if ('ok' !== response.result) {
res.send(488);
throw new Error(`failed allocating rtpengine endpoint: ${JSON.stringify(response)}`);
}
const sdp = await this.other.modify(response.sdp);
const opts = Object.assign({sdp, 'to-tag': res.getParsedHeader('To').params.tag},
rtpEngineOpts.answer);
response = await answer(opts);
if ('ok' !== response.result) {
res.send(488);
throw new Error(`failed allocating rtpengine endpoint: ${JSON.stringify(response)}`);
}
res.send(200, {body: response.sdp});
} catch (err) {
logger.error(err, 'Error handling reinvite');
}
}

67
lib/middleware.js Normal file
View File

@@ -0,0 +1,67 @@
const debug = require('debug')('jambonz:sbc-outbound');
const parseUri = require('drachtio-srf').parseUri;
const Registrar = require('jambonz-mw-registrar');
const {selectHostPort}= require('./utils');
function setLogger(logger) {
return (req, res, next) => {
debug('setting logger');
req.locals = req.locals || {};
req.locals.logger = logger.child({callId: req.get('Call-ID')});
next();
};
};
function route(opts) {
const registrar = new Registrar(opts);
return async(req, res, next) => {
const logger = req.locals.logger;
req.locals = req.locals || {};
logger.info(`received outbound INVITE to ${req.calledNumber} from server at ${req.server.hostport}`);
debug(`received outbound INVITE to ${req.calledNumber} from server at ${req.server.hostport}`);
// E.164 numbers go through least-cost routing and out a sip trunk
if (req.calledNumber.startsWith('+')) {
logger.info('sending call to LCR');
req.locals.target = 'lcr';
return next();
}
// otherwise, its a call to a sip user
const uri = parseUri(req.uri);
const aor = `sip:${uri.user}@${uri.host}`;
debug(`searching for registered user ${aor}`);
const reg = await registrar.query(aor);
if (!reg) {
logger.info(`sbc-outbound call to a non-registered user ${aor}`);
debug(`sbc-outbound call to a non-registered user ${aor}`);
return res.send(404, 'User Not Found');
}
// user is registered..find out which sbc is handling it
// if us => we can put the call through
// if another sbc => proxy the call there
if (req.server.hostport !== reg.sbcAddress) {
//proxy
const proxyAddress = selectHostPort(reg.sbcAddress, 'udp');
logger.info(`proxying call to SBC at ${proxyAddress}`);
debug(`proxying call to SBC at ${proxyAddress}`);
req.proxy(`sip:${proxyAddress[1]}`, {followRedirects: true});
return;
}
req.locals.registration = reg;
req.locals.target = 'user';
logger.info(`sending call to registered user at ${JSON.stringify(req.locals.registration)}`);
debug(`sending call to registered user at ${JSON.stringify(req.locals.registration)}`);
next();
};
}
module.exports = {
setLogger,
route
};

57
lib/utils.js Normal file
View File

@@ -0,0 +1,57 @@
const config = require('config');
let idx = 0;
const debug = require('debug')('jambonz:sbc-outbound');
function fromInboundTrunk(req) {
const trunks = config.has('trunks.inbound') ?
config.get('trunks.inbound') : [];
if (isWSS(req)) return false;
const trunk = trunks.find((t) => t.host.includes(req.source_address));
if (!trunk) return false;
req.carrier_name = trunk.name;
return true;
}
function isWSS(req) {
return req.getParsedHeader('Via')[0].protocol.toLowerCase().startsWith('ws');
}
function selectHostPort(hostport, protocol) {
debug(`selectHostPort: ${hostport}, ${protocol}`);
const sel = hostport
.split(',')
.map((hp) => {
const arr = /(.*)\/(.*):(.*)/.exec(hp);
return [arr[1], arr[2], arr[3]];
})
.filter((hp) => {
return hp[0] === protocol && hp[1] !== '127.0.0.1';
});
return sel[0];
}
function getAppserver() {
const len = config.get('trunks.appserver').length;
return config.get('trunks.appserver')[ idx++ % len];
}
function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp) {
const from = req.getParsedHeader('from');
const common = {'call-id': req.get('Call-ID'), 'from-tag': from.params.tag};
const rtpCharacteristics = config.get('transcoding.rtpCharacteristics');
const srtpCharacteristics = config.get('transcoding.srtpCharacteristics');
return {
common,
offer: Object.assign({'sdp': req.body, 'replace': ['origin', 'session-connection']}, common,
dstIsUsingSrtp ? srtpCharacteristics : rtpCharacteristics),
answer: Object.assign({}, common, srcIsUsingSrtp ? srtpCharacteristics : rtpCharacteristics)
};
}
module.exports = {
fromInboundTrunk,
isWSS,
getAppserver,
makeRtpEngineOpts,
selectHostPort
};

4741
package-lock.json generated Normal file
View File

File diff suppressed because it is too large Load Diff

48
package.json Normal file
View File

@@ -0,0 +1,48 @@
{
"name": "sbc-outbound",
"version": "0.0.1",
"main": "app.js",
"engines": {
"node": ">= 8.10.0"
},
"keywords": [
"sip",
"drachtio"
],
"repository": {
"type": "git",
"url": "git+https://github.com/jambonz/sbc-outbound.git"
},
"author": "Dave Horton",
"license": "MIT",
"bugs": {
"url": "https://github.com/jambonz/sbc-outbound/issues"
},
"homepage": "https://github.com/jambonz/sbc-outbound#readme",
"description": "jambonz session border controller application for outbound calls",
"scripts": {
"start": "node app",
"test": "NODE_ENV=test node test/ | ./node_modules/.bin/tap-spec",
"coverage": "./node_modules/.bin/nyc --reporter html --report-dir ./coverage npm run test",
"jslint": "eslint app.js lib"
},
"dependencies": {
"config": "^3.2.2",
"debug": "^4.1.1",
"drachtio-fn-b2b-sugar": "^0.0.12",
"drachtio-srf": "^4.4.14",
"eslint": "^6.2.2",
"eslint-plugin-promise": "^4.2.1",
"jambonz-mw-registrar": "0.0.3s",
"pino": "^5.13.2",
"rtpengine-client": "^0.0.8"
},
"devDependencies": {
"blue-tape": "^1.0.0",
"clear-module": "^4.0.0",
"nyc": "^14.1.1",
"tap": "^14.6.1",
"tap-dot": "^2.0.0",
"tap-spec": "^5.0.0"
}
}

60
test/docker-compose-testbed.yaml Normal file
View File

@@ -0,0 +1,60 @@
version: '3'
networks:
sbc-outbound:
driver: bridge
ipam:
config:
- subnet: 172.39.0.0/16
services:
sbc:
image: drachtio/drachtio-server:latest
command: drachtio --contact "sip:*;transport=udp" --loglevel debug --sofia-loglevel 9
ports:
- "9060:9022/tcp"
networks:
sbc-outbound:
ipv4_address: 172.39.0.10
redis:
image: redis:5-alpine
ports:
- "16379:6379/tcp"
networks:
sbc-outbound:
ipv4_address: 172.39.0.11
rtpengine:
image: drachtio/rtpengine:latest
ports:
- "12222:22222/udp"
networks:
sbc-outbound:
ipv4_address: 172.39.0.12
registrar:
image: jambonz/sbc-registrar:latest
command: npm start
env_file: registrar.env
networks:
sbc-outbound:
ipv4_address: 172.39.0.13
auth-server:
image: jambonz/customer-auth-server:latest
command: npm start
env_file: http-auth.env
networks:
sbc-outbound:
ipv4_address: 172.39.0.14
carrier:
image: drachtio/sipp:latest
command: sipp -sf /tmp/uas.xml
volumes:
- ./scenarios:/tmp
tty: true
networks:
sbc-outbound:
ipv4_address: 172.39.0.20

12
test/docker_start.js Normal file
View File

@@ -0,0 +1,12 @@
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
View File

@@ -0,0 +1,12 @@
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() ;
});

2
test/http-auth.env Normal file
View File

@@ -0,0 +1,2 @@
# 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"}]}}

3
test/index.js Normal file
View File

@@ -0,0 +1,3 @@
require('./docker_start');
require('./sip-tests');
require('./docker_stop');

3
test/registrar.env Normal file
View File

@@ -0,0 +1,3 @@
# tip: to minify json into a single string as below, see https://codebeautify.org/jsonminifier
NODE_CONFIG={"drachtio":{"host":"sbc","port":9022,"secret":"cymru"},"redis":{"port":6379,"host":"172.39.0.11","prefix":"jb"},"authCallback":{"uri":"http://auth-server:4000/auth","auth":{"username":"foo","password":"bar"}},"logging":{"level":"info"}}
DEBUG=jambonz:*

3
test/scenarios/bad_password.csv Normal file
View File

@@ -0,0 +1,3 @@
SEQUENTIAL
# user, domain, authentication
jane;jambonz.org;[authentication username=jane password=8765]
1 SEQUENTIAL
2 # user, domain, authentication
3 jane;jambonz.org;[authentication username=jane password=8765]

3
test/scenarios/good_user.csv Normal file
View File

@@ -0,0 +1,3 @@
SEQUENTIAL
# user, domain, authentication
john;jambonz.org;[authentication username=john password=1234]
1 SEQUENTIAL
2 # user, domain, authentication
3 john;jambonz.org;[authentication username=john password=1234]

BIN
test/scenarios/pcap/g711a.pcap Normal file
View File

Binary file not shown.

132
test/scenarios/uac-device-invalid-password.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=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>

132
test/scenarios/uac-device-unknown-realm.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]@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="401" auth="true">
</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>
<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: 2 INVITE
[authentication username=john 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]@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: 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>

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-outbound-user-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:john@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: <sip:john@jambonz.org>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-outbound-user-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:john@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: <sip:john@jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Max-Forwards: 70
Subject: uac-outbound-user-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:john@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: <sip:john@jambonz.org ]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Subject: uac-outbound-user-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>

116
test/scenarios/uac-pcap-carrier-success.xml Normal file
View File

@@ -0,0 +1,116 @@
<?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:+16173333456@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: <sip:+16173333456@jambonz.org>
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>
<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:sip:+16173333456@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: <sip:sip:+16173333456@jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
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="2000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:sip:+16173333456@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: <sip:sip:+16173333456@jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Max-Forwards: 70
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>

77
test/scenarios/uac-pcap-device-404.xml Normal file
View File

@@ -0,0 +1,77 @@
<?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:jane@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: <sip:jane@jambonz.org>
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="404" 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:sip:jane@jambonz.org SIP/2.0
[last_Via]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:sip:jane@jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Subject: uac-pcap-device-success
Content-Length: 0
]]>
</send>
</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>

116
test/scenarios/uac-pcap-device-success.xml Normal file
View File

@@ -0,0 +1,116 @@
<?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:john@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: <sip:john@jambonz.org>
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="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:sip:john@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: <sip:sip:john@jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 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="2000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:sip:john@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: <sip:sip:john@jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 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
View File

@@ -0,0 +1,45 @@
<?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>

64
test/scenarios/uac-register-auth-success.xml Normal file
View File

@@ -0,0 +1,64 @@
<?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">
<send retrans="500">
<![CDATA[
REGISTER sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
From: "sipp" <sip:[field0]@[field1]>;tag=[call_number]
To: "sipp" <sip:[field0]@[field1]>
Call-ID: reg///[call_id]
CSeq: 7 REGISTER
Contact: <sip:sipp@[local_ip]:[local_port]>
Expires: 3600
Content-Length: 0
User-Agent: SIPp
]]>
</send>
<recv response="401" auth="true" rtd="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
From: "sipp" <sip:[field0]@[field1]>;tag=[call_number]
To: "sipp" <sip:[field0]@[field1]>
Call-ID: reg///[call_id]
CSeq: 8 REGISTER
Contact: <sip:sipp@[local_ip]:[local_port]>
Expires: 3600
Content-Length: 0
User-Agent: SIPp
[field2]
]]>
</send>
<recv response="200">
</recv>
</scenario>

45
test/scenarios/uac-register-expect-480.xml Normal file
View File

@@ -0,0 +1,45 @@
<?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
View File

@@ -0,0 +1,45 @@
<?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>

71
test/scenarios/uac-unregister-auth-success.xml Normal file
View File

@@ -0,0 +1,71 @@
<?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">
<send retrans="500">
<![CDATA[
REGISTER sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
From: "sipp" <sip:[field0]@[field1]>;tag=[call_number]
To: "sipp" <sip:[field0]@[field1]>
Call-ID: reg///[call_id]
CSeq: 7 REGISTER
Contact: <sip:sipp@[local_ip]:[local_port]>
Expires: 0
Content-Length: 0
User-Agent: SIPp
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="401" auth="true" rtd="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
From: "sipp" <sip:[field0]@[field1]>;tag=[call_number]
To: "sipp" <sip:[field0]@[field1]>
Call-ID: reg///[call_id]
CSeq: 8 REGISTER
Contact: <sip:sipp@[local_ip]:[local_port]>
Expires: 0
Content-Length: 0
User-Agent: SIPp
[field2]
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="200">
</recv>
<ResponseTimeRepartition value="10, 20"/>
<CallLengthRepartition value="10"/>
</scenario>

118
test/scenarios/uac.xml Normal file
View File

@@ -0,0 +1,118 @@
<?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>

115
test/scenarios/uas-simple.xml Normal file
View File

@@ -0,0 +1,115 @@
<?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">
</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:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
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"
optional="true"
rtd="true"
crlf="true">
</recv>
<recv request="BYE">
</recv>
<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>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<timewait milliseconds="4000"/>
<!-- 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
test/scenarios/uas.xml Normal file
View File

@@ -0,0 +1,130 @@
<?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>

40
test/sip-tests.js Normal file
View File

@@ -0,0 +1,40 @@
const test = require('blue-tape');
const { output, sippUac } = require('./sipp')('test_sbc-outbound');
const debug = require('debug')('jambonz:sbc-outbound');
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('sbc-outbound tests', async(t) => {
const {srf} = require('../app');
try {
await connect(srf);
await sippUac('uac-register-auth-success.xml', {ip: '172.39.0.31', data_file: 'good_user.csv'});
t.pass('sip user/device registered over udp');
const p = sippUac('uas.xml', {ip: '172.39.0.31'});
await sippUac('uac-pcap-device-success.xml');
await p;
t.pass('successfully completed outbound call to the registered user/devices');
await sippUac('uac-pcap-device-404.xml');
t.pass('return 404 to outbound attempt to unregisteted user/device');
await sippUac('uac-pcap-carrier-success.xml');
t.pass('successfully completed outbound call to configured sip trunk');
srf.disconnect();
} catch (err) {
console.error(err);
srf.disconnect();
t.error(err);
}
});

75
test/sipp.js Normal file
View File

@@ -0,0 +1,75 @@
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, opts) => {
opts = opts || {};
const cmd = 'docker';
let args = [
'run', '-ti', '--rm', '--net', `${network}`
]
.concat(opts.ip ? ['--ip', opts.ip] : [])
.concat([
'-v', `${__dirname}/scenarios:/tmp/scenarios`,
'drachtio/sipp', 'sipp'
])
.concat(opts.remote_host ? opts.remote_host : [])
.concat(opts.data_file ? ['-inf', `/tmp/scenarios/${opts.data_file}`] : [])
.concat([
'-sf', `/tmp/scenarios/${file}`,
'-m', '1',
'-sleep', '100ms',
'-nostdin',
'-cid_str', `%u-%p@%s-${idx++}`,
'sbc'
]);
//console.log(`args: ${args}`);
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());
});
});
};