Real-Time Communications Quick Start Guide
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Real-Time Communications Quick Start Guide

Daniel Pocock

Table of Contents

Preface

1. Introduction

Federation

Independent and decentralized alternatives to federation

Private networks
Decentralized networks
Conclusion

Choosing between SIP and XMPP

Choice of operating system

Using a ready-to-run or turn-key solution
Using a generic GNU/Linux distribution

Use latest software versions

Using IPv6

Example network used in the documentation

2. Architecture overview

The big picture

TLS is essential
All SIP connectivity through a SIP proxy

SIP federation between two autonomous sites

Routing calls within a site

WebRTC peer-to-peer calling

WebRTC calling to call centers

3. User Experience

First time setup and provisioning

Dialing

Usernames or phone numbers?
Dial plans
Dialing Internet addresses

4. Optimizing Connectivity

Codec selection

Recommendations

Media stream encryption compatibility

Supporting multiple schemes
Recommendations for maximizing connectivity
Recommendations for security

Use ICE and a TURN server

Use the TLS transport for SIP signalling

Getting through firewalls

5. DNS setup

Using non-standard ports
Sample DNS zone file
Testing the DNS settings

6. Firewall rules

Overview of firewall ports
NAT considerations
Setup with iptables on Linux

7. User and credential storage

Credentials

Personal account names or extension numbers
Password encryption
HA1 in detail

Databases

RADIUS
LDAP
SQL databases
Product-specific file formats

Conclusion

8. Server setup

9. TLS certificate creation

Certificate Common Name
Install the OpenSSL utility
Install the Let's Encrypt certbot utility
Install a TLS certificate using Let's Encrypt (certbot)
Install a TLS certificate manually

10. ICE/STUN/TURN server installation

Choosing a TURN server

reTurnServer from reSIProcate

Installation
Configuration
Provisioning users

Testing the TURN server

11. SIP proxy server installation

Choose your SIP proxy

repro SIP proxy

Package installation
Configuration
Testing with s_client
Login to web administration
User management
Adding a user
Adding routes for numeric dialing

Kamailio SIP proxy

Package installation
Configuration

12. XMPP (Jabber) server installation

Choosing an XMPP server

Prosody XMPP server

Package installation
Configuration
User management
Further reading

ejabberd XMPP server

Package installation
Configuration

jabberd2 XMPP server

Package installation
Configuration
Further reading

13. WebRTC

Technical overview

Media streaming capabilities
Signalling protocols
User privacy and security
Authentication

Practical WebRTC deployment

WebRTC clients and firewalls
JsSIP and JSCommunicator
Content Management Systems and other frameworks
Troubleshooting

14. Client devices and softphones

The Firefox Telify plugin
Mozilla Thunderbird and GNOME Evolution address books
Using sipdialer
Using Asterisk or FreeSWITCH

15. Adding ENUM to DNS

How ENUM works

Consuming ENUM data

Publishing ENUM data

Public ENUM
Private and third-party ENUM suffixes
Dynamic ENUM from LDAP with dlz-ldap-enum

16. Troubleshooting

Common problems and solutions

Google Talk/Hangouts users not receiving XMPP chat messages
Audio and video quality issues

Techniques

Monitoring tools
Check the logs
Check the web interface
Operating system utilities
Packet sniffers
Debugging mode
WebRTC and WebSockets

17. PBX Setup

The all-in-one myth

Choosing between Asterisk and FreeSWITCH

Official packages
Contributing patches
Licensing
Community
Scalabiltiy and code quality

Using Asterisk with the repro SIP proxy

18. PSTN connectivity

Methods of PSTN connectivity
ingress call handling
egress call handling
Emergency calls

19. Frequently Asked Questions

20. Community support

Mailing lists

Major announcements
Strategy and advocacy
Collaboration between operators and service providers
Server support
Softphones

Popular blogs and news sites

A. Building reSIProcate packages on Debian/Ubuntu

B. Building reSIProcate RPMs on RHEL and CentOS

C. Configuring Nagios to monitor SIP, XMPP and TURN

Nagios plugins
Nagios service checks

D. Mitigating VoIP fraud risk

Legal insurance
Trade body membership
Set a credit limit
Use a different phone company for inbound numbers

List of Figures

2.1. Overview
2.2. SIP federation between two sites
2.3. Four stages of call routing
2.4. WebRTC basic peer-to-peer
2.5. WebRTC from customer web browser to call center
4.1. Metcalfe's law
11.1. repro web administration: adding a domain
11.2. repro web administration: adding a user
11.3. repro web administration: listing users
11.4. repro web administration: adding a route
11.5. repro web administration: listing routes
11.6. repro web administration: routing test
13.1. DruCall/JSCommunicator/JsSIP software stack

List of Tables

4.1. Common codecs
5.1. DNS records for the example
5.2. Protocols using port 443
6.1. Firewall rules summary
6.2. Firewall rules summary (IPv6)
11.1. Comparison of SIP proxy servers
11.2. TLS client verification modes
17.1. repro route configuration

List of Examples

2.1. Splitting Asterisk extensions.conf
5.1. ISC Bind zone file entries
5.2. Inspecting DNS entries with dig
6.1. Firewall setup with iptables
7.1. Computing HA1
7.2. OpenLDAP ACL for protecting ha1Password
7.3. SQL table for repro users
7.4. SQL view presenting Asterisk users to repro
7.5. Install PostgreSQL on Debian or Ubuntu
7.6. Configure PostgreSQL and load schema
8.1. Adding IP addresses in /etc/network/interfaces
9.1. Installing openssl on Debian/Ubuntu
9.2. Installing openssl on Fedora/RHEL/CentOS
9.3. Installing certbot on Debian/Ubuntu
9.4. Installing certbot on Fedora/RHEL/CentOS
9.5. PKI directories (Debian/Ubuntu)
9.6. PKI directories (Fedora/RHEL/CentOS)
9.7. Creating RSA key pair and CSR
9.8. Installing the certificate
10.1. Installing reTurnServer on Debian/Ubuntu
10.2. Install reTurnServer on Fedora/RHEL/CentOS
10.3. reTurnServer.config entries
10.4. Restarting the reTurnServer daemon (systemd)
10.5. Using netstat to verify reTurnServer is running
10.6. crontab entry for psql-user-extract
10.7. Sample /etc/reTurn/psql-user-extract.config
10.8. Installing the stun client utility
10.9. Using the stun client utility
11.1. Installing repro on Debian/Ubuntu
11.2. Install repro on Fedora/RHEL/CentOS
11.3. Sample values for repro.config
11.4. Using PostgreSQL
11.5. Using MySQL
11.6. Using htdigest to set admin user password
11.7. Restarting the repro daemon (systemd)
11.8. Using s_client to test SIP ports (Debian/Ubuntu)
11.9. Using s_client to test SIP ports (Fedora/RHEL/CentOS)
11.10. Installing kamailio on Debian/Ubuntu
11.11. Install kamailio on Fedora/RHEL/CentOS
11.12. Restarting the kamailio daemon (systemd)
12.1. Installing Prosody on Debian/Ubuntu
12.2. Install Prosody on Fedora/RHEL/CentOS
12.3. Domain configuration file (Let's Encrypt / certbot)
12.4. Domain configuration file (manual)
12.5. Restarting the prosody daemon (systemd)
12.6. Using prosodyctl to add a user
12.7. prosody.cfg.lua settings for mod_auth_ldap
12.8. Installing ejabberd on Debian/Ubuntu
12.9. Install ejabberd on Fedora/RHEL/CentOS
12.10. ejabberd interface example
12.11. Installing jabberd2 on Debian/Ubuntu
12.12. Install jabberd2 on Fedora/RHEL/CentOS
12.13. jabberd2 c2s.xml example
12.14. jabberd2 sm.xml example
13.1. repro.config settings for cookie and URL parameter authentication
15.1. Using dig to perform ENUM queries
15.2. Installing dlz-ldap-enum on Debian/Ubuntu
15.3. Install dlz-ldap-enum on Fedora/RHEL/CentOS
15.4. Sample dlz_ldap_enum.conf
15.5. Additions to named.conf for Debian/Ubuntu
15.6. Additions to named.conf for Fedora/RHEL/CentOS
17.1. Asterisk sip.conf
17.2. Asterisk extensions.conf
18.1. Asterisk extensions.conf for specifying caller ID
A.1. Installing the debuild command
A.2. Installing the compiler and dependencies
A.3. Running the debuild command
A.4. Running the debuild command using code from Git
B.1. Installing the rpmbuild command
B.2. Installing the compiler and dependencies
B.3. Creating the rpmbuild directories
B.4. Running the rpmbuild command
C.1. Sample /etc/nagios-plugins/config/stun.cfg
C.2. Sample /etc/nagios-plugins/config/sip.cfg
C.3. Sample /etc/nagios-plugins/config/xmpp.cfg
C.4. Sample service check for STUN/TURN
C.5. Sample service check for SIP over TLS
C.6. Sample service check for SIP over TLS (port 443)
C.7. Sample service check for XMPP

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