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- Herausgeber: Packt Publishing
- Kategorie: Wissenschaft und neue Technologien
- Sprache: Englisch
Build high-speed and highly scalable telephony systems using OpenSIPS
About This Book
- Install and configure OpenSIPS to authenticate, route, bill, and monitor VoIP calls
- Gain a competitive edge using the most scalable VoIP technology
- Discover the latest features of OpenSIPS with practical examples and case studies
Who This Book Is For
If you want to understand how to build a SIP provider from scratch using OpenSIPS, then this book is ideal for you. It is beneficial for VoIP providers, large enterprises, and universities. This book will also help readers who were using OpenSER but are now confused with the new OpenSIPS.
Telephony and Linux experience will be helpful to get the most out of this book but is not essential. Prior knowledge of OpenSIPS is not assumed.
What You Will Learn
- Learn to prepare and configure a Linux system for OpenSIPS
- Familiarise yourself with the installation and configuration of OpenSIPS
- Understand how to set a domain and create users/extensions
- Configure SIP endpoints and make calls between them
- Make calls to and from the PSTN and create access control lists to authorize calls
- Install a graphical user interface to simplify the task of provisioning user and system information
- Implement an effective billing system with OpenSIPS
- Monitor and troubleshoot OpenSIPS to keep it running smoothly
In Detail
OpenSIPS is a multifunctional, multipurpose signalling SIP server. SIP (Session Initiation Protocol) is nowadays the most important VoIP protocol and OpenSIPS is the open source leader in VoIP platforms based on SIP. OpenSIPS is used to set up SIP Proxy servers. The purpose of these servers is to receive, examine, and classify SIP requests. The whole telecommunication industry is changing to an IP environment, and telephony as we know it today will completely change in less than ten years. SIP is the protocol leading this disruptive revolution and it is one of the main protocols on next generation networks. While a VoIP provider is not the only kind of SIP infrastructure created using OpenSIPS, it is certainly one of the most difficult to implement.
This book will give you a competitive edge by helping you to create a SIP infrastructure capable of handling tens of thousands of subscribers.
Starting with an introduction to SIP and OpenSIPS, you will begin by installing and configuring OpenSIPS. You will be introduced to OpenSIPS Scripting language and OpenSIPS Routing concepts, followed by comprehensive coverage of Subscriber Management. Next, you will learn to install, configure, and customize the OpenSIPS control panel and explore dialplans and routing. You will discover how to manage the dialog module, accounting, NATTraversal, and other new SIP services. The final chapters of the book are dedicated to troubleshooting tools, SIP security, and advanced scenarios including TCP/TLS support, load balancing, asynchronous processing, and more.
A fictional VoIP provider is used to explain OpenSIPS and by the end of the book, you will have a simple but complete system to run a VoIP provider.
Style and approach
This book is a step-by-step guide based on the example of a VoIP provider. You will start with OpenSIPS installation and gradually, your knowledge depth will increase.
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Table of Contents
Building Telephony Systems with OpenSIPS Second Edition
Building Telephony Systems with OpenSIPS Second Edition
Copyright © 2016 Packt Publishing
All rights reserved. No part of this book may be reproduced, stored in a retrieval system, or transmitted in any form or by any means, without the prior written permission of the publisher, except in the case of brief quotations embedded in critical articles or reviews.
Every effort has been made in the preparation of this book to ensure the accuracy of the information presented. However, the information contained in this book is sold without warranty, either express or implied. Neither the authors, nor Packt Publishing, and its dealers and distributors will be held liable for any damages caused or alleged to be caused directly or indirectly by this book.
Packt Publishing has endeavored to provide trademark information about all of the companies and products mentioned in this book by the appropriate use of capitals. However, Packt Publishing cannot guarantee the accuracy of this information.
First published: January 2010
Second edition: January 2016
Production reference: 1250116
Published by Packt Publishing Ltd.
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ISBN 978-1-78528-061-0
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Credits
Authors
Flavio E. Goncalves
Bogdan-Andrei Iancu
Reviewers
Saúl Ibarra Corretgé
Vyacheslav Kobzar
Mfawa Alfred Onen
Ali Pey
Commissioning Editor
Neil Alexander
Acquisition Editor
Kevin Colaco
Content Development Editor
Amey Varangaonkar
Technical Editor
Pranil Pathare
Copy Editor
Tasneem Fatehi
Project Coordinator
Suzanne Coutinho
Proofreader
Safis Editing
Indexer
Monica Ajmera Mehta
Graphics
Disha Haria
Production Coordinator
Conidon Miranda
Cover Work
Conidon Miranda
About the Authors
Flavio E. Goncalves was born in 1966 in Brazil. Having a strong interest in computers, he got his first personal computer in 1983, and since then, it has been almost an addiction. He received his degree in engineering in 1989 with a focus on computer-aided designing and manufacturing.
He is also the CTO of SipPulse Routing and Billing Solutions in Brazil—a company dedicated to the implementing of small-to-medium telephone companies, VoIP providers, and large-scale new generation telephony systems. Since 1993, he has participated in a series of certification programs and been certificated as Novell MCNE/MCNI, Microsoft MCSE/MCT, Cisco CCSP/CCNP/CCDP, Asterisk dCAP, and some others.
He started writing about open source software because he thinks that the way certification programs have worked is very good for learners. Some books are written by strictly technical people who sometimes do not have a clear idea on how people learn. He tried to use his 15 years of experience as an instructor to help people learn about the open source telephony software. Together with Bogdan, he created the OpenSIPS boot camp followed by the e-learning program, OpenSIPS eBootcamp.
His experience with networks, protocol analyzers, and IP telephony combined with his teaching experience gave him an edge to write this book. This is the fourth book written by him. The first one was Configuration Guide for Asterisk PBX, by BookSurge Publishing, the second was Building Telephony Systems with OpenSER, by Packt Publishing, and the third was Building Telepopny Systems With OpenSIPS 1.6, by Packt Publishing.
As the CTO of SipPulse, Flavio balances his time between family, work, and fun. He is the father of two children and lives in Florianopolis, Brazil—one of the most beautiful places in the world. He dedicates his free time to water sports such as surfing and sailing.
Bogdan-Andrei Iancu entered the SIP world in 2001, right after graduating in computer science from the Politehnica University of Bucharest, Romania. He started as a researcher at the FOKUS Fraunhofer Institute, Berlin, Germany. For almost four years, Bogdan accumulated a quick understanding and experience of VoIP/SIP, being involved in research and industry projects and following the evolution of the VoIP world closely.
In 2005, he started his own company, Voice System. The company entered the open source software market by launching the OpenSER/OpenSIPS project—a free GPL-SIP proxy implementation. As the CEO of Voice System, Bogdan pushes the company in two directions: developing and supporting.
The OpenSIPS public project (Voice System being the major contributor and sponsor of the project) creates professional solutions and platforms (OpenSIPS-based) for the industry. In other words, Bogdan's interest was to create knowledge (through the work with the project) and to provide the knowledge where needed (embedded in commercial products or raw format as consultancy services). In the effort of sharing the knowledge of the SIP/OpenSIPS project, he started to run the OpenSIPS Bootcamp in 2008 together with Flavio E. Goncalves, which is intensive training dedicated to people who want to learn and get hands-on experience on OpenSIPS from experienced people. Bogdan's main concern is to research and develop new technologies or software for SIP-based VoIP (this is the reason for his strong involvement with the OpenSIPS project) and pack all these cutting-edge technologies as professional solutions for the industry.
About the Reviewers
Saúl Ibarra Corretgé started working in the VoIP industry over a decade ago. He has worked in many different areas and projects, from development and configuration to deployment.
In 2006, when OpenSER 1.0.0 (the project where OpenSIPS was forked from) was released, Saúl began to experiment with it. Several years later, he started using it heavily and contributing with code until he became an OpenSIPS core team member in 2010. His contributions to the project have been diverse but mainly focused on improving the presence part.
He also maintains several projects on GitHub (https://github.com/saghul) and you can contact him through his website (http://bettercallsaghul.com) or on Twitter (@saghul).
When not in front of the computer, he likes to travel around the world.
Vyacheslav Kobzar is the chief of software development at Modulis.ca Inc. He graduated from Donetsk State Technical University in 2006, where he was studying software development. Right after graduation, he started to work as a freelance developer on different projects, mostly web development. Since 2008, he started to work remotely in the Canadian company, Modulis.ca Inc. He moved to Canada in 2009 where he continued working at Modulis.ca Inc as a developer on multiple web projects.
He started to work on VoIP in 2011, mostly with Asterisk. He has been working on AGI and AMI modules for different VoIP projects. He was certificated with Asterisk dCAP in 2012. In 2014, Vyacheslav participated in the OpenSIPS eBootcamp session. He has been an OpenSIPS's Foundation member since 2014.
In 2013, he participated in the designing and developing of the Modulis VoIP start-up project, which was later successfully deployed in multiple companies and organizations in Quebec. OpenSIPS is the core part of the project along with other VoIP technologies and protocols (UNIStim, Skinny, and others).
Being a Linux user for almost 10 years, Vyacheslav contributes to different open source projects on GitHub and also works on his own.
I'd like to thank OpenSIPS developers and contributors for this amazing project. I would also like to thank the Modulis team for sharing their knowledge and ideas and always being open for new challenges. Finally, I would like to thank my wife, Anna, for her support and patience.
Mfawa Alfred Onen is a system administrator with more than 6 years' experience in the field of UNIX/Linux system administration. He studied electrical and electronics engineering in his bachelor of engineering undergraduate program and has continued to venture into the area of telecommunications with a postgraduate certificate from Birmingham City University, UK. He currently resides in Nigeria and has worked with both private and education sectors, including numerous consulting jobs for clients at home and abroad.
Being a software developer and having an operations background, he is heavily involved with cloud computing (DevOps) using open source software such as OpenStack, OpenShift, Docker, Asterisk, OpenSIPS, and FreeSWITCH, to name a few. He also helps to manage a Google Developer Group (GDG Bingham University), where software developers and technology enthusiasts come to learn Google developer tools and services in the form of Developer Festivals (DevFest), Hackathons, and Code labs.
When Mfawa is not busy with technology, he is an avid gamer (Call of Duty, NFS, and Forza) and a blogger at http://www.maomuffy.com/ with much content on OpenShift, OpenStack, RADIUSDesk, Linux/UNIX system administration, and so on.
My special thanks goes to my family (Professor Alfred Ikpi Onen, Mrs. Jummai Alfred Onen, Dr. Eno Alfred Onen, Williams Alfred Onen, and Ikpi Alfred Onen Jnr.), friends (Aderogba Otunla, Alhamdu Bello, Suzanne Coutinho, and others) and well-wishers.
Ali Pey is a senior software engineer architect with more than 23 years experience in telephony, networking, and VoIP. He has an electronics engineering degree with a focus on telecommunication and software design. He has worked for companies such as Nortel, TalkSwitch, and j2 Global, and has been developing VoIP solutions since the start of the technology. He has developed software for proxy servers, registrar servers/clients, user agents, and other VoIP components in both SIP and H.323 protocols. Currently, Ali is an independent consultant and has successfully used OpenSIPS and other open source applications such as Asterisk and FreeSWITCH to provide global telephony cloud solutions.
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Preface
This book will be your companion when working with OpenSIPS using a case study for an Internet Telephony Service Provider (ITSP). With the help of this book, you should be able to build a system that is able to authenticate, route, bill, and monitor VoIP calls. Topics and advanced scenarios such as TCP/TLS support, load balancing, asynchronous processing, and more are discussed in depth in this book. You will create dynamic dialplans, route calls using advanced routing, integrate OpenSIPS with a media server, account calls and generate CDRs, provision the system using a Web GUI, and use tools to monitor and check the health of your server. You will also learn some advanced topics such as support for TLS/TCP and the newest technology called asynchronous callbacks.
By the end of this book, you should be able to build a system that is able to authenticate, route, bill, and monitor VoIP calls. Whenever you are thinking big on telephony, OpenSIPS is your savior and this book is your friend!
What this book covers
Chapter 1, Introduction to SIP, introduces you to the SIP server. You will see how to recognize a SIP request and reply according to RFC 3261, identify the mandatory SIP headers, and describe the SIP routing process for initial and sequential requests.
Chapter 2, Introducing OpenSIPS, shows you how OpenSIPS is used in the market, the basic architecture of the system, use cases, and the main target market.
Chapter 3, Installing OpenSIPS, shows you how to download the OpenSIPS source and its dependencies, compile and install OpenSIPS with MySQL and Radius support, and configure the Linux system to start OpenSIPS at boot time.
Chapter 4, OpenSIPS Language and Routing Concepts, introduces you to the OpenSIPS scripting language and OpenSIPS routing concepts. After reading this chapter, you should be able to recognize the OpenSIPS script language, describe its mains commands, process initial requests, and drop or route requests.
Chapter 5, Subscriber Management, shows you how to manage subscribers in the system using the subscriber, location, group, and address databases. You will learn how to implement a multidomain system that is able to support multitenant implementations.
Chapter 6, OpenSIPS Control Panel, demonstrates how to install a web GUI to help with the provisioning of users, dialplan, routes, and other information that is required to run OpenSIPS. You will see how to install, use, configure, and customize the OpenSIPS control panel.
Chapter 7, Dialplan and Routing, enables you to integrate OpenSIPS with PSTN through gateways, selecting the best gateway, and failing over automatically if a response code is negative.
Chapter 8, Managing Dialogs, shows you how to activate the dialog module, limit the number of simultaneous calls, disconnect hanged calls, impose a maximum duration time for a call, and implement SIP session timers integrated with the dialog module.
Chapter 9, Accounting, demonstrates how account calls generate a CDR (Call Detail Record), account correctly forwarded calls, prevent calls without BYE, and add extra fields to the CDR.
Chapter 10, SIP NAT Traversal, helps you implement an OpenSIPS solution for clients behind NAT. You will see how to implement OpenSIPS in a data center such as Amazon AWS where all the servers are behind NAT.
Chapter 11, Implementing SIP Services, implements services such as call forward, forward on busy, and forward on no answer in cooperation with a media server and SIP phone.
Chapter 12, Monitoring Tools, enables you to detect performance issues using the built-in statistics. These include protocol issues using SIP trace, database issues using the benchmark module, script issues using the script trace, and software and hardware issues using GDB.
Chapter 13, OpenSIPS Security, shows you how to increase the security of your OpenSIPS installation.
Chapter 14, Advanced Topics with OpenSIPS 2.1, covers some advanced topics that can be important for specific installations. Topics such as asynchronous processing, TCP and TLS support, binary replication, and NoSQL integration for clusters are discussed.
What you need for this book
All you need for this book is a working installation of OpenSIPS on either Linux or Debian. We will go through the installation of OpenSIPS in detail in this book.
Who this book is for
System integrators who need to scale their VoIP projects, universities, and other entities who need to provide large-scale communication systems based on the SIP protocol can make the best use of this book.
Conventions
In this book, you will find a number of text styles that distinguish between different kinds of information. Here are some examples of these styles and an explanation of their meaning.
Code words in text, database table names, folder names, filenames, file extensions, pathnames, dummy URLs, user input, and Twitter handles are shown as follows: "It also contains a parameter called branch that identifies this transaction."
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When we wish to draw your attention to a particular part of a code block, the relevant lines or items are set in bold:
Any command-line input or output is written as follows:
New terms and important words are shown in bold. Words that you see on the screen, for example, in menus or dialog boxes, appear in the text like this: "In the following Select your location screen, choose your location to be used in the installation process."
Note
Warnings or important notes appear in a box like this.
Tip
Tips and tricks appear like this.
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Chapter 1. Introduction to SIP
Before we dive into OpenSIPS, it is very important to understand some important concepts related to Session Initiation Protocol (SIP). In this chapter, we will cover a brief tutorial regarding the concepts used later in this book. By the end of this chapter, we will have covered the following topics:
SIP was standardized by Internet Engineering Task Force (IETF) and is described in several documents known as Request for Comments (RFC). The RFC 3261 describes SIP version 2. SIP is an application layer protocol used to establish, modify, and terminate sessions or multimedia calls. These sessions can be audio and video sessions, e-learning, chatting, or screen sharing sessions. It is similar to Hypertext Transfer Protocol (HTTP) and designed to start, keep, and close interactive communication sessions between users. Nowadays, SIP is the most popular protocol used inInternet Telephony Service Providers (ITSPs), IP PBXs, and voice applications.
The SIP protocol supports five features to establish and close multimedia sessions:
The SIP protocol was designed as a part of a multimedia architecture containing other protocols such as Resource Reservation Protocol (RSVP), Real-Time Protocol (RTP), Real-Time Session Protocol (RTSP), Session Description Protocol (SDP), andSession Announcement Protocol (SAP). However, it does not depend on them to work.
Understanding the SIP architecture
SIP has borrowed many concepts from the HTTP protocol. It is a text-based protocol and uses the same Digest mechanism for authentication. You will also notice similar error messages such as 404 (Not found) and 301 (Redirect). As a protocol developed by the IETF, it uses an addressing scheme similar to Simple Mail Transfer Protocol (SMTP). The SIP address is just like an e-mail address. Another interesting feature used in SIP proxies are aliases; you can have multiple SIP addresses for a single subscriber such as the following:
In the SIP architecture, there are user agents and servers. SIP uses a peer-to-peer distributed model with a signaling server. The signaling server only handles the SIP signaling, while the user agent clients and servers handle signaling and media. This is depicted in the following figure:
In the traditional SIP model, a user agent, usually a SIP phone, will start communicating with its SIP proxy, seen here as the outgoing proxy (or its home proxy) to send the call using a message known as INVITE.
The outgoing proxy will see that the call is directed to an outside domain. According to RFC 3263, it will seek the DNS server for the address of the target domain and resolve the IP address. Then, the outgoing proxy will forward the call to the SIP proxy responsible for DomainB.
The incoming proxy will query its location table for the IP address of agentB if its address was inserted in the location table by a previous registration process. It will forward the call to agentB.
After receiving the SIP message, agentB will have all the information required to establish an RTP session (usually audio) with agentA sending a 200 OK response. Once agentA receives the response from agentB, a two-way media can be established. A BYE request message can terminate the session.
Here, you can see the main components of the SIP architecture. The entire SIP signaling flows through the SIP proxy server. On the other hand, the media is transported by the RTP protocol and flows directly from one endpoint to another. Some of the components will be briefly explained in the sequence.
In the preceding image, you can see the following components:
The proxy, redirect, and registrar servers are usually available physically in the same computer and software.
The SIP registration process
The SIP registration process is shown as follows:
The SIP protocol employs a component called Registrar. It is a server that accepts REGISTER requests and saves the information received in these packets on the location server for their managed domains. The SIP protocol has a discovery capacity; in other words, if a user starts a session with another user, the SIP protocol has to discover an existent host where the user can be reached. The discovery process is done (among others) by a Registrar server that receives the request and finds the location to send it. This is based in a location database maintained by the Registrar server per domain. The Registrar server can accept other types of information, not only the client's IP addresses. It can receive other information such as Call Processing Language (CPL) scripts on the server.
Before a telephone can receive calls, it needs to be registered with the location database. In this database, we will have all the phones associated with their respective IP addresses. In our example, you will see the sip user, [email protected], registered with the IP address, 200.180.1.1.
RFC 3665 defines best practices to implement a minimum set of functionalities for a SIP IP communications network. In the following table, the flows are defined according to RFC 3665 for registration transactions. According to RFC 3665, there are five basic flows associated with the process of registering a user agent.
Message flow
Description
A successful new registration: After sending the Register request, the user agent will be challenged against its credentials. We will see this in detail in Chapter 5, Subscriber Management.
An update of the contact list: As it is not a new registration, the message already contains the Digest and a 401 message won't be sent. To change the contact list, the user agent just needs to send a new register message with the new contact in the CONTACT header field.
A request for the current contact list: In this case, the user agent will send the CONTACT header field empty, indicating that the user wishes to query the server for the current contact list. In the 200 OK message, the SIP server will send the current contact list in the CONTACT header field.
The cancellation of a registration: The user agent now sends the message with an EXPIRES header field of 0 and a CONTACT header field configured as * to apply to all the existing contacts.
Unsuccessful Registration: The UAC sends a REGISTER request and receives a 401 Unauthorized Message in exactly the same way as the successful registration. In the sequence, it produces a hash and tries to authenticate. The server, detecting an invalid password, sends a 401 Unauthorized message again. The process will be repeated for the number of retries configured in the UAC.
Types of SIP servers
There are a few different types of SIP servers. Depending on the application, you can use one or all of them in your solution. OpenSIPS can behave as a proxy, redirect, B2BUA, or Registrar server.
The proxy server
In the SIP proxy mode, all SIP signaling goes through the SIP proxy. This behavior will help in processes such as billing and is, by far, the most common choice. The drawback is the overhead caused by the server in the middle of all the SIP communications during the session establishment. Regardless of the SIP server role, the RTP packets will go directly from one endpoint to another even if the server is working as a SIP proxy.
The redirect server
The SIP proxy can operate in the SIP redirect mode. In this mode, the SIP server is very scalable because it doesn't keep the state of the transactions. Just after the initial INVITE, it replies to the UAC with a 302 Moved Temporarily and is removed from the SIP dialog. In this mode, a SIP proxy, even with very few resources, can forward millions of calls per hour. It is normally used when you need high scalability but don't need to bill the calls.
The B2BUA server
The server can also work as a Back-to-Back User Agent (B2BUA). B2BUAs are normally applied to hide the topology of the network. They are also useful to support buggy clients unable to route SIP requests correctly based on record routing. Many PBX systems such as Asterisk, FreeSwitch, Yate, and others work as B2BUAs.
SIP request messages
There are several types of message requests. SIP is transactional, communicating through requests and replies. The most important types of requests are described in the following table:
Message
Description
RFC
ACK
Acknowledges an INVITE
RFC 3261
BYE
Terminates an existing session
RFC 3261
CANCEL
Cancels a pending registration
RFC 3261
INFO
Provides mid-call signaling information
RFC 2976
INVITE
Session establishment
RFC 3261
MESSAGE
Instant message transport
RFC 3428
NOTIFY
Sends information after subscribing
RFC 3265
PRACK
Acknowledges a provisional response
RFC 3262
PUBLISH
Uploads the status information to the server
RFC 3903
REFER
Asks another UA to act onUniform Resource Identifier (URI)
RFC 3515
REGISTER
Registers the user and updates the location table
RFC 3261
SUBSCRIBE
Establishes a session to receive future updates
RFC 3265
UPDATE
Updates a session state information
RFC 3311
Most of the time, you will use REGISTER, INVITE, ACK, BYE, and CANCEL. Some messages are used for other features. For example, INFO is used forDual-tone Multi-frequency (DTMF) relay and mid-call signaling information. PUBLISH, NOTIFY, and SUBSCRIBE give support to the presence systems. REFER is used for call transfer and MESSAGE for chat applications. Newer requests can appear depending on the protocol standardization process. Responses to these requests are in the text format as in the HTTP protocol. Some of the most important replies are shown as follows:
The SIP dialog flow
Let's examine this message sequence between two user agents as shown in the following figure. You can see several other flows associated with the session establishment in RFC 3665:
The messages are labeled in sequence. In this example, userA uses an IP phone to call another IP phone over the network. To complete the call, two SIP proxies are used.
The userA calls userB using its SIP identity called the SIP URI. The URI is similar to an e-mail address, such as <sip:[email protected]>. A secure SIP URI can be used too, such as <sips:[email protected]>. A call made using sips: (Secure SIP) will use a secure transport, Transport Layer Security (TLS), between the caller and callee.
The transaction starts with userA sending an INVITE request addressed to userB. The INVITE request contains a certain number of header fields. Header fields are named attributes that provide additional information about the message and include a unique identifier, the destination, and information about the session.
The first line of the message contains the method name and request URI. The following lines contain a list of header fields. This example contains the minimum set required. The header fields have been described as follows:
Session details such as the media type and codec are not described in SIP. Instead, it uses theSession Description Protocol (SDP) (RFC 2327). This SDP message is carried by the SIP message, similar to an e-mail attachment.
The phone does not know the location of userB or the server responsible for domainB. Thus, it sends the INVITE request to the server responsible for the domain, sipA. This address is configured in the phone of userA or can be discovered by DNS. The server sipA.com is also known as the SIP proxy for the domain sipA.com.
The sequence is as follows:
The first line contains the response code and a description (OK). The following lines contain the header fields. The Via, To, From, Call-ID, and CSeq fields are copied from the INVITE request and the To tag is attached. There are three Via fields: one added by userA, another by ProxyA, and finally, ProxyB. The SIP phone of userB adds a tag parameter for the To and From headers and will include this tag on all the future requests and responses for this call.
The Contact header field contains the URI by which userB can be contacted directly in its own IP phone.
The Content-Type and Content-Length header fields give some information about the SDP header. The SDP header contains media-related parameters used to establish the RTP session.
After answering the call, the following occurs:
In some cases, it can be important for the proxies to stay in the middle of the signaling to see all the messages between the endpoints during the whole session. If the proxy wants to stay in the path after the initial INVITE request, it has to add the Record-Route header field to the request. This information will be received by userB's phone and will send back the message through the proxies with the Record-Route header field included too. Record-routing is used in most scenarios. Without record-routing, it is not possible to account the calls and there is no control of the SIP dialog in the proxy.
The REGISTER request is the way that ProxyB learns the location of userB. When the phone initializes or in regular time intervals, the SIP phoneB sends a REGISTER request to a server on domain sipB known as SIP Registrar. The REGISTER messages associate a URI (<[email protected]>) to an IP address. This binding is stored in a database in the Location server. Usually the Registrar, Location, and Proxy servers are in the same computer and use the same software such as OpenSIPS. A URI can only be registered by a single device at a certain time.
SIP transactions and dialogs
It is important to understand the
