Resources For You

  1. 5 Essential Marketing Strategies for VoIP Businesses

    5 Essential Marketing Strategies for VoIP Businesses

  2. 5 Technologies Set to Revolutionise Webphones

    5 Technologies Set to Revolutionise Webphones

  3. 5 Unique Types of VoIP Gateways Explained!

    5 Unique Types of VoIP Gateways Explained!

  4. 5 Ways a Cloud PBX System Benefits Remote Work

    5 Ways a Cloud PBX System Benefits Remote Work

  5. 5 Ways SBCs Facilitate Unified Communications as a Service

    5 Ways SBCs Facilitate Unified Communications as a Service

  6. 5 Ways to Optimise ASR To Grow Profitability

    5 Ways to Optimise ASR To Grow Profitability

  7. 7 Additional Important Components of a VoIP Carrier Network Explained

    7 Additional Important Components of a VoIP Carrier Network Explained

  8. 7 Important Factors to Consider When Implementing LCR

    7 Important Factors to Consider When Implementing LCR

  9. 7 Ways to Optimize AHT

    7 Ways to Optimize AHT

  10. 9 Key Functions of an SBC Explained

    9 Key Functions of an SBC Explained

  11. 10 Factors to Consider While Choosing a Webphone

    10 Factors to Consider While Choosing a Webphone

  12. 10 Important Components of a VoIP Carrier Network Explained

    10 Important Components of a VoIP Carrier Network Explained

  13. 10-Point Security Checklist for VoIP Carriers

    10-Point Security Checklist for VoIP Carriers

  14. 10 Tips For Effective Implementation of LCR

    10 Tips For Effective Implementation of LCR

  15. 10 Webphone Features that Benefit Your Business

    10 Webphone Features that Benefit Your Business

  16. An Out of the Box Telecoms Network

    An Out of the Box Telecoms Network

  17. Are Call Centers Still Relevant in 2023?

    Are Call Centers Still Relevant in 2023?

  18. Automated Dialler vs Manual Dialler - Knowing the 7 Key Differences

    Automated Dialler vs Manual Dialler - Knowing the 7 Key Differences

  19. Call Center vs Contact Center - Understanding the Differences

    Call Center vs Contact Center - Understanding the Differences

  20. Choosing SIP over TCP,TLS and UDP in 2022

    Choosing SIP over TCP,TLS and UDP in 2022

  21. Class 4 Softswitch vs Class 5 Softswitch - Understanding the Difference

    Class 4 Softswitch vs Class 5 Softswitch - Understanding the Difference

  22. Combatting Covid-19 with Carrier-Grade Communications Solutions to Help Users Work Remotely

    Combatting Covid-19 with Carrier-Grade Communications Solutions to Help Users Work Remotely

  23. Comprehensive Cloud Softswitch Documentation

    Comprehensive Cloud Softswitch Documentation

  24. ConnexCS expands AnyEdge SIP Load Balancer to India

    ConnexCS expands AnyEdge SIP Load Balancer to India

  25. ConnexCS for Africa

    ConnexCS for Africa

  26. ConnexCS WebPhone SDK Connector

    ConnexCS WebPhone SDK Connector

  27. Discover the Different Types of NAT: An Essential Guide for Network Administrators

    Discover the Different Types of NAT: An Essential Guide for Network Administrators

  28. Discussing the Future and Top 9 Benefits of WebRTC

    Discussing the Future and Top 9 Benefits of WebRTC

  29. DNO And DNC Lists - Everything Carriers Should Know

    DNO And DNC Lists - Everything Carriers Should Know

  30. Email and SMS Alerts

    Email and SMS Alerts

  31. Employers' Guide to Winning at Remote Work

    Employers' Guide to Winning at Remote Work

  32. Exploring the Top 10 Types of Web Phones in 2023!

    Exploring the Top 10 Types of Web Phones in 2023!

  33. False Answer Supervision Detection - The Ultimate Tool for Preventing VoIP Fraud

    False Answer Supervision Detection - The Ultimate Tool for Preventing VoIP Fraud

  34. Far-End NAT Traversal - An In-Depth Guide

    Far-End NAT Traversal - An In-Depth Guide

  35. Feature Releases for June 2024

    Feature Releases for June 2024

  36. From Cost Savings to Mobility - 15 Benefits of Web Phones for Businesses

    From Cost Savings to Mobility - 15 Benefits of Web Phones for Businesses

  37. Get Your FCC Registration Number in 5 Easy Steps!

    Get Your FCC Registration Number in 5 Easy Steps!

  38. How to Build Your API on ConnexCS

    How to Build Your API on ConnexCS

  39. How to Build Your Own Dialer (BYOD) – Part 1

    How to Build Your Own Dialer (BYOD) – Part 1

  40. How to Establish a VoIP Interconnect in 10 Easy Steps

    How to Establish a VoIP Interconnect in 10 Easy Steps

  41. How to Get Operating Company Number (OCN) in 4 Easy Steps

    How to Get Operating Company Number (OCN) in 4 Easy Steps

  42. How to Identify Robocall Scam Traffic - A Comprehensive Guide for Telecom and VoIP Operators

    How to Identify Robocall Scam Traffic - A Comprehensive Guide for Telecom and VoIP Operators

  43. How to Improve CX? Ensure your Call Center Agents are Happy!

    How to Improve CX? Ensure your Call Center Agents are Happy!

  44. How to Prepare for a VoIP Network Security Audit

    How to Prepare for a VoIP Network Security Audit

  45. How to Properly Prepare for Setting up a VoIP Interconnect

    How to Properly Prepare for Setting up a VoIP Interconnect

  46. How to Register for the Robocall Mitigation Database: A step-by-step guide!

    How to Register for the Robocall Mitigation Database: A step-by-step guide!

  47. How to Successfully Implement LCR is 5 Easy Steps

    How to Successfully Implement LCR is 5 Easy Steps

  48. How Using Web Phones Can Benefit These 10 Industries?

    How Using Web Phones Can Benefit These 10 Industries?

  49. Importance of Balancing Cost Minimization and Reliable Call Quality when implementing LCR

    Importance of Balancing Cost Minimization and Reliable Call Quality when implementing LCR

  50. Introducing ConnexCS WebPhone

    Introducing ConnexCS WebPhone

  51. Introducing ConneXML - The Best TwiML Alternative

    Introducing ConneXML - The Best TwiML Alternative

  52. Introducing Smart CLI Select - An Effective Way to Improve your ASR

    Introducing Smart CLI Select - An Effective Way to Improve your ASR

  53. LTE vs VoLTE: Diving Into The Differences

    LTE vs VoLTE: Diving Into The Differences

  54. Operating Company Numbers (OCN) - Understanding Function, Importance and Relevance

    Operating Company Numbers (OCN) - Understanding Function, Importance and Relevance

  55. Populating Our Support Area With Cloud Softswitch Video Guides

    Populating Our Support Area With Cloud Softswitch Video Guides

  56. Predictive Dialler vs Progressive Dialler - Understanding the Differences

    Predictive Dialler vs Progressive Dialler - Understanding the Differences

  57. Preview Dialler vs Power Dialler - Understanding Top 5 Differences

    Preview Dialler vs Power Dialler - Understanding Top 5 Differences

  58. Rate Card Profit Assurance

    Rate Card Profit Assurance

  59. Redundant Redundancies (Backups of backups)

    Redundant Redundancies (Backups of backups)

  60. Revolutionise Your Outbound Calls - 8 Types of VoIP Diallers Explained

    Revolutionise Your Outbound Calls - 8 Types of VoIP Diallers Explained

  61. Scalability – Grow at Speeds That Suit You

    Scalability – Grow at Speeds That Suit You

  62. ScriptForge – Javascript Routing

    ScriptForge – Javascript Routing

  63. Simplifiying our Softswitch Pricing

    Simplifiying our Softswitch Pricing

  64. SIP 101 - The Best Guide of 2022

    SIP 101 - The Best Guide of 2022

  65. The 3CX Supply Chain Attack - Understanding Everything That Happened

    The 3CX Supply Chain Attack - Understanding Everything That Happened

  66. The 5 Best Strategies for Mitigating Robocall Scams

    The 5 Best Strategies for Mitigating Robocall Scams

  67. The Anatomy of Robocall Scams

    The Anatomy of Robocall Scams

  68. The Art of Cost Optimization - Least Cost Routing and Its 7 Benefits

    The Art of Cost Optimization - Least Cost Routing and Its 7 Benefits

  69. The Best Multi-POP Cloudswitch

    The Best Multi-POP Cloudswitch

  70. The Essential Guide to Business Continuity Plans for VoIP Carriers

    The Essential Guide to Business Continuity Plans for VoIP Carriers

  71. The Essential Guide to Implementing STIR/SHAKEN

    The Essential Guide to Implementing STIR/SHAKEN

  72. The Ultimate Guide to STIR/SHAKEN

    The Ultimate Guide to STIR/SHAKEN

  73. Timeout Protections (SIP Ping, SST)

    Timeout Protections (SIP Ping, SST)

  74. TLS and 2FA Security on the ConnexCS Platform

    TLS and 2FA Security on the ConnexCS Platform

  75. Top 5 Alternative Marketing Strategies for VoIP Businesses

    Top 5 Alternative Marketing Strategies for VoIP Businesses

  76. Top 5 Call Center Challenges and How To Overcome Them

    Top 5 Call Center Challenges and How To Overcome Them

  77. Top 5 Important Types of VoIP Gateways Explained

    Top 5 Important Types of VoIP Gateways Explained

  78. Top 7 Strategies For Ensuring Call Quality While Minimizing Costs with LCR

    Top 7 Strategies For Ensuring Call Quality While Minimizing Costs with LCR

  79. Top 9 Indicators that Help You Identify a Bad Carrier

    Top 9 Indicators that Help You Identify a Bad Carrier

  80. Top 10 Points of Differences Between a Traditional and VoIP Carrier

    Top 10 Points of Differences Between a Traditional and VoIP Carrier

  81. Top 10 Types of Robocall Scams Explained!

    Top 10 Types of Robocall Scams Explained!

  82. Top 10 VoIP Vulnerabilities You Must Know About

    Top 10 VoIP Vulnerabilities You Must Know About

  83. Understanding Global RTP Servers (Lowest Latency Possible, High Availability)

    Understanding Global RTP Servers (Lowest Latency Possible, High Availability)

  84. Understanding Network Address Translation (NAT) - A Beginner's Guide

    Understanding Network Address Translation (NAT) - A Beginner's Guide

  85. Understanding the 9 Key Objectives of a VoIP Network Security Audit

    Understanding the 9 Key Objectives of a VoIP Network Security Audit

  86. Understanding the Complete Scope of a VoIP Network Security Audit

    Understanding the Complete Scope of a VoIP Network Security Audit

  87. Understanding the Crucial Role of Session Border Controllers in Carrier-Grade VoIP Networks

    Understanding the Crucial Role of Session Border Controllers in Carrier-Grade VoIP Networks

  88. Understanding VoIP Anycast Load Balancing

    Understanding VoIP Anycast Load Balancing

  89. Understanding What a PBX System is and How it Benefits Your Business

    Understanding What a PBX System is and How it Benefits Your Business

  90. VoIP Carrier Network Components - Understanding Session Border Controllers

    VoIP Carrier Network Components - Understanding Session Border Controllers

  91. VoIP Carrier Network Security - How to Conduct Security Audit?

    VoIP Carrier Network Security - How to Conduct Security Audit?

  92. VoIP Carrier's Ultimate Guide to Cleaning Up Their Traffic

    VoIP Carrier's Ultimate Guide to Cleaning Up Their Traffic

  93. VoIP Interconnects - Learning How VoIP Carrier Connect and Exchange Traffic

    VoIP Interconnects - Learning How VoIP Carrier Connect and Exchange Traffic

  94. VoLTE - An Evolution in Voice Communication

    VoLTE - An Evolution in Voice Communication

  95. WebPones Explained: Understanding Web-Based Telephonic Communication

    WebPones Explained: Understanding Web-Based Telephonic Communication

  96. WebRTC 101 - The Best Guide for Beginners

    WebRTC 101 - The Best Guide for Beginners

  97. What Are SIP Traces - A Beginners Guide

    What Are SIP Traces - A Beginners Guide

  98. What Are The Top 10 Essential Call Center KPIs?

    What Are The Top 10 Essential Call Center KPIs?

  99. What Are VoIP Gateways and How Do They Work? A Comprehensive Guide

    What Are VoIP Gateways and How Do They Work? A Comprehensive Guide

  100. What is a Contact Center and Why Does Your Business Need One?

    What is a Contact Center and Why Does Your Business Need One?

  101. What is Robocall Mitigation Database? A Guide for Carriers and VoIP Operators

    What is Robocall Mitigation Database? A Guide for Carriers and VoIP Operators

SIP 101 - The Best Guide of 2022

The Internet is a very big place and enables users to do many wonderful things. But the internet is also chaotic. It's like having everyone in the world in the same place at the same time.

Now, the internet lets you communicate with anyone you’d like. There’s one catch, how do you communicate with a particular person in this ocean of internet users?

You could scream someone’s name out, digitally of course, but there would be so many people with the same name on the internet.

Internet communication seems like a tricky task now, doesn’t it?

Thanks to signaling, it's not! Signaling allows internet users to communicate with particular people via text, voice and video.

The signaling protocol that forms the backbone of Internet communication and enables all of this is the Session Initiation Protocol (SIP).

Let’s understand and learn everything about it, shall we?

What is the Session Initiation Protocol (SIP)?

Going by the standard definition, Session Initiation Protocol is a signaling protocol developed by Internet Engineering Task Force for initiating, maintaining, modifying and terminating real-time communications sessions between IP devices.

Text, voice and video and other forms of communication are made possible between two or more IP-enabled devices using SIP.

SIP is widely used in IP telephony, private IP-based communication systems as well as mobile phone calling over Voice over LTE (VoLTE).

SIP is a text-based protocol and many of its elements are based on two more protocols viz. Hypertext Transfer Protocol (HTTP) and Simple Mail Transfer Protocol (SMTP).

So, it follows the same request/response model of HTTP. Being a text-based protocol means that SIP messages are not only easy to read but also easy to debug.

Similar to email headers in SMTP, SIP messages contain all the required metadata to establish a communication session between two users.

With that, you know what a Session Initiation Protocol is. We can not head on and check all its important features.

SIP Features

  • SIP is a signaling protocol and resides in the application layer. It is only used to initiate, modify, manage and terminate communication sessions.
  • Being in the application layer, SIP is independent of the underlying network transport protocol. Network engineers can choose from TCP, UDP, TLS and other NTPs for SIP.
  • The media details are handled by the Session Description Protocol. The bidirectional exchange of text, voice and video data is handled by the Real-time Transport Protocol (RTP).

SIP in OSI Model

  • SIP sessions include Internet Telephony, VoIP, Video Conferencing, or other different forms of unified communication.
  • SIP is secure and supports end-to-end and hop-to-hop authentication. End-to-end encryption is also possible with SIP with an added penalty in the form of latency.
  • Users in a SIP session can communicate using unicast, multicast, broadcast or a combination of these methods.

Now you know the basics and features of the Session Initiation Protocol. Next, we will learn about the Network Elements that make use of SIP for communication.

Network Elements

Something as simple as calling requires a few key players to enable the communication. Let us learn who these players are and what role they play.

User Agent

A User Agent is a network endpoint that sends, receives SIP requests and manages SIP sessions. User agents are the client and server components in the network.

The user agent that sends SIP requests is known as the User Agent Client (UAC) and the one receiving and responding to SIP messages is known as the User Agent Server (UAS).

In other protocols such as HTTP, the role of each UA is fixed as either the client or server. In SIP, a UA can play both roles. The roles are specified and played accordingly only for the duration of that particular SIP session.

Proxy Server

While any two SIP devices can directly communicate with one another, they need an intermediary. This intermediary handles the SIP request, finds its designated recipient, routes the call via the best route and connects the two SIP devices.

A proxy SIP server interprets the SIP message and rewrites its section if necessary before forwarding it to its destination or the nearest proxy server to the destination.

SIP proxy servers are also useful for enforcing telecommunication policies, ensuring the authenticity of the call and determining whether a user is authorized to make the call and use the specific caller ID.

Registrar

A Registrar is a SIP Network Component that accepts REGISTER requests from SIP User Agents. It also records their IP address and other parameters and stores them on an Address on Record (AoR).

When a UAC sends out a SIP request for a UAS, the registrar checks the AoR and forwards the SIP request to the latest registered IP of the particular UAS/SIP endpoint.

The Registrar also manages the expiry for IP addresses. Each UA is allotted a specific time after which their entry in the AoR expires. The UA is required to send a new REGISTER request and continue the cycle.

Session Border Controller

As the name implies, a Session Border Controller (SBC) is a special purpose device that is deployed at the network borders. An SBC protects and regulates the flow of IP-based communication sessions.

Along with maintaining a full session state, an SBC can also undertake the following functions:

Security - Ensuring secure signaling through TLS, IPSec, etc. and secure media transfer using SRTP.

Connectivity - Ensuring seamless connectivity through different parts and components of telephony and IP-based networks.

An SBC achieves this by using NAT traversal, SIP normalization, VPN connectivity, IPv4 to IPv6 networking and many other get-arounds.

Quality of Service - an SBC is usually responsible for implementing and enforcing network policy and traffic prioritization.

This is done by Rate limiting, resource allocation, call admission control, traffic policing and other similar measures/frameworks.

Gateway

A SIP gateway is an interconnect that enables telecom networks using different technologies and protocols to connect with one another.

The telecom sector is not homogeneous in terms of the technology, protocols and communication methods used.

A SIP Gateway allows a user on a VoIP network to easily connect with a user on the analog legacy telephony system or the PSTN.

We only have one part to cover in this article now. So let's head on and learn how SIP works!

How Does SIP Work?

To properly understand how SIP works to connect a call, we will follow the entire process step-wise.

SIP Network Diagram

Step 1 - Sending an Invite

The caller dials a number or places a VoIP call through an App. The App sends the necessary information in the form of a SIP packet to a SIP server.

The SIP server then sends the invitation to the designated recipient. The SIP server informs the caller with a 100 Trying SIP message that it is trying to connect the call.

Step 2 - Connecting

Once the connection request is received by the call recipient, it responds with a 180 Ringing SIP message to the SIP server and rings the phone/device.

The SIP server informs the caller with the same 180 Ringing message and the ringing tone is played at the caller’s end.

Step 3 - Accepted Call

Once the call recipient notices the device ringing and answers the call, the device sends a 200 OK SIP message to the SIP server. The SIP server relays the 200 OK SIP message to the caller.

The caller device sends an ACK SIP message to the SIP server which is relayed by it to the call recipient. Once the ACK SIP message is received, the connection has been established.

Step 4 - Media Transfer

Once the connection has been established, media exchange begins via RTP. SIP is not involved in this step. The media transfer can be a voice, video, text or a combination of the three or some other format of media too.

Step 5 - Termination

Once the caller and call recipient are done talking, either party can terminate the connection by pressing disconnect on their device.

Once the disconnect button is pressed, the device sends a BYE SIP message to the SIP server. The SIP server relays the same to the other party.

The other party responds with an ACK SIP message which is relayed to the first party by the SIP server and thus the connection is terminated.

Other Possibilities

This is how SIP works for a call that connects. For other instances when a call recipient is busy or doesn’t answer, the first two steps remain unchanged.

In case the call recipient doesn’t respond, his device responds with a 486 BUSY SIP message and the SIP server relays it to the caller.

SIP Diagram for an Unanswered Call

The caller device responds with an ACK SIP message and informs the caller that the call recipient is busy.

In case the call recipient declines the call, his device responds with a 603 Decline SIP message and the SIP server relays the same to the caller.

SIP Diagram for a Declined Call

The caller device responds with an ACK SIP message and informs the caller that the call recipient has declined the call.

Wrapping Up

Session Initiation Protocol is important for not only VoIP communication but also for other forms of communication.

SIP being simple and text-based make it the top choice for establishing communication over IP. With the knowledge you’ve gained through this article, you will be able to understand the ins and out of IP-based communication.

Learning about how technologies that drive our daily lives is helpful in many ways. We hope this article covered everything you wanted to learn about SIP.