Implemented firewall passthrough (STUN, TURN)

This article is a brief introduction about firewall passthrough technologies in relation with Ozeki VoIP SIP SDK. After reading through this page you will be fully familiar with all the essential terms concerning firewall passthrough methods and what you will need for creating your own solution using Ozeki VoIP SIP SDK.

Introduction

Increased penetration of broadband Internet is driving the adoption of Voice over Internet Protocol (VoIP) both for consumer and business markets, and VoIP is now on its way to becoming the main mode of multimedia communications in the coming years. But for something becoming so mainstream it is inevitable not to bump into some smaller difficulties on its way to popularity.

In case of VoIP technology this small bump is the issue of Network Address Translators, or shortly, NATs. NATs are common devices that translate private IP addresses to public IP addresses, this way, they 'hide' the private networks. It provides security but it also obstruct VoIP technology, because connections can only be initiated from the private network to the Internet, but not in reverse. Since VoIP's main essence is to establish connections both ways, this can mean a problem (Figure 1).

Figure 1 - The STUN firewall passthrough method

The same issue can also stand in case of an impenetrable firewall. Many firewalls only allow connections to be initiated from the private network. They commonly deny access to port numbers associated with VoIP. They can even inspect the packet contents to identify and reject VoIP traffic.

Now we have seen how NAT/firewalls present a challenge to VoIP call completion. But there are some clever solutions which have to be mentioned at this point. There are two protocols that can be helpful in this situation: STUN and TURN.

STUN is an acronym for Session Traversal Utilities for NAT. STUN is a lightweight client-server protocol requiring only simple query and response. The client side is implemented in the user's communications application, such as a Voice over Internet Protocol (VoIP) phone or instant messaging client. The base protocol operates essentially as follows. The client, often operating inside a private network, sends a binding request to a STUN server on the public Internet (Figure 1). The STUN server sends a success response that contains the IP address and port as observed from its perspective. The result is usually XOR mapped to avoid translation of packet contents.

Traversal Using Relay NAT protocol, in short term TURN is an extension for STUN protocol to facilitate NAT traversal when both endpoints are behind symmetric NAT. With TURN, media traffic for the session will have to go to a relay server. Since relaying is expensive, in terms of bandwidth that must be provided by the provider and additional delay for the media traffic, TURN is normally used as the last resort when endpoints cannot talk directly.

Ozeki VoIP SIP SDK provides a great background support for all the technologies and functionalities that are mentioned above in this article. If you want to create a VoIP solution that has all this support, you only need to download, install and use Ozeki VoIP SIP SDK.

This article introduced you the basic knowledge about firewall passthrough technologies and showed how Ozeki VoIP SIP SDK can help you to fulfill your wishes about this topic. If you have read through this page carefully, you already have all the knowledge you need to start on your own solution.

As you are now familiar with all the terms concerning this topic, now it is time to take a step further and explore what other extraordinary solution Ozeki VoIP SIP SDK can provide to you.

If you have any question or need assistance, please contact us at info@voip-sip-sdk.com

You can select a suitable Ozeki VoIP SIP SDK license for your project on Pricing and licensing information page

Related Pages

• Setup Ozeki VoIP SIP SDK efficiently: Quick start guide
• Download Ozeki VoIP SIP SDK form the Ozeki VoIP SIP SDK download page
• You can find licensing information of Ozeki VoIP SIP SDK on Pricing and licensing information page