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articles/communication-services/tutorials/proxy-calling-support-tutorial.md

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# How to force calling traffic to be relayed and proxyed across your own servers
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# How to force calling traffic to be relayed and proxied across your own servers
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In certain situations it might be useful to have all your client traffic be proxyed to a TURN server that you can control. This tutorial will walk you through the process to ena. When the SDK is initializing you can provide the details of your TURN servers that you would like the traffic to route to. When this is enabled all the media traffic (audio/video/screen sharing) will flow through the provided TURN servers instead of the Azure Communication Services defaults. Once your provide your TURN server details while initializing the `CallClient`, all the calls to and from this `CallClient` will be using the provided TURN servers. Below is a guide on how to do provide your TURN configurations to the WebJS SDK.
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In certain situations, it might be useful to have all your client traffic be proxied to a TURN server that you can control. When the SDK is initializing, you can provide the details of your TURN servers that you would like the traffic to route to. Once enabled all the media traffic (audio/video/screen sharing) travels through the provided TURN servers instead of the Azure Communication Services defaults. When you define that a TURN server should be used wyen initializing the `CallClient`, all the calls to and from this `CallClient` will be using the provided TURN servers. This tutorial will guide on how to do provide your TURN configurations to the WebJS SDK.
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>[!IMPORTANT]
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> The custom proxy feature is available starting in the public preview version [1.13.0-beta.4](https://www.npmjs.com/package/@azure/communication-calling/v/1.13.0-beta.4) of the Calling SDK. Please ensure that you use this or a newer SDK when trying to use this feature. This quickstart uses the Azure Communication Services Calling SDK version greater than `1.13.0`.
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[!INCLUDE [Public Preview](../includes/public-preview-include-document.md)]
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## What is a STUN/TURN server?
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Many times establishing connection between two peers is not straightforward and a direct connection will not work because of many reasons - firewalls with strict rules, peers sitting behind a private network, etc. In many situations your device does not have a public IP address to establish a connection straightaway and so relaying data via a relay server, that is usually known to both peers, is a way to allow this connection to happen. The WebRTC ICE (Interactive Connectivity Establishment) framework helps make this happen. STUN and TURN servers are the relay servers here. See [Network Traversal Concepts](../concepts/network-traversal.md) for more details on how ACS mitigates network challenges with STUN and TURN.
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Many times establishing connection between two peers is not straightforward and a direct connection will not work because of many reasons - firewalls with strict rules, peers sitting behind a private network, etc. In many situations your device does not have a public IP address to establish a connection straightaway and so relaying data via a relay server, that is usually known to both peers, is a way to allow this connection to happen. STUN and TURN servers are the relay servers here. See [Network Traversal Concepts](../concepts/network-traversal.md) for more details on how ACS mitigates network challenges with STUN and TURN.
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## Providing your TURN servers to the SDK
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To provide the details of your TURN servers, you need to pass details of what TURN server to use as part of `CallClientOptions` while initializing the `CallClient`. See the quickstart on how to setup Voice and Video calling using the Web SDK at - [Azure Communication Services Web SDK](../quickstarts/voice-video-calling/get-started-with-video-calling.md?pivots=platform-web)).

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