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Frequently Asked Questions

How does Twilio charge for usage?

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Twilio charges according to the amount of data that is relayed by the TURN server. The TURN Client is responsible for allocating a relayed address on the TURN server, which is also referred to as a TURN session. The total data that has been relayed is calculated as the sum of bytes sent and received by the TURN Client. A charge is then issued to the Twilio Account SID associated with the TURN Client that created the session. The charge is based on the total data relayed, which is measured in megabytes. Note that there are different usage rates per Twilio region.

What are STUN, TURN, and ICE?

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STUN, TURN, and ICE are a set of IETF standard protocols for negotiating traversing NATs when establishing peer-to-peer communication sessions. WebRTC and other VoIP stacks implement support for ICE to improve the reliability of IP communications.

A host uses Session Traversal Utilities for NAT (STUN) to discover its public IP address when it is located behind a NAT/Firewall. When this host wants to receive an incoming connection from another party, it provides this public IP address as a possible location where it can receive a connection. If the NAT/Firewall still won't allow the two hosts to connect directly, they make a connection to a server implementing Traversal Using Relay around NAT (TURN), which will relay media between the two parties.

Interactive Connectivity Establishment (ICE) is a blanket standard that describes how to coordinate STUN and TURN to make a connection between hosts. Twilio's Network Traversal Service implements STUN and TURN for ICE-compatible clients, such as browsers supporting the WebRTC standard.

Want to learn more? Check out the following RFCs:

How do STUN, TURN and ICE work?

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Let's assume our users, Alice and Bob, are both using a WebRTC video chat application, and that Alice wants to call Bob. Here's what happens next.

To connect to Bob's browser, Alice's browser needs to generate a Session Description Protocol (SDP) offer. The SDP generation process begins when the application she's using calls createOffer on an RTCPeerConnection object.

An SDP offer contains a bunch of information about the session Alice's browser wants to establish-what codecs to use, whether this will be an audio or video session, and more. It also contains a list of ICE candidates, which are the IP and port pairs that Bob's browser can attempt to use to connect to Alice.

To build the list of ICE candidates, Alice's browser makes a series of requests to a STUN server. The server returns the public IP address and port pair that originated the request. Alice's browser adds each pair to the list of ICE candidates. This process is called gathering ICE candidates. Once Alice's browser has finished gathering ICE candidates, it can return an SDP.

Next, Alice's browser needs to pass the SDP to Bob's browser through a signaling channel between the browsers-WebRTC leaves this signaling implementation up to the developer. The ins and outs of signaling are beyond the scope of this discussion, but let's assume Bob receives Alice's SDP offer via some signaling channel.

Now, Bob's browser needs to generate an SDP answer. Bob's browser follows the same steps Alice's browser used above: gathering ICE candidates, etc. Bob's browser then needs to return this SDP answer to Alice's browser.

Once Alice and Bob have exchanged SDPs, they then perform a series of connectivity checks. The ICE algorithm in each browser takes a candidate IP/port pair from the list it received in the other party's SDP, and sends it a STUN request. If a response comes back from the other browser, the originating browser considers the check successful and will mark that IP/port pair as a valid ICE candidate.

After connectivity checks have finished on all of the IP/port pairs, the browsers negotiate and decide to use one of the remaining, valid pairs. Once a pair is selected, media begins flowing between the browsers. This entire process usually takes milliseconds.

If browsers cannot find an IP/port pair that passes connectivity checks, they will initiate STUN requests to the TURN server to obtain a media relay address. A relay address is a public IP address and port that forwards packets received to and from the browser that established the relay address. This relay address is then added to the candidate list and exchanged via the signaling channel.

If you're building a WebRTC application, the WebRTC stack includes an ICE Agent that takes care of most of this for you. You just need to implement a signaling mechanism to exchange SDPs and send along new ICE candidates whenever they're discovered.

What TLS version and cipher suites are supported?

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The Twilio Network Traversal Service only supports TLS 1.2. The following is the supported cipher suite:


How can I troubleshoot ICE negotiation in my application?

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Using Google Chrome

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In a new tab, open chrome://webrtc-internals. In a separate tab, make a WebRTC call using your application. In the webrtc-internals page, you'll see a tab for each active PeerConnection object. That page lists the ICE negotiation events that took place when attempting to setup the call (iceGatheringStateChange, onIceCandidate, etc.). You can expand each node in the tree to see more detail about the event.

In a new tab, open about:webrtc. In a separate tab, make a WebRTC call using your application. In the about:webrtc page click the Connection Log button. This will reveal a log of events. Search for ICE and STUN/TURN events by searching for the string 'ICE' in this log file.

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