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WebRTC with SIP Over WebSockets

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- product:voice - product:video SignalWire supports industry-standard WebRTC SIP over WebSockets! This means that you can use off-the-shelf JavaScript libraries with SIP to connect to SignalWire services. Services enabled in a WebRTC compliant browser include: - Audio calling to/from Web and PSTN - product:video calling between Web and SIP endpoints - Programmatically creating SIP endpoints and directing calls to your WebRTC endpoints To get started, navigate to your SignalWire Space and create a new SIP endpoint. To use secure protocols, make sure the Encryption is **required**. Now, you can test the newly created endpoint on a popular JS SIP (JSSIP) library:

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If you click the **gear** icon, you can configure the settings needed to connect to SignalWire. The **SIP URI** is your SignalWire SIP username and domain. The password set on your SIP endpoint can be used in the **SIP password** section to authenticate. The **WebSocket URI** is your domain with an appended **wss://**. To match the encryption on SignalWire, set the transport to TLS. The Registrar Server should be set to your domain.
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Once youve saved the entered information, you will see the endpoint register and turn from yellow to green as it appears below:
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Then, set up another SIP endpoint with the same process as above:
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Now that both of the endpoints are registered, you can place the first call. If youre using SIP endpoints in the same project, you can use short-hand names (for example, jssip1 and jssip2 as seen in the images above) to place the call:
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If everything was set up correctly, you should see something similar to this:
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In addition, you can attribute a phone number to your created sip endpoint and receive calls from the PSTN! You can also place the outbound calls to the PSTN from the WebRTC client.
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