Objective
Explore and test SIP Interconnect using Playground. Dial a SIP call to connect in the Video API session.
Applies To
- Video API
- SIP Interconnect
Procedure
- Go to Playground Opentok Environment or Playground Unified Environment.
- Create a new session. You can use the default values or modify them if needed. Click on Create.
- You will now see your session information. Click on Connect to join the session.
- Click on Publish Stream to publish your audio and video.
- Go to SIP. Add your SIP URI. The SIP URI is a required field. Confirm that you have the correct format of the SIP URI sip:<sipNumber>@<sipDomain>. For example, sip:12222221111@sip-us.vonage.com where 12222221111 is the Vonage Number or Long Virtual Number (LVN). If using a Vonage Number or LVN, refer to Vonage SIP Endpoint for more information on the SIP URI domain.
- If required by your SIP platform, add your Username, Password and From. When using a Vonage Number or LVN, you have to use your API Key as the username, API Secret as the password and the required Vonage Number (LVN) as the From. If you need a Vonage Number for a quick test, you can sign up for a Vonage account on this sign-up link.
- Then, click on Dial SIP Call.
- Once the SIP endpoint has been answered and connected to the session, you will see the SIP endpoint as a subscriber with audio-only.
Additional Information
Learn more about the SIP Interconnect Opentok Environment or SIP Interconnect Unified Environment feature in this link.
Related to:
Articles in this section
- How to dial up Third Party applications using SIP Interconnect
- How to set up and test SIP Outbound Video call using linphone and playground?
- How to use SIP Interconnect in Playground
- How to Configure SIP Monitoring Callbacks
- Which IP addresses should I allow for SIP interconnect when using Video API?
- Unable to Initiate SIP Call from Playground
- How to Send DTMF in a SIP Call using Playground
- When initiating a SIP call, it fails with '706 Call Reached Maximum Inactive Duration'
- How to Dial DTMF or Digit Extension in SIP Interconnect?
- Can a SIP user mute/unmute using DTMF?