WEBRTC GATEWAY FUNCTIONALITIES
Bridge voice & video communication between WebRTC and SIP
Use our WebRTC Gateway functionalities for WebRTC to SIP or SIP to WebRTC communication.
What are the WebRTC Gateway functionalities?
With our gateway, WebRTC clients can be integrated to a PBX or contact center that does not natively support WebRTC. The WebRTC gateway ensures that the communication between the application in the web browser and the PBX/contact center is securely established using public or private SIP trunks.
The WebRTC-enabled device is, just like any other SIP device, registered and configured using the iotcomms.io SIP Server functionality.
How does the WebRTC Gateway work?
The WebRTC gateway works as an interworking function between a mix of SIP and WebRTC devices, which significantly simplifies the communication between browser-based applications to SIP devices. Calling between devices with different protocols have never been easier!
Voice calls between WebRTC clients and external numbers are set up using public SIP trunks. Voice or video calls to an external application such as a PBX or contact center are set up using a private SIP trunk.
- Voice & video calling between WebRTC clients and SIP devices
- Voice calling between WebRTC clients and external numbers with public SIP trunk
- Voice & video calling between WebRTC clients and external applications with private SIP trunk
Manage your WebRTC clients and SIP devices with RESTful APIs
Create and manage user accounts using standard RESTful API calls or accessing the customer web portal. Upon authentication and registration of the accounts, sessions can be set-up end-to-end. Our SIP server functionality will take care of your connectivity in a reliable and secure way. Additional APIs can be used to monitor the status and activity of the WebRTC clients and SIP devices as well as retrieve logs and statistics.
A device monitoring API is provided to listen for SIP registration and dialog state change callback events. The monitoring API can be used to develop applications showing if a device is on-line or in a call.
Connect to external applications or make external voice calls with private or public SIP trunks
Encryption & Authentication
Traffic is encrypted and authenticated using well-proven encryption technologies. Devices that encrypt signaling using SIP/TLS or SIP/Secure websocket and media encryption using DTLS/SDES-SRTP are extra well protected.
For devices not supporting this an IPSEC VPN connection is used. All data is stored encrypted in our systems.
APIs tailored for developers
We package telecom complexity and protocol fragmentation in easy-to-use APIs for you to save time and cost, and offer superior performance. Explore getting started guides and APIs in our Developer Area.
Benefits of the WebRTC Gateway
The WebRTC Gateway can be used by many
Health & Social Care
Build video visit solutions, virtual home care and monitoring services.
Establish video calls between users and agents as a result of an alarm or sensor event.
Connect clients on your web with phone based operators in your contact center.
Add softphone capabilities and let co-workers make videocalls from computers.
Power your buildings with video communication and mobile surveillance.
Talk to us today
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Built for mission critical alarm, voice & video services – delivering superior reliability, security and availability.
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