Bridge voice & video communication between WebRTC and SIP

Use our WebRTC Gateway functionalities for WebRTC to SIP or SIP to WebRTC communication.

Bridge voice & video communication between WebRTC and SIP with iotcomms.io WebRTC Gateway as a Service


What are the WebRTC Gateway functionalities?

Simplify communication between browser-based applications and SIP devices with iotcomms.io WebRTC Gateway as a Service
The iotcomms.io WebRTC Gateway provides the functionalities required to enable voice & video communication between WebRTC and SIP devices.

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​​
Connect to external applications or make external voice calls with private or public SIP trunks

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 (PBX, Contact Center, Customer Support Center etc.) using private SIP trunks, and make external voice calls using public SIP trunks. The SIP Server automatically handles NAT traversal.

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

  • Everything runs in the cloud​

    No need to setup, run or maintain a SIP infrastructure. Just configure the SIP user accounts using our simple API, configure your devices and start deploying your solution.

  • Add WebRTC to existing system​

    Add WebRTC users to your non-WebRTC supported voice system – enable Contact Centers, PBXs etc to use WebRTC functionality.

  • Out-of-the-box security​

    No need to worry about eavesdropping or manipulation of SIP or media traffic since all traffic is encrypted & authenticated using well-proven encryption technologies. All data is stored encrypted in our systems.

  • Avoid codec mismatching

    Automatic transcoding of codecs between your WebRTC client/SIP device and your trunk provider.

  • Broad range of codecs & protocols

    Codecs: D G.711, OPUS, VP8, H.264.
    Protocols: SIP, WebSocket, RTP, SRTP, DTLS, IC.

  • Pay only for what you use

    The web portal provides an overview of service usage, and as for all other functionality the data is accessible using APIs. Our microservice-based design enables you to pay only for what you use.


The WebRTC Gateway can be used by many

Health & Social Care

Build video visit solutions, virtual home care and monitoring services.

Security Industries

Establish video calls between users and agents as a result of an alarm or sensor event.

Contact Centers

Connect clients on your web with phone based operators in your contact center.

Unified Communication

Add softphone capabilities and let co-workers make videocalls from computers.


Power your buildings with video communication and mobile surveillance.

Get Started

Learn more about designing your solution with our granular & feature rich communication APIs.

Modern CPaaS built cloud-native from ground up – we run the operations for you so you can focus on your customers’ experience.


Built for mission critical alarm, voice & video services – delivering superior reliability, security and availability.

Built with serverless functions in AWS for unlimited scale, reach and global deployment – extend to new markets quick and easy.

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