WebRTC brings voice, video and data communication to the browser and provides Web developers with no VoIP experience easy access to VoIP communication technology. To enrich the user experience and create enhanced services integrated with the Web, existing VoIP deployments need to be connected with WebRTC. Meeting this need while providing a cost effective, high quality solution, requires supporting WebRTC both at the demarcation point as well as on the client side.
AudioCodes provides a comprehensive WebRTC offering featuring the following components:
- Support for WebRTC in the Mediant SBC family functioning as a WebRTC Gateway
- Opus and media encryption support in the 440HD SIP Phone making it a WebRTC IP Phone
- Detect and correct call quality using the Session Experience Management (SEM)
WebRTC enabled Session Border Controllers (SBC)
The SBC is the enterprise demarcation point between the business’s VoIP network and the service provider’s SIP Trunk. It performs SIP protocol mediation, security and media handling. Given its important position in the network, the SBC is the natural network element for terminating WebRTC traffic and bridging it in to the enterprise network.
The AudioCodes Mediant SBC family supports the following capabilities for WebRTC:
- Conversion of SIP/WebSockets to SIP/UDP, TCP or TLS
- Opus transcoding when required
Support for Opus media end-to-end
For achieving high quality audio in a cost effective solution, it is preferred not to perform media transcoding in the SBC but rather to maintain Opus end-to-end from the browser to the business IP phone. To serve this need AudioCodes added the required media capabilities in its 440HD SIP Phone.
Support for the WebRTC voice Opus codec and encryption algorithms natively on the enterprise IP Phone yield the following benefits:
- Better call quality
- Reduced cost
- Easier migration to multi-purpose cloud platforms
AudioCodes included support for Opus and WebRTC media encryption in its 440HD SIP Phone. The following architecture allows for experiencing this technology:
In this architecture, the user calls the AudioCodes 440HD SIP Phone from the browser. The AudioCodes SBC terminates the WebSockets coming from the browser and passes the media directly to the IP Phone.
Call quality monitoring and enhancement
Traffic traverses between networks and different network types (Wi-Fi, Wireline Ethernet, public Internet and enterprise networks) while devices themselves can vary as well. WebRTC enabled devices may reside on all of these networks, additionally, when calls are connected between WebRTC to non-WebRTC endpoints, clients and codecs used are not always built to handle the network impairments introduced.
Handling of network impairments should be done in the entity that connects between these networks as it is familiar with the requirements of each and has the per-session knowledge of source and destination of traffic. The AudioCodes Mediant SBC family functions as a demarcation point between these devices and networks, it utilizes the AudioCodes media enhancement algorithms to mitigate network impairments and yield best call quality possible.
Another missing piece in typical deployments is the ability to monitor the traffic, know what is happening on the network and impact SBC decisions for quality improvements.
The AudioCodes Session Experience Management (SEM) not only monitors and detects voice call quality issues, it also works in harmony with the SBC to allow for smart, quality based, routing and quality improvements.