AudioCodes WebRTC Solution

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The key takeaways are that AudioCodes provides a comprehensive WebRTC solution featuring infrastructure, endpoint and quality monitoring products to bring WebRTC traffic into enterprise networks reliably and with high call quality.

AudioCodes' WebRTC solution includes infrastructure products like the Mediant SBC family, the 440HD SIP phone, and the Session Experience Management solution for call quality monitoring and enhancement.

AudioCodes' Mediant SBC utilizes its media enhancement algorithms to mitigate network impairments between different devices and networks, yielding the best possible call quality.

WebRTC Solution

Enterprises Contact Centers Service Providers


The AudioCodes WebRTC solution brings the WebRTC experience to the fingertips of enterprise UC and Contact Center
solutions. Comprising infrastructure, endpoint and quality monitoring products, the AudioCodes WebRTC solution
includes all that is required to bring WebRTC traffic into the enterprise and service providers network with the quality
and reliability expected from these networks.

AudioCodes provides a comprehensive WebRTC offering featuring the following components:

Support for WebRTC in the Opus and media encryption support Detect and correct call quality using the
Mediant SBC family in the 440HD SIP Phone Session Experience Management (SEM)

Solution benefits

Fully integrated in the AudioCodes SBC, no external box required

Secured and reliable

Scalable

Minimal transcoding required while maintaining high call quality

Call quality monitoring and enhancement

Opus transcoding

Available in Hardware and Virtualized configurations


Support for WebRTC in the SBC
The SBC is the enterprise demarcation point between the businesss VoIP network and the service providers 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.

Some solutions on the market have chosen to implement the WebRTC termination in an external box, a complex architecture
that also hinders security. By implementing WebRTC natively in the AudioCodes Session Border Controller (SBC), an entity that is
designed to reside on the public Internet and sustain security threats, and by avoiding the need to add this capability to an external
box, network security is maintained.

The AudioCodes SBC supports SIP over WebSockets for communicating with WebRTC clients. Choosing the commonly used signaling
protocol in enterprise and service providers networks allows for easy integration of WebRTC clients with telephony features available
in the existing network.

AudioCodes Mediant SBC supported features for WebRTC include:


ICE WebSockets and Secured WebSockets Opus transcoding VP8 and H.264 (Video) pass trough DTLS RTP multiplexing
RTP/SAVPF RFC5124 Client sample application Available in M800, Mediant 2600, Mediant 4000, Mediant 9000 Mediant VE/SE SBCs

Hybrid SBC and Media Gateway


The AudioCodes hybrid SBC and Media GW products such as the Mediant 800B are also optionally enabled with WebRTC capabilities.
By this it is possible to streamline WebRTC traffic directly to PSTN and E1/T1 interfaces.

Call quality monitoring and enhancement


Traffic traverses between networks and different network types (Wi-Fi, Wireline Ethernet, public Internet and enterprise networks)
while the 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 to the network.

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 the best possible call quality.

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.
Support for Opus media end-to-end
For achieving high quality audio in a cost effective solution, it is preferred, when possible, not to perform media transcoding but
rather to maintain Opus end-to-end from the browser to the business IP phone. To meet this need AudioCodes added the required
media capabilities to its 440HD SIP Phone.

Support for the WebRTC voice Opus codec and encryption algorithms natively on the enterprise IP Phone yields the following
benefits:

Better call quality


Privacy
Scalability
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 the benefits of this technology:

Hosted UC/CC;Support
ST Services adding
for WebRTC a WebRTCCommunication
in Enterprise client

SIP Trunk

Multi-Tenant
Services
WAN Peering SBC

Hosted UC/CC
or SIP Trunk

Multitenant Access SBC

Mediant E-SBC

MediaPack

Analog Devices IP Phones

In this architecture, the user calls the AudioCodes 440HD SIP Phone from the browser.
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The AudioCodes SBC terminates the WebSockets coming from the browser and passes the media directly to the IP Phone.
WebRTC technology on the AudioCodes blog

Going Native with WebRTC


http://blog.audiocodes.com/going-native-with-webrtc-2

WebRTC Connectivity Solution with Focus on Quality and Scale


http://blog.audiocodes.com/webrtc-connectivity-solution-with-focus-on-quality-and-scale

1 Answer to WebRTC Signaling


http://blog.audiocodes.com/1-answer-to-webrtc-signaling

Why not All WebRTC GWs Were Born Equal


http://blog.audiocodes.com/why-not-all-webrtc-gws-were-born-equal

Moving WebRTC Into Your Network Through the Front Door


http://blog.audiocodes.com/moving-webrtc-into-your-network-through-the-front-door

About AudioCodes
AudioCodes Ltd. (NasdaqGS: AUDC) designs, develops and sells advanced Voice over IP (VoIP) and converged VoIP and Data networking products and
applications to Service Providers and Enterprises. AudioCodes is a VoIP technology market leader focused on converged VoIP & data communications
and its products are deployed globally in Broadband, Mobile, Enterprise networks and Cable. The company provides a range of innovative, cost-effective
products including Media Gateways, Multi-Service Business Routers, Session Border Controllers (SBC), Residential Gateways, IP Phones, Media Servers
and Value Added Applications. AudioCodes underlying technology, VoIPerfectHD, relies on AudioCodes leadership in DSP, voice coding and voice
processing technologies. AudioCodes High Definition (HD) VoIP technologies and products provide enhanced intelligibility and a better end user
communication experience in Voice communications.

International Headquarters AudioCodes Inc. 2015 AudioCodes Ltd. All rights reserved. AudioCodes, AC,
HD VoIP, HD VoIP Sounds Better, IPmedia, Mediant, MediaPack,
1 Hayarden Street, 27 Worlds Fair Drive, OSN, SmartTAP, VMAS, VoIPerfect, VoIPerfectHD, Your Gateway
Airport City Somerset, NJ 08873 To VoIP and 3GX are trademarks or registered trademarks of
AudioCodes Limited. All other products or trademarks are
Lod 7019900, Israel Tel:+1-732-469-0880 property of their respective owners. Product specifications are
Tel: +972-3-976-4000 Fax:+1-732-469-2298 subject to change without notice.
Fax: +972-3-976-4040 Ref # LTRM-20103 03/15 V.1

Contact us: www.audiocodes.com/info


Website: www.audiocodes.com

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