Information Added to IEFT Draft on WebRTC Audio Codecs for Interoperability
Given all the buzz in the past year regarding “codec wars” and the need for universal interoperability regarding WebRTC adoption, it is always useful to check in on The Internet Engineering Task Force (IETF) Networking Group to see what is on the minds of those seeking to push WebRTC ahead.
The latest informational submission provided by Stephane Proust from one of WebRTC’s biggest supports Orange, is titled, Additional WebRTC audio codecs for interoperability.draft-ietf-rtcweb-audio-codecs-for-interop-03. As a WebRTC Solutions Community alert, this is one that deserves some attention.
What Proust lays out in the abstract of the draft is:
“To ensure a baseline level of interoperability between WebRTC clients, a minimum set of required codecs is specified. However, to maximize the possibility to establish the session without the need for audio transcoding, it is also recommended to include in the offer other suitable audio codecs that are available to the browser. This document provides some guidelines on the suitable codecs to be considered for WebRTC clients to address the most relevant interoperability use cases.”
I will dispense with the suspense. Proust provides a description, WebRTC relevant use case and suggested guidelines for usage and implementation for three “other” codecs with the goal being the elimination of transcoding. These include:
The Adaptive Multi-Rate WideBand (AMR-WB): a 3GPP defined speech codec that is mandatory to implement in any 3GPP terminal that supports wideband speech communication. It is being used in circuit switched mobile telephony services and new multimedia telephony services over IP/IMS like for voice over LTE as specified by GSMA.
Adaptive Multi-Rate (AMR): a 3GPP defined speech codec that is mandatory to implement in any 3GPP terminal that supports voice communication, i.e. several hundred millions of terminals. This includes both mobile phone calls using GSM and 3G cellular systems as well as multimedia telephony services over IP/IMS and 4G/VoLTE, such as GSMA voice IMS profile for VoLTE.
G.722: an ITU-T defined wideband speech codec. G.722 was approved by ITU-T in 1988. It is a royalty free codec that is common in a wide range of terminals and endpoints supporting wideband speech and requiring low complexity. The complexity of G.722 is estimated to 10 MIPS, which is 2.5 to 3 times lower than AMR-WB. G.722 has been chosen by ETSI DECT as the mandatory wideband codec for New Generation DECT, with a goal to greatly increase voice quality by extending the bandwidth from narrowband to wideband. G.722 is the wideband codec required for CAT-iq DECT certified terminals and the V2.0 of CAT-iq specifications have been approved by GSMA as minimum requirements for HD voice logo usage on "fixed" devices; i.e., broadband connections using the G.722 codec.
As has been mentioned previously by many members of the WebTRC, the history of communications has demonstrated over and over that all boats really do rise when the tide comes in. This has been true with basic telephony, fax, SMS, email, etc. It obviously needs to be true with WebRTC and as this contribution from Proust demonstrates, assuring there are not islands of communications that can be obstacles to WebRTC adoption, the community is hard at work to resolve these types of issues.
Edited by Kyle Piscioniere