AMR

To ensure a baseline of interoperability between WebRTC endpoints, a minimum set of required codecs is specified. However, to maximize the possibility of establishing 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 endpoints to address the use cases most relevant to interoperability.

This document specifies a Real-time Transport Protocol (RTP) payload format to be used for Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) encoded speech signals. The payload format is designed to be able to interoperate with existing AMR and AMR-WB transport formats on non-IP networks. In addition, a file format is specified for transport of AMR and AMR-WB speech data in storage mode applications such as email. Two separate media type registrations are included, one for AMR and one for AMR-WB, specifying use of both the RTP payload format and the storage format. This document obsoletes RFC 3267. [STANDARDS-TRACK]

This document specifies a Real-time Transport Protocol (RTP) payload format for Extended Adaptive Multi-Rate Wideband (AMR-WB+) encoded audio signals. The AMR-WB+ codec is an audio extension of the AMR-WB speech codec. It encompasses the AMR-WB frame types and a number of new frame types designed to support high-quality music and speech. A media type registration for AMR-WB+ is included in this specification. [STANDARDS-TRACK]