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According to reports from developers of Internet audio applications and operators of Internet audio services, there are no standardized, high-quality audio codecs that meet all of the following three conditions: 1. Are optimized for use in interactive Internet applications. 2. Are published by a recognized standards development organization (SDO) and therefore subject to clear change control. 3. Can be widely implemented and easily distributed among application developers, service operators, and end users. There exist codecs that provide high quality encoding of audio information, but that are not optimized for the actual conditions of the Internet; according to reports, this mismatch between design and deployment has hindered adoption of such codecs in interactive Internet applications.

The goal of this working group is to develop a single high-quality audio codec that is optimized for use over the Internet and that can be widely implemented and easily distributed among application developers, service operators, and end users. Core technical considerations include, but are not necessarily limited to, the following: 1. Designing for use in interactive applications (examples include, but are not limited to, point-to-point voice calls, multi-party voice conferencing, telepresence, teleoperation, in-game voice chat, and live music performance) 2. Addressing the real transport conditions of the Internet as identified and prioritized by the working group 3. Ensuring interoperability with the Real-time Transport Protocol (RTP), including secure transport via SRTP 4. Ensuring interoperability with Internet signaling technologies such as Session Initiation Protocol (SIP), Session Description Protocol (SDP), and Extensible Messaging and Presence Protocol (XMPP); however, the result should not depend on the details of any particular signaling technology