Hello Osmocom, I have written a new specification, in the style of 3GPP specs, for enhanced RTP transport of FR and EFR codec frames in an IP-based GSM RAN, addressing problem areas where the general standard RTP format for these codecs creates functional regressions relative to the original T1/E1 Abis architecture with TRAU frames. The new spec has its official home here: https://www.freecalypso.org/specs/tw-ts-001-v010001.txt and here is a pair of patches adding OsmoBTS support for accepting this TW-TS-001 format in RTP input and optionally (per vty config) emitting it in RTP output: https://cgit.osmocom.org/osmo-bts/log/?h=falconia/rtp_traulike The code patches are just for better context - they will go to Gerrit for review later - but right now I am soliciting a review of the specification, rather than code implementation. I am not aware of any established process in Osmocom for reviewing new specifications (different from code review), as writing new 3GPP-style specs is not something this community does often. But in the present case I am genuinely convinced that the Internet standard RTP format for GSM-FR and GSM-EFR (written with VoIP rather than GSM RAN in mind) is truly deficient for GSM RAN purposes, especially for those who wish to deploy their GSM networks in a retronetworking fashion, and the only way to bring back the full set of E1 Abis bells and whistles over IP is to invent our own (completely optional) pseudostandard format that will likely only be used within {Osmocom+Themyscira} community. With this context in mind, I cordially invite all of you, esteemed GSM FOSS developers, to review the new specification linked above. Sincerely, Mother Mychaela