Re: [payload] draft-ramalho-payload-g7110-00 to be discussedinAVTEXT

[Hadriel Kaplan <[e-mail]>]

On Jul 26, 2011, at 6:43 PM, Michael Ramalho (mramalho) wrote:

> While a transcode from G.711 to G.729 "saves bandwidth" ... it also
> DEGRADES VOICE QUALITY.

It doesn't matter whether a particular codec or payload transform is lossless or not, from the perspective of SDP and RTP.  SRTP encrypt/decrypt and authentication is also completely "lossless".  You could, in theory, do SRTP between two middleboxes.  All you'd need is a way to exchange keys between the two middleboxes, just like for G.711.0 you need a way to exchange payload types and companding.  Obviously SBCs and other middleboxes can and do SRTP termination/origination, but we would never have done it the way this draft is describing to do G.711.0.  I must be missing something about your use-case that makes it different.


> Do we signal to the endpoints that RTP header compression occurred
> somewhere on the end-to-end path? I don't think so.
> 

ROHC for RTP only works in extremely constrained deployment scenarios, and as far as I'm aware can only be done either by the devices generating the RTP so they know what IP/UDP combo stream is RTP before-hand, or by channelization/separation and signaling for a context.  And as far as I know it has signaling in ROHC itself - it's not just blindly starting to compress without knowing the far-end is capable of decompressing and what modes. 

-hadriel