Re: [MMUSIC] Offer/Answer PT Questions

[Roman Shpount <[e-mail]>]

On Wed, Feb 24, 2016 at 12:15 PM, Dale R. Worley <[e-mail]> wrote:

> This may break down if there is an RTP gateway in the middle, and while
> the ultimate destination of the RTP is different, the sender is always
> given the same destination address in the rewritten SDP that it is sent.
> But such a gateway has problems knowing which destination to forward the
> RTP to.
>
> What do people at sip-implementors say about this?
>

Speaking with an implementer hat on, based on the interop that I have
conducted for audio based networks, most of them ignore the MUST NOT reuse
PT rule. You routinely see PT changed or reused.

To deal with this, if implementation is robust remote address change, SSRC
change, or payload type change causes codec re-initialization based on the
codec parameters in the latest SDP send by this end point. Such change is
triggered by the packets after the jitter buffer, so all the packets using
the previous codec profile for the particular PT should be processed and
out of the system. If any of those packet do arrive late, they will be
discarded either by jitter buffer as late packets or based on some sort of
probation counter which starts on unexpected SSRC, remote address, or
payload change.

Regards,
_____________
Roman Shpount