[asterisk-dev] Strategies for handling RTCP feedback in codec modules

[Lorenzo Miniero]

Hi all,

apologies if this has been discussed before, but I couldn't find anything
in the recent months on this group so I thought I'd write anyway.

As a few others, I believe, I have been trying to find a way to make codec
modules more aware of what's happening on the wire. In particular, the
motivation for that comes from an attempt to make the open source Opus
codec module more responsive and adaptive to changes in the network, taking
advantage of the functionality the library provides in that respect (e.g.,
dynamic bitrate adaptation). The best approach to do that would obviously
be providing codec modules with info on the RTCP feedback loop, e.g., in
terms of losses the recipient has experienced, so that you can, for
instance, change the bitrate in the encoder. Unfortunately, as of now there
doesn't seem to be any way to make this possible, at least not in an easy
way, in Asterisk out of the box.
I've been investigating a few ways to do that, and have come up with some
possible approaches, that I first wanted to discuss with you guys though,
first of all to make sure I'm actually on the right path, and then to
evaluate whether or not any of those can actually be integrated within the
Asterisk code base (as I do believe such a feedback loop would be
beneficial to other codec modules as well, and not only Opus). If you're
interested in some more motivation behind this, you can read my discussion
with Alexander Traud in a comment to his fork to the asterisk-opus repo
here: https://github.com/traud/asterisk-opus/issues/3


For the sake of completeness, Alexander himself thought of a possible
approach for integrating this feedback in a comment to another post:
https://github.com/seanbright/asterisk-opus/issues/25#issuecomment-249420010
where the idea is to pass a reference of the ast_rtp_instance into the
codec module itself. While this could possibly do the trick, I don't
believe this would be a viable option, as it would break the architecture
and module relationships, but I thought I'd mention it anyway.

One possible option that I had thought about was extending ast_frame to
convey RTCP feedback to modules, along with media to translate. This would
allow such feedback to take the same "path" as media packets, meaning codec
modules wouldn't need to be aware of any additional core-related feature,
but only that sometimes they might receive control data instead of media to
translate. Anyway, this could probably be problematic to integrate with the
translator's architecture, and would probably need "cooperation" from
channel modules as well, so may be a bit overkill and bug-prone.

Another possible approach, and possibly the way to go, is to make use of
the Stasis message bus, something I was not aware of until I watched Matt's
excellent presentation at a conference recently. I saw how the RTP engine
in Asterisk does publish RTCP feecback on the bus, and that you can
subscribe to that as other events (as the HEP integration does, for
instance) from other parts of the code. I tried doing the same within the
Opus codec implementation, and apart from some quirks (e.g., weird fields
in report blocks, like negative source_ssrc) it seemed to do exactly what I
was looking for. The only problem, though, is that while a Stasis event
contains a whole lot of info, codec modules are pretty much clueless and
have no way of matching a specific event related to a specific call to a
translator context they're handling. In fact, AFAIK codec modules have no
visibility at all of the channel a translator is associated with, or of
other identifiers it could rely on. Thinking about this I did find a way to
implement some kind of loose mapping by extending the ast_frame structure
with two new properties, "ssrc" and "themssrc": basically, anytime an RTP
packet is received, the RTP engine copies the local and remote SSRC to the
frame before passing it to the core. When the first packet gets to the
codec module, it can keep track of them and save them locally to its own
internal struct. So, when a new event comes later from Stasis (e.g., an
RTCP RR), it can look at the SSRC it relates to and match it with the SSRCs
it is aware of, so that it knows it's related to a specific translator
context and react accordingly. While this seems effective, it has a few
issues, though. For instance, there's no way to assume remote SSRCs will be
unique, which means this could result in either missing or misleading
feedback in some cases, or even that they'll stay the same for the whole
call. Another aspect I haven't considered is the possible overhead,
although I don't think crawling a JSON object should take much resources.
Besides, I still haven't understood how Asterisk hashtables work, so this
part is still just theory :-)


That's basically it. Do you have any feeling/feedback on this? Is this
actually worth investigating, and do you believe I'm on the right track?
Any suggestion on how to make the mapping more effective in the codec
module? I'm of course willing to contribute to such an effort, if it's
deemed worthwhile.

Thanks,
Lorenzo
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