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

[Joshua Colp]

Lorenzo Miniero wrote:
> 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.

I don't think the ast_frame needs to be extended, but merely a new frame 
type added that has a payload defined for this purpose. Right now 
reading RTCP data returns a null frame. It can be changed to return this 
new frame with the data. The ast_read function can recognize this frame 
type and invoke what ever logic may be required. This can include giving 
this data to the translation path (implementations can be changed to 
explicitly define that they support it). This also does not alter the 
threading model and gives a guarantee that the codec won't have to 
protect itself, like would need to be done for a stasis message.

>
> 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 :-)

Personally I'm not a huge fan of using the stasis messages for exactly 
the reason you've given in your follow-up email. There's no obvious 
reliable data to key off of and it also requires synchronization between 
the thread handling the message and the thread doing the transcoding. 
This changes the threading model for transcoding enough to be of a 
concern to me and also introduces another lock.

>
> 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.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org