Re: [codec] #16: Multicast?

[stephen botzko <[e-mail]>]

Inline

On Wed, Apr 21, 2010 at 1:27 PM, Christian Hoene <[e-mail]>wrote:

>  Hi Stephen,
>
>
>
> not too bad. You answered faster than the mailing list distributes…
>
Not sure how that happened!

>
>
> Comments inline:
>
>
>
>
>
> *From:* stephen botzko [mailto:[e-mail]]
> *Sent:* Wednesday, April 21, 2010 7:10 PM
> *To:* Christian Hoene
> *Cc:* [e-mail]
>
> *Subject:* Re: [codec] #16: Multicast?
>
>
>
> I agree there are lots of use cases.
>
>
> Though I don't see why high quality has to be given up in order to be
> scalable.
>
> CH: These are just experiences from our lab. A spatial audio conference
> server including the acoustic 3D sound rendering needs a LOT of processing
> power. In the end, we have to remain realistic. Processing power is always
> limited thus if we need a lot then we cannot serve many clients.
>
> Also, I am not sure why you think central mixing is more scalable than
> multicast (or why you think it is lower quality either).
>
> CH: With multicast, you need N times 1:N multicast distribution trees
> (somewhat small tan O(n)=n²).  With central mixing you need N times 2
> transmission paths (O(n)=n). Also, this distributed mixing you need N times
> the mixing at each client. With centralized, you can live with one mixing
> for all (and some tricks for serving the talkers).
>
I agree you need more distribution trees for multicast if you allow every
site to talk. There is a corresponding benefit, since there is no central
choke point and also less bandwidth on shared WAN links.

In the distributed case,  you don't need an N-way mixer at each client, and
you also don't need to continuously receive payload on all N streams at each
client either.  In practice you can cap N at a relatively small number (in
the 3-8 range) no matter how large the conference gets.  In a large
conference, you can even choose to drop your comfort noise if you are
receiving two or more streams, and just send enough to keep your firewall
pinhole open.  This is all assuming a suitable voice activity measure in the
RTP packet.  Of course in the worst case, you will receive all N streams.


> Cheers,
>
>  Christian
>
>
> Stephen Botzko
>
> On Wed, Apr 21, 2010 at 12:58 PM, Christian Hoene <[e-mail]>
> wrote:
>
> Hi,
>
>
>
> the teleconferencing issue gets complex. I am trying to compile the
> different requirements that have been mentioned on this list.
>
> -          low complexity (with just one active speaker) vs. multiple
> speaker mixing vs. spatial audio/stereo mixing
>
> -          centralized vs. distributed
>
> -          few participants vs. hundreds of listeners and talkers
>
> -          individual distribution of audio streams vs. IP multicast or
> RTP group communication
>
> -          efficient encoding of multiple streams having the same content
> (but different quality).
>
> -           I bet I missed some.
>
>
>
> To make things easier, why not to split the teleconferencing scenario in
> two: High quality and Scalable?
>
>
>
> The high quality scenario, intended for a low number of users, could have
> features like
>
> -          Distributed processing and mixing
>
> -          High computational resources to support spatial audio mixing
> (at the receiver) and multiple encodings of the same audio stream at
> different qualities (at the sender)
>
> -          Enough bandwidth to allow direct N to N transmissions of audio
> streams (no multicast or group communication). This would be good for the
> latency, too.
>
>
>
> The scalable scenario is the opposite:
>
> -          Central processing and mixing for many participants .
>
> -          N to 1 and 1 to N communication using efficient distribution
> mechanisms (RTP group communication and IP multicast).
>
> -          Low complexity mixing of many using tricks like VAD, encoding
> at lowest rate to support many receivers having different paths, you name
> it...
>
>
>
> Then, we need not to compare apples with oranges all the time.
>
>
>
> Christian
>
>
>
> ---------------------------------------------------------------
>
> Dr.-Ing. Christian Hoene
>
> Interactive Communication Systems (ICS), University of Tübingen
>
> Sand 13, 72076 Tübingen, Germany, Phone [phone]
> http://www.net.uni-tuebingen.de/
>
>
>
> *From:* [e-mail] [mailto:[e-mail]] *On Behalf
> Of *stephen botzko
> *Sent:* Wednesday, April 21, 2010 4:34 PM
> *To:* Colin Perkins
> *Cc:* [e-mail]; [e-mail]
> *Subject:* Re: [codec] #16: Multicast?
>
>
>
> in-line
>
> Stephen Botzko
>
> On Wed, Apr 21, 2010 at 8:17 AM, Colin Perkins <[e-mail]> wrote:
>
> On 21 Apr 2010, at 12:20, Marshall Eubanks wrote:
>
> On Apr 21, 2010, at 6:48 AM, Colin Perkins wrote:
>
> On 21 Apr 2010, at 10:42, codec issue tracker wrote:
>
> #16: Multicast?
> ------------------------------------+----------------------------------
> Reporter:  hoene@…                 |       Owner:
>  Type:  enhancement             |      Status:  new
> Priority:  trivial                 |   Milestone:
> Component:  requirements            |     Version:
> Severity:  Active WG Document      |    Keywords:
> ------------------------------------+----------------------------------
> The question arrose whether the interactive CODEC MUST support multicast in
> addition to teleconferencing.
>
> On 04/13/2010 11:35 AM, Christian Hoene wrote:
>
>  P.S. On the same note, does anybody here cares about using this CODEC
> with multicast? Is there a single commercial multicast voice deployment?
> From what I've seen all multicast does is making IETF voice standards harder
> to understand or implement.
>
>
> I think that would be a mistake to ignore multicast - not because of
> multicast itself, but because of Xcast (RFC 5058) which is a promising
> technology to replace centralized conference bridges.
>
>
> Regarding multicast:
>
> I think we shall start at user requirements and scenarios. Teleconference
> (including mono or spatial audio) might be good starting point. Virtual
> environments like second live would require multicast communication, too. If
> the requirements of these scenarios are well understand, we can start to
> talk about potential solutions like IP multicast, Xcast or conference
> bridges.
>
>
>
> RTP is inherently a group communication protocol, and any codec designed
> for use with RTP should consider operation in various different types of
> group communication scenario (not just multicast). RFC 5117 is a good place
> to start when considering the different types of topology in which RTP is
> used, and the possible placement of mixing and switching functions which the
> codec will need to work with.
>
>
> It is not clear to me what supporting multicast would entail here. If this
> is a codec over RTP, then what is to stop it from being multicast ?
>
>
>
> Nothing. However group conferences implemented using multicast require end
> system mixing of potentially large numbers of active audio streams, whereas
> those implemented using conference bridges do the mixing in a single central
> location, and generally suppress all but one speaker. The differences in
> mixing and the number of simultaneous active streams that might be received
> potentially affect the design of the codec.
>
>
> Conference bridges with central mixing almost always mix multiple
> speakers.  As you add more streams into the mix, you reduce the chance of
> missing onset speech and interruptions, but raise the noise floor. So even
> if complexity is not a consideration, there is value in gating the mixer
> (instead of always doing a full mix-minus).
>
> More on point, compressed domain mixing and easy detection of VAD have both
> been advocated on these lists, and both simplify the large-scale mixing
> problem.
>
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
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