Re: Help calculating bandwidth cost for TURN server

[Vikas]

Answers inline

On Mar 18, 9:58 pm, Rene Tapia <rta...@nuxlabs.com> wrote:
> Hello,
>
> I have a free dating site, and I already have instant messaging in my site
> using XMPP. I was extremely happy when I learned about WebRTC, because it
> fits perfectly into my site. XMPP serves perfectly as a signaling channel
> for WebRTC, and the possibility for peers to connect directly without
> incurring on high server bandwidth costs for voice/video calls, makes it
> very attractive for free sites like my own.
>
> I am almost done with the integration, however, from my testing so far, it
> seems that most likely I am going to have to deploy my own TURN server, and
> I am a bit concerned about the bandwidth cost of deploying my own TURN
> server, and so I would like to ask a few questions in order to make a rough
> estimate of the bandwidth cost.
>
> For now, I am not going to add video for a couple of reasons: 1) video for
> dating sites is problematic, and 2) video bandwidth (and therefore cost) is
> much higher than audio.
>
> In regards to audio bandwidth:
>
> 1) What is the bitrate of opus/48000?
>
> From Chrome's tracing, it seems like the bitrate for opus (as configured in
> the apprtc demo) is 64000 bps (i.e. Selected voice codec opus/48000/2
> (111), bitrate=64000). However, the 2 after 48000, I guess means 2
> channels, so is the actual bit rate then 128000? On the other hand, I
> believe opus uses variable bit rate, and so is the actual expected bitrate
> below 128000? In summary, for opus opus/48000/2 as selected above from the
> code in the apprtc demo, what would be the actual expected bitrate?
>
As i understand, the bitrate for opus would be 64000 for 2 channels.

> 2) What is the bitrate of isac/16000?
>
> If I comment out the call to preferOpus() in the code from the apprtc demo,
> then isac/16000 gets selected by default (i.e. Selected voice codec
> ISAC/16000/1 (103), bitrate=-1). For ISAC/16000/1 as selected above from
> the code in the apprtc demo, is the actual expected bitrate 16000 bps?
>
I think iSAC is adaptive bitrate if bitrate = -1.

> 3) For voice calls, which codec would you recommend, if you want to
> minimize bandwidth usage, but still have an audible voice call?
>
Well it's a good question, i think you can refer to codec bit rate
comparisons ( http://www.opus-codec.org/comparison/ ).

> In regards to direct p2p connections:
>
> 4) In a typical residential scenario, approximately what percentage of
> connections will be direct p2p vs TURN server relayed connections?
> The scenario for my site involves a desktop/laptop behind a DSL/cable
> modem, and so it is basically a typical residential scenario. Do you have a
> rough estimate, or at a rough guess, what percentage of connections will
> succeed as direct p2p connections (as opposed to TURN server relayed
> connections)?
>
"Close enough" answer is 10% relayed. In our actual deployment it was
closer to 15%, but I think it depends a lot on the target user type
(e.g. home or corporate).

> Thank you in advance.
>
> Regards,
> Rene Tapia

[Rene Tapia]

Thank you Vikas, this is very helpful information. And also thanks and congratulations to the whole webrtc team for all your great work, and for this fantastic technology.

Cheers