Setting up QOS for WebRTC

[lindend]

Use of WebRTC by apps is increasing so I'd like to setup QOS for it in Merlin on my N66U. Problem is, I don't think it has a predefined port (or port range). Anyone have QOS configured for WebRTC on their ASUS device? If not, any ideas on how it could be setup?

 

[lindend]

Bump...any ideas on this one?

 

[mromero]

The ASUS QOS is very basic, almost primitive compared to say Tomato.

Enabling QOS has the unwanted side effect of disabling hardware acceleration (CTF) on the router.

If you need good QOS I think you need move up to Toastman Tomato, DD-WRT, or a dedicated box running pFSense where you can do really powerful traffic shaping.

I would like to know why Asus has not addressed this very serious shortcoming in the firmware by now.

 

[PrivateJoker]

In my limited view, I feel like I see more people solving problems on their consumer routers by disabling, and not enabling QOS.

I also think that, in part due to marketing, that many consumers are left with the sense that QOS has preeminent importance on their network, even over the quality & transfer speed of their wireless connection and even WAN connection.

Everyone is doing video streaming to the home, and that is one activity that is pretty sensitive to an uninterrupted stream of bandwidth to your premise and then into your device. Look at just FaceTime usage (which most people do over wifi), a lot of people who have never had a video and/or audio call over IP, and don't use skype, suddenly are making hundreds of thousands of these calls globally each day, on their existing wifi equipment (often using devices with downright subpar wifi radios & antenna).

FaceTime uses SIP, RTP, & SRTP (among other standards), and there is some overlap between it and the standards used by WebRTC. I'd wager that an insignificant percentage of those FaceTime calls are aided by using QOS tweaked for SIP.

I'm also unclear what you're expecting out of a firmware or a router for a spec that is exclusively using browser based standards to operate.

http://dev.w3.org/2011/webrtc/editor/webrtc.html

So, IMHO, if our routers are good enough to do stuff like airplay of local audio & video, FaceTime calls, do Google Hangouts with multi-party audio and/or video and screencast streaming, and stream HD Netflix (all without tweaking even one single setting), then a new set of standards to do browser to browser direct communication should be able to do so without any problem. My biggest concern of how I want my router to deal with WebRTC would be a focus on strict security. I don't want to have to patch any zero day exploits on my router's f/w.