Le lundi 29 juin 2015 21:46:25 UTC+2, Adam Roach a écrit : > I think there's some confusion here. > > Ethan and "Daasoundz DJ" are talking about Hello. > > Byron and Randell are talking about WebRTC more generally. > > Daasoundz: The confusion probably arises from the fact that you say > you're testing "your webapp" and that you're "using Firefox Hello" to do > so. I have no idea how you plan to make this work, but the current > design of Hello doesn't really contemplate interacting with non-Hello > endpoints, and I can think of a large handful of problems you're going > to run into in trying to do so. (To be clear, there is still a desire -- > at least, among some people -- to eventually federate Hello with other > clients, but that's way beyond our planning horizon for now.) > > I think you need to give a more detailed explanation of what you're > trying to accomplish before anyone on this list can provide you with > useful answers. > > /a > > On 6/29/15 09:37, Randell Jesup wrote: > > On 6/29/2015 2:38 AM, Daasoundz DJ wrote: > >> Le vendredi 26 juin 2015 17:21:11 UTC+2, Byron Campen a écrit : > >>> You can also rewrite the local SDP to alter the priority. > > > >> It's an idiot question but ... how can I rewrite the local SDP ? > >> and/or where do I find it ? > > > > After you get the offer (from createOffer) or receive it (from the > > other end), modify the m=audio line in offer.sdp to put the payload > > number in the a=rtpmap line for G722 before all other audio codecs. > > > > i.e. > > ... > > m=audio <number> <transport> <list of payload values> > > ... > > a=rtpmap:<payload> G722/8000/1 (note: the /1 is optional) > > <other a=rtpmap's> > > > > You want <payload> to be the first in the list of payload values in > > the m=audio line. > > > > It's all just (careful) string manipulation. Don't assume any > > particular number will be used for <payload>! > > > > > -- > Adam Roach > Principal Platform Engineer > [email protected] > +1 650 903 0800 x863
I'm using 2 officials webRTC-app : Hello Firefox, apprtc And 2 homemade webrtc app from a traineeship then another one developped by colleagues. The 2 home-made's app have the same problem : When I fiddle de resolution I don't have audio. And when I don't change de resolution, the audio works. I have to make subjective tests to compare VP8 vs. H.264 & opus vs. G.722 (or G.711) with different kinds of profils : - injecting 1080p / 720p / VGA contents (with audio) without a Netdisturb (to add some lags, jitter...) with VP8 & opus, - Same operation with H.264 & G.722, - Same operation with a Netdisturb using VP8 & opus, - Same operation with a Netdisturb using H.264 & G.722. All these op' are recorded for subjective tests. I wanted to use Firefox Hello for H.264. but I can't manage the resolution and I don't really know how to force G.722. That's why I'm asking how to do it. My colleagues are working on a new version of the home-made webrtc app to force the use of H.264 & G.722 and they fixed the bug of the audio. So at the moment, I'm waiting if it's ok or not. _______________________________________________ dev-media mailing list [email protected] https://lists.mozilla.org/listinfo/dev-media
