How to force Hello Firefox to chose H.264 instead of VP8?

[Daasoundz DJ]

Hi,

Last week I posted a request to understand what went wrong with the webrtc-app I use. I didn't how to correct it. So I'd like to use Firefox Hello for my tests.
When I look into SDP packet, I can see VP8 & H.264 (and G.722 & opus), but selected coders are VP8 & opus anytime.
I'd like to know if it's possible to force Firefox Hello to use H.264 & G.722 only.
Is this possible ?

thanks in advance

[Ethan Hugg]

You can make it prefer H264 by adding the about:config parameter "media.
navigator.video.preferred_codec" as an integer type and setting its value
to 126. That will make Hello and others sites like it negotiate H264
provided your OpenH264 plugin has downloaded of course. I don't have a
solution for you for G.722.

-EH


On Fri, Jun 26, 2015 at 12:27 AM, Daasoundz DJ <daaso...@gmail.com>
wrote:

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> dev-media mailing list
> dev-...@lists.mozilla.org
> https://lists.mozilla.org/listinfo/dev-media
>

[Byron Campen]

You can also rewrite the local SDP to alter the priority.

Best regards,
Byron Campen

[Daasoundz DJ]

Hi,

Thank you for this answer. I tried it and it works perfectly.
Is there no parameter for the G.722 codec ?

[Daasoundz DJ]

It's an idiot question but ... how can I rewrite the local SDP ? and/or where do I find it ?

[Randell Jesup]

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>!

--
Randell Jesup, Mozilla

[Adam Roach]

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
a...@mozilla.com
+1 650 903 0800 x863

[Daasoundz DJ]

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.

[Daasoundz DJ]

Le lundi 29 juin 2015 21:46:25 UTC+2, Adam Roach a écrit :

I've forgot the fact that I'm using the same webapp on both side of the communication. I'm not trying to use Hello Firefox on th one hand then apprtc (or something else) on the other hand. I know it can't work :).

[Alexander Abagian]

> I'd like to know if it's possible to force Firefox Hello to use H.264 & G.722 only.

To use H.264 & G.722 only you could modify remote SDP rtpmap, deleting undesirable codec lines.