Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11

"Chenxin (Xin)" <> Tue, 24 September 2013 11:56 UTC

Return-Path: <>
Received: from localhost (localhost []) by (Postfix) with ESMTP id 1EE3F11E811A for <>; Tue, 24 Sep 2013 04:56:16 -0700 (PDT)
X-Virus-Scanned: amavisd-new at
X-Spam-Flag: NO
X-Spam-Score: -6.823
X-Spam-Status: No, score=-6.823 tagged_above=-999 required=5 tests=[AWL=-0.024, BAYES_00=-2.599, GB_I_INVITATION=-2, J_CHICKENPOX_44=0.6, J_CHICKENPOX_62=0.6, J_CHICKENPOX_93=0.6, RCVD_IN_DNSWL_MED=-4]
Received: from ([]) by localhost ( []) (amavisd-new, port 10024) with ESMTP id hmdqP8GM8j1O for <>; Tue, 24 Sep 2013 04:56:11 -0700 (PDT)
Received: from ( []) by (Postfix) with ESMTP id D0B4D11E8110 for <>; Tue, 24 Sep 2013 04:56:09 -0700 (PDT)
Received: from (EHLO ([]) by (MOS 4.3.5-GA FastPath queued) with ESMTP id AVU69040; Tue, 24 Sep 2013 11:56:08 +0000 (GMT)
Received: from ( by ( with Microsoft SMTP Server (TLS) id; Tue, 24 Sep 2013 12:54:57 +0100
Received: from ( by ( with Microsoft SMTP Server (TLS) id; Tue, 24 Sep 2013 12:55:41 +0100
Received: from ([]) by ([]) with mapi id 14.03.0146.000; Tue, 24 Sep 2013 19:55:35 +0800
From: "Chenxin (Xin)" <>
To: Karl Stahl <>, "" <>, "" <>
Thread-Topic: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
Thread-Index: AQHOtl/hYQwJQvwb8keiBCP8AmFFOZnUHYEwgABdJ7CAAEbaMA==
Date: Tue, 24 Sep 2013 11:55:34 +0000
Message-ID: <>
References: <> <> <> <07a601ceb64e$5caaba00$16002e00$> <07b001ceb65f$ce3f0cf0$6abd26d0$> <> <09d801ceb8f4$3b50dfd0$b1f29f70$>
In-Reply-To: <09d801ceb8f4$3b50dfd0$b1f29f70$>
Accept-Language: en-US, zh-CN
Content-Language: zh-CN
x-originating-ip: []
Content-Type: text/plain; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable
MIME-Version: 1.0
X-CFilter-Loop: Reflected
Subject: Re: [rtcweb] WGLC of draft-ietf-rtcweb-use-cases-and-requirements-11
X-Mailman-Version: 2.1.12
Precedence: list
List-Id: Real-Time Communication in WEB-browsers working group list <>
List-Unsubscribe: <>, <>
List-Archive: <>
List-Post: <>
List-Help: <>
List-Subscribe: <>, <>
X-List-Received-Date: Tue, 24 Sep 2013 11:56:16 -0000

Hi Karl,
>>While reading the draft-ietf-rtcweb-use-cases-and-requirements-11, here
>>are a few "telephony related" WebRTC things I think should be clarified
>>in the use cases.
>>3.2.1.  Simple Video Communication Service  Description ...
>>The invited user might accept or reject the session.
>>[Suggest adding] The invited user might accept only audio, rejecting
>>video (even if a camera is enabled). A user may also select to initiate
>>an audio session, without video.
>>And in API requirements:
>>   ----------------------------------------------------------------
>>   A1      The Web API must provide means for the application to ask the
>>browser for permission to use cameras and microphones, individually as
>>input devices. (One must be able to answer with voice only - declining
>>   ----------------------------------------------------------------
>>Same under
>>6.2.  Browser Considerations
>>The browser is expected to provide mechanisms for users to revise and
>>even completely revoke consent to use device resources such as camera
>>and microphone. [Suggest adding] Specifically, a user must be given the
>>opportunity to only accept audio in a video call invitation.
>[Xin] it is a common use case to accept only audio call and reject the video
>and quite useful. But I am doubt that this function should be mixed with
>video or audio device access permission . Do I misunderstand your proposal?
>I think we could just disable the video stream when signaling. So we could
>make video call with one and reject it with other in the same web-service. I
>think the audio and video device access permission is not for each call(peer
>   Xin
>[Karl] Try using a WebRTC application with Chrome and you will see: The
>Permission/Allowance to use Camera and Microphone comes up at a bar at the
>top of the browser window and is the actual answering of a call.

[Xin] yes, it come up because invoking the getUserMedia API. When we write the webrtc app, we could call getUserMedia API just once and use the same mediaStream later. The webrtc app could decide to send the video stream to the other side or not by configure the signaling(SDP or other).  It is trivial to click the Permission bar at the top every time when I get a call, even terrible when join a p2p conference. 
That is the reason I think the use case you mentioned should not mix with permission, which should be a signaling configuration problem. Now in Chrome,we could control it by using creatOffer or createAnswer and setting the OfferToReceiveVideo constraint.