Re: [rtcweb] Use cases - recording and voicemail

Harald Alvestrand <harald@alvestrand.no> Mon, 22 August 2011 11:09 UTC

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Date: Mon, 22 Aug 2011 13:10:19 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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On 08/22/11 12:17, Avasarala, Ranjit wrote:
> Hi Partha
>
> I agree with Paul that we could use the recording mechanism discussed in siprec mailing list could be added here as webbrowser would essentially be a SIP UA which could initiate and terminate SIP sessions.
mandatory standard clarification (?):

.. as somewhere in the combination of the Web browser, the Javascript 
and the Web server it connects to, there would be something that 
performs the functions of a SIP UA which could initiate and terminate 
SIP sessions ...
> Regards
> Ranjit
>
>
> -----Original Message-----
> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
> Sent: Monday, August 22, 2011 1:46 PM
> To: Avasarala, Ranjit; Elwell, John; Stefan Håkansson LK; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: RE: [rtcweb] Use cases - recording and voicemail
>
> Hi Ranjit,
>
> I have mentioned as JavaScript API because there is no specific dependency on browser. Let us discuss about the exact mechanism later. Currently, let us have common understanding whether recording usecase has to be added in RTCWeb or not.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: Avasarala, Ranjit [mailto:Ranjit.Avasarala@Polycom.com]
>> Sent: Monday, August 22, 2011 12:22 PM
>> To: Ravindran Parthasarathi; Elwell, John; Stefan Håkansson LK;
>> rtcweb@ietf.org; public-webrtc@w3.org
>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>
>> Hi
>>
>> We could use websockets protocol to pass metadata information.
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Ravindran Parthasarathi
>> Sent: Monday, August 22, 2011 12:12 PM
>> To: Elwell, John; Stefan Håkansson LK; rtcweb@ietf.org; public-
>> webrtc@w3.org
>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>
>> John,
>>
>> I agree with you. JavaScript API should have the provision to pass the
>> metadata.
>>
>> Thanks
>> Partha
>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>> Sent: Monday, August 22, 2011 11:59 AM
>>> To: Ravindran Parthasarathi; Stefan Håkansson LK; rtcweb@ietf.org;
>>> public-webrtc@w3.org
>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>
>>> Partha,
>>>
>>> You are talking here about the metadata, I think. I assume the web page
>>> / JavaScript has to deal with that - not the browser.
>>>
>>> John
>>>
>>>
>>>> -----Original Message-----
>>>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>>>> Sent: 19 August 2011 18:19
>>>> To: Stefan Håkansson LK; Elwell, John; rtcweb@ietf.org;
>>>> public-webrtc@w3.org
>>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>>
>>>> Stefan,
>>>>
>>>> In case recording similar to SIPREC, it is little bit more
>>>> than spanning two media (RTP stream) alone because recording
>>>> has to include some context data about recording apart from
>>>> the media stream.
>>>>
>>>> Thanks
>>>> Partha
>>>>
>>>>> -----Original Message-----
>>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>>> Behalf Of Stefan Håkansson LK
>>>>> Sent: Friday, August 19, 2011 8:26 PM
>>>>> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>>>>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>>>>
>>>>>> However, I did suggest (in other text in my previous
>>>> message) that one
>>>>> possible solution might be to record locally and use a
>>>> second RTC-Web
>>>>> session to transmit from the local file to the>remote
>>>> recorder. What I
>>>>> failed to say was that in this case the local file would be
>>>> a temporary
>>>>> repository - just a buffer between the two sessions.
>>>>> This makes sense. Also, if you look at the API proposals
>>>> available, it
>>>>> would be quite easy to forward (in real time) a stream
>>>> being received
>>>>> to another entity. There is no explicit recording, a stream being
>>>>> received (via RTP) is just streamed to another entity (via
>>>> a separate
>>>>> RTC-Web session). I think this would solve this case.
>>>>>
>>>>> Stefan
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
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