Re: [rtcweb] SIP MUST NOT be used in browser?

"Ravindran Parthasarathi" <pravindran@sonusnet.com> Tue, 13 September 2011 16:43 UTC

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Thread-Topic: [rtcweb] SIP MUST NOT be used in browser?
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From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
To: "Olle E. Johansson" <oej@edvina.net>, "Tim Panton" <tim@phonefromhere.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SIP MUST NOT be used in browser?
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Tim,

Thanks for pointing it out and my example is wrong. 

My point is that RTCWeb1.0 deliverables is not a gating factor for
Javascript based SIP Stack or other signaling protocol using Javascript.
Also, RTCWeb signaling protocol selection is important for interop. 

Thanks
Partha

>-----Original Message-----
>From: Olle E. Johansson [mailto:oej@edvina.net]
>Sent: Tuesday, September 13, 2011 7:16 PM
>To: Tim Panton
>Cc: Ravindran Parthasarathi; rtcweb@ietf.org
>Subject: Re: [rtcweb] SIP MUST NOT be used in browser?
>
>
>13 sep 2011 kl. 12:11 skrev Tim Panton:
>
>> Minor nit - Phono isn't  a javascript SIP Stack. It is a javascript
>XMPP stack.
>> The server-side gateways out to SIP etc.
>
>A major thing is that you call it a minor nit. That is an important
>observation for the specs.
>
>/O
>>
>> Tim.
>> On 11 Sep 2011, at 21:13, Ravindran Parthasarathi wrote:
>>
>>> Hi Aaron,
>>>
>>> Javascript SIP stacks (Ex: phono.com) exists already and RTCWeb1.0
is
>>> not a gating factor for those development. My worry is that
RTCWeb1.0
>is
>>> standardized and then identify the gap in signaling which is not a
>good
>>> protocol design. It is better to discuss with signaling rather than
>just
>>> solving media protocol requirement alone. In case any implementation
>>> deployed, the backward compatibility has to be provided till the end
>of
>>> the product and RTCWeb1.0 is a not an exception.
>>>
>>> For the time factor concern, let us work for the quick closer and I
>have
>>> no disagreement there. But I have problem in case it is mentioned as
>the
>>> issues will not be solved to meet the WG deadline.
>>>
>>> Thanks
>>> Partha
>>>
>>>> -----Original Message-----
>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> Behalf
>>>> Of Aaron Clauson
>>>> Sent: Friday, September 09, 2011 7:26 PM
>>>> To: rtcweb@ietf.org
>>>> Subject: Re: [rtcweb] SIP MUST NOT be used in browser?
>>>>
>>>> Another 2 cents from a SIP person.
>>>>
>>>> I think attempting to incorporate SIP (or Jingle et al) into RTCWeb
>>>> would be
>>>> a bad idea for the reason that it would significantly slow down and
>>>> complicate the standard. If SIP is included in RTCWeb then there
>will
>>>> need
>>>> to be a discussion, already emerging here, about which parts of SIP
>to
>>>> include and all the other intricacies of SIP; transports, sips vs
>sip,
>>>> request routing etc etc.
>>>>
>>>> Writing a javascript SIP stack is a small task compared to getting
>the
>>>> RTCWeb media capabilities built into browsers. As soon as the first
>>>> browser
>>>> appears that supports RTP then javascript SIP stacks will start
>popping
>>>> up
>>>> all over the place.
>>>>
>>>> I for one would love to be able to process calls in my browser and
>to
>>> be
>>>> able to do it yesterday. Slowing the RTCWeb process down for
>something
>>>> that
>>>> will take care of itself anyway, namely readily available
javascript
>>>> signalling libraries, would be a shame.
>>>>
>>>> Aaron
>>>>
>>>> _______________________________________________
>>>> 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
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>---
>* Olle E Johansson - oej@edvina.net
>* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
>
>