Re: [rtcweb] Question about JSEP createOffer

Roman Shpount <roman@telurix.com> Thu, 10 May 2012 16:54 UTC

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Date: Thu, 10 May 2012 12:53:58 -0400
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From: Roman Shpount <roman@telurix.com>
To: Justin Uberti <juberti@google.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Question about JSEP createOffer
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On Thu, May 10, 2012 at 9:48 AM, Justin Uberti <juberti@google.com> wrote:

>
> ****
>>
>> ** **
>>
>> I agree Roman's case of a B2BUA seems like it would need a full offer,
>> although the idea of connecting an existing call to a new destination seems
>> like an uncommon case; the application could simply start a new call (i.e.
>> create a new PeerConnection), right?****
>>
>> ** **
>>
>> If we decide we need to support this, the simplest way of handling this
>> would be to simply extend the |hints| parameter in createOffer to have some
>> sort of "complete" flag.****
>>
>> <partha> your API proposal looks good  as it is possible to re-use the
>> existing or offer full based on the application needs. But the application
>> developer  MUST aware of the implication as it calls for interop issue in
>> case of federation scenario if the full offer is not sent across. </partha>
>>
>
> The application developer only needs to do this if supporting these edge
> cases is relevant to their application.
>
>> ****
>>
>>
If you inter-operate with anything SIP, you need to do a full new offer
every time you get a re-INVITE without a body. The reason for this is you
are working with an external system. Thus you cannot guess if this
re-INVITE was generated by an existing call or some sort of third party
call control application which will need a full offer. This is not
something that is specific to SIP only. Every time you will implement any
type of interconnect between disjoint systems you will deal with a similar
problem.
_____________
Roman Shpount