Re: [rtcweb] JSEP Walk Through on Wednesday the 18th of September at 17.00 CEST (8 AM PDT)

[Christer Holmberg <[e-mail]>]

Hi,

I may not be able to attend the phone meeting today, so I send my input by the e-mail.

As requested by the WG chairs, I’ll focus on clarification issues/questions, and will for now not repeat the issues I have with pranswer and forking (also, we may have found a way forward to solve that).



Q_1:      Meaning of “session”:

When the document talks about “session”, I think it needs to be more clear that the scope of a session is a PeerConnection.

For example, as described in section 3.5, in the case of forking there might be multiple PeerConnections, each forming different “sessions” – eventhough they will all be part of e.g. the same higher-level SIP session.

Similar, when the text talks about “initial Offer”, it needs to be clear that it refers to a given PeerConnection.



Q_2:      Forking

I would suggest some re-wording (I am willing to provide text) of the forking sections. At least the following I want more clear:


1.       Similar to parallel forking, sequential forking can also be implemented using multiple PeerConnections.


2.       When a new PeerConnection is created, it will have different address properties than the “mother” PeerConnection, so a new Offer needs to be created immediately, in order to inform the remote peer about those properties. The draft DOES say that the same local media streams are added to the new PeerConnection, but it does not make it clear that the new PeerConnection will have different address properties, and possibly also other properties that vary. The draft also DOES talk about the “UPDATE approach”, but that is very unclear (see Q_3).



3.       Parallel forking does not necessarily mean you receive media simultaneously from multiple endpoints. You might establish a media connection (e.g. using SIP preconditions) with multiple endpoints, but that does not mean you will receive media from all of them at the same time. I can provide some 3GPP call flows that show examples of that.


Q_3:      UPDATE approach

I don’t understand the text talking about the “UPDATE approach” and “UPDATE strategy”, referring to RFC 3960. With my SIP background, I kind of understand what you’re thinking is, but I think non-SIP people will have very difficult to understand exactly what you mean. I think it is much better to simply say that, for the new PeerConnection, an Offer needs to be sent (see Q_2).


Q_4:      Appendix A.2.

As the example contains BUNDLE, I would also like to see “2nd Offer” (with identical port numbers), for completeness.


Q_5:      Appendix Forking

I would like to see some examples with forking.


Regards,

Christer










From: [e-mail] [mailto:[e-mail]] On Behalf Of Justin Uberti
Sent: 18. syyskuuta 2013 1:29
To: Magnus Westerlund
Cc: [e-mail]
Subject: Re: [rtcweb] JSEP Walk Through on Wednesday the 18th of September at 17.00 CEST (8 AM PDT)

The draft has now been posted at http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-04.txt, with 13 new pages of text.

Readers should focus on Section 5, which specifies the behavior of createOffer/createAnswer, as well as Appendix A.2, which provides examples of the generated SDP.

On Tue, Sep 17, 2013 at 1:51 AM, Magnus Westerlund <[e-mail]<mailto:[e-mail]>> wrote:
On 2013-09-17 10:47, Harald Alvestrand wrote:
> Magnus,
>
> what is the expected duration of the call?
2 hour maximum

Magnus

>
> On 09/17/2013 10:38 AM, Magnus Westerlund wrote:
>> WG,
>>
>> I know that the new JSEP Draft is not yet submitted. I am hoping that it
>> will show up during the morning hours for the US based people. That way
>> everyone gets at least some few hours to look at the draft prior to the
>> call. If it hasn't we will postpone the call, but I really hope we can
>> avoid this. I will send a notice to the WG if we have to postpone it.
>>
>> Cheers
>>
>> Magnus
>>
>> On 2013-09-11 07:58, Magnus Westerlund wrote:
>>> WG,
>>>
>>> There will be a new version of JSEP draft coming out Monday. As a follow
>>> up to get as much progress on this topic as possible. The chairs and
>>> authors are offering a informal walk through session on Wednesday the
>>> 18th. People will be strongly recommended to have reviewed the new
>>> draft.
>>>
>>> The intention of the session is enable the authors to inform you about
>>> the latest changes, any issues to consider and what needs further
>>> discussion. The will also enable you as participant after your review of
>>> the document to ask clarifying questions or for motivations behind
>>> choices, and raise issues you have spotted.
>>>
>>> The session is not intended to solve issues or judge consensus on part
>>> of the draft. Discussions of what is the best solution etc, will to be
>>> cut off.
>>>
>>> This will be a webex conference per details below.
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>>
>>> Topic: RTCWeb - JSEP
>>> Date: Wednesday, September 18, 2013
>>> Time: 8:00 am, Pacific Daylight Time (San Francisco, GMT-07:00)
>>> Meeting Number: 302 061 238
>>> Meeting Password: ietf
>>>
>>> -------------------------------------------------------
>>> To join the online meeting (Now from mobile devices!)
>>> -------------------------------------------------------
>>> 1. Go to
>>> https://cisco.webex.com/ciscosales/j.php?ED=206400898&UID=0&PW=NYjk4NmE0ODBj&RT=MiM0
>>>
>>>
>>> 2. Enter your name and email address.
>>> 3. Enter the meeting password: ietf
>>> 4. Click "Join Now".
>>>
>>> To view in other time zones or languages, please click the link:
>>> https://cisco.webex.com/ciscosales/j.php?ED=206400898&UID=0&PW=NYjk4NmE0ODBj&ORT=MiM0
>>>
>>>
>>>
>>> -------------------------------------------------------
>>> To join the teleconference only
>>> -------------------------------------------------------
>>> 1. Dial into Cisco WebEx (view all Global Access Numbers at
>>> http://cisco.com/en/US/about/doing_business/conferencing/index.html
>>> 2. Follow the prompts to enter the Meeting Number (listed above) or
>>> Access Code followed by the # sign.
>>>
>>> San Jose, CA: +1.408.525.6800<tel:%2B1.408.525.6800> RTP: +1.919.392.3330<tel:%2B1.919.392.3330>
>>>
>>> US/Canada: +1.866.432.9903<tel:%2B1.866.432.9903> United Kingdom: +44.20.8824.0117<tel:%2B44.20.8824.0117>
>>>
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>>>
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>>>
>>>
>>> ----------------------------------------------------------------
>>> ALERT – PLEASE READ: DO NOT DIAL THE TOLL FREE NUMBERS FROM WITHIN THE
>>> (408) OR (919) AREA CODES
>>> ----------------------------------------------------------------
>>> Please dial the local access number for your area from the list below:
>>> - San Jose/Milpitas (408) area: 525-6800
>>> - RTP (919) area: 392-3330
>>>
>>> Dialing the WebEx toll free numbers from within 408 or 919 area codes is
>>> not enabled (non-Cisco phones). “ If you dial the toll-free numbers
>>> within the 408 or 919 area codes you will be instructed to hang up and
>>> dial the local access number.” Please use the call-back option whenever
>>> possible and otherwise dial local numbers only. The affected toll free
>>> numbers are: [phone] for the San Jose/Milpitas area and (866)
>>> 349-3520 for the RTP area.
>>>
>>
>
>
>

--

Magnus Westerlund

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