Re: [rtcweb] Comments on draft-ietf-rtcweb-stun-consent-freshness-00

["Ram Mohan R (rmohanr)" <[e-mail]>]

Hi Magnus,

Thanks for your comments. Please see inline for responses.


-----Original Message-----
From: Magnus Westerlund <[e-mail]>
Date: Tuesday, 12 November 2013 8:28 PM
To: "[e-mail]"
<[e-mail]>,
"[e-mail]" <[e-mail]>
Subject: Comments on draft-ietf-rtcweb-stun-consent-freshness-00
Resent-From: <[e-mail]>
Resent-To: <[e-mail]>, <[e-mail]>, Cisco Employee
<[e-mail]>, <[e-mail]>
Resent-Date: Tuesday, 12 November 2013 8:27 PM

>Hi,
>
>Here are my review comments on draft-ietf-rtcweb-stun-consent-freshness-00
>
>1. General:
>I understood one reason for splitting this out in its own document is
>that we might have other future users of this mechanism. Thus I think it
>should be written to be less WebRTC specific. It is appropriate to use
>WebRTC as motivation for requirements etc, but the actual description of
>the mechanism should be done in a neutral way.

Ok. We will add text in Abstract/Introduction to reflect this point.

>
>2. Section 1.
>   When a session is first established, WebRTC implementations are
>   required to perform STUN connectivity checks as part of ICE
>   [RFC5245].  That initial consent is not described further in this
>   document.
>
>When generalizing this, I think it is important to make clear that it is
>assumed that ICE is being used for that initial consent.


Yes. We will add text to make it clear.

>
>3. Section 1:
>"This document describes a new STUN usage with a request and response
>   which verifies the remote peer consents to receive traffic, and
>   detects loss of liveness."
>
>I am missing a word after "request and response", transaction or
>messages maybe?

NEW:
This document describes a new STUN usage with a request and response
messages which verifies the remote peer consents to receive traffic, and
detects loss of liveness.



>
>4. Section 3:
> A
>   response is necessary both for consent to continue sending traffic,
>   as well as to ensure connectivity.
>
>Is verify rather than "ensure" better?

Agree will replace with "verify"

>
>5. Section 4:
>   Consent freshness serves as a circuit breaker (so that if the path or
>   remote peer fails the WebRTC browser stops sending all traffic on
>   that remote peer), determining session liveness serves the purpose of
>   notifying the application of connectivity failure so that the
>   application can take appropriate action.
>
>traffic on/traffic to

Thanks. will fix this.


>
>What is the definition of a peer here?

Peer is the sink for the traffic originated from the sender for that flow
(5 tuple). If a browser uses different 5 tuple for each stream(Audio,
video, data) it sends then it should do consent for each flow. If it uses
same 5-tuple (mux case) then there will be only one consent.



>What scope does the stop sending
>have?

The stream that is being sent on a flow(5 tuple) for which a consent has
failed will be stopped. If all the streams are muxed on a same 5 tuple the
entire traffic will be stopped.

>
>6. Section 4:
>   1.  A consent timer, Tc, whose value is determined by the browser.
>       This value MUST be 15 seconds.
>
>I see a contradiction here, should it be determined by the browser or be
>15 s? If it is 15, can you please motivate why it is this value, or
>point to where it is motivated?

The default value of 15 was some thing that EKR and Justin gave us based
on the implementation/testing and fine tuning they have done on their
browsers. I agree we can have some text around it that explains how this
number is arrived. We will add the same.


>
>7. Section 4:
>   If a valid STUN Binding Response is received, the consent timer is
>   reset and fires again Tc seconds later.
>
>Is it reset to the time of receiving it or the time of sending the
>initial request this is the response for?

Consent timer is reset to the time of receiving it


>
>8. Section 4:
>   If a valid STUN Binding Response is not received after 500ms, the
>   STUN Binding Request is retransmitted (with the same Transaction ID
>   and all other fields).  As long as a valid STUN Binding Response is
>   not received, this retransmission is repeated every 500ms until Tf
>   seconds have elapsed or a valid response is received.
>
>So no exponential back-off on the retransmission timer. I think that is
>fine. However, I think it do require you to expand a bit on what happens
>when one gets multiple response on the same Transaction ID. For example
>additional responses do they or do they not reset the Tc timer?

Additional responses MUST reset the Tc timer.


>
>Then I wonder about the cases where the RTT is above 500 ms, should one
>then scale this factor to be once per RTT?
>
>9. Section 4:
>"with the same Transaction ID"
>
>Why not use a new transaction ID for each sent packet? It would allow
>tracking loss and determine RTT more reliable. All for the small cost of
>keeping track of slightly more Transaction IDs?

Yes, new transaction ID will help to determine the packet loss and RTT.



>
>10. Section 4.
>   Liveness timer: If no packets have been received on the local port in
>   Tr seconds, the WebRTC browser MUST inform the JavaScript that
>   connectivity has been lost.  The JavaScript application will use this
>   notification however it desires (e.g., cease transmitting to the
>   remote peer, provide a notification to the user, etc.).  Note the
>   definition of a received packet is liberal, and includes an SRTP
>   packet that fails authentication, a STUN Binding Request with an
>   invalid USERNAME or PASSWORD, or any other packet.
>
>I think this requires some considerations in regards to RTT. If the RTT
>is 700 ms, high for a real-time app but not unheard of at all. Then a Tr
>= 1 second would fire on a single loss.

Ok.  will start a discussion for this item in a separate email.


Regards,
Ram


>
>Cheers
>
>Magnus Westerlund
>
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