Hi Iñaki,
  Thanks for the quick reply. Please find responses inline.

On 04/16/2013 04:40 AM, Iñaki Baz Castillo wrote:
> Thanks a lot for your review Suresh. Please read replies inline:
> 
> 
> 2013/4/16 Suresh Krishnan <[email protected]>:
>> Summary: This draft is almost ready for publication as a Proposed
>> Standard, but has some minor issues that need to be fixed.
>>
>> * Section 4.1
>>
>> There is no error handling specified. i.e. What happens if the server
>> does not send a 101 reply with sip in the Sec-WebSocket-Protocol header?
> 
> It should be considered a SIP transport error, so a 503 SIP response
> should be locally generated. This will be added to that section.

Great.

> 
> 
> 
>> * Section 5.2
>>
>> The updated ABNF provided in section 5.2.1. is confusing. e.g. RFC4168
>> specifies the entire transport rule as follows
>>
>> transport         =  "UDP" / "TCP" / "TLS" / "SCTP" / "TLS-SCTP"
>>                         / other-transport
>>
>> while this document just uses an incremental alternative and calls it an
>> updated rule. I would have expected an *updated* rule to say
>>
>> transport         =  "UDP" / "TCP" / "TLS" / "SCTP" / "TLS-SCTP" / "WS"
>>                         / "WSS" / other-transport
> 
> There was a long discussion in SIPCORE WG about this subject and
> finally it was chosen to use the incremental alternative (so the
> updated ABNF would be also longer in the future when new SIP
> transports are defined).
> 
> Do you consider anyhow that using the fully updated rule is required?

I saw Robert's response on the same topic and I will reply to that. If
the WG's consensus is to go incremental, please stick with that. I just
want to ensure that there is no ambiguity as to the final result.

> 
> 
> 
>> * Section 5.2.2
>>
>> RFC4168 does not update transport-param. Is this a copy and paste error?
> 
> Yes, will be fixed.

OK.

> 
> 
> 
>> * Section 5.2.4
>>
>> Maybe it is just me, but the way I read it, this section leaves the
>> client without any mandatory to implement transport protocol for SIP
>> since it relaxes the need to implement UDP and TCP, as well as allowing
>> the implementation of the websocket transport using a MAY.
> 
> Yes, such a MAY sounds a look confusing. What the text wants to say is
> that a SIP WebSocket Client (which obviously implements the SIP
> WebSocket transport) is not mandated to implement UDP and TCP
> transports. I wil improve the text by removing the last words and
> leave it as follows:
> 
> ----------------
>    The specification of this transport enables SIP to be used as a
>    session establishment protocol in scenarios where none of other
>    transport protocols defined for SIP can be used.  Since some
>    environments do not enable SIP elements to use UDP and TCP as SIP
>    transport protocols, a SIP element acting as a SIP WebSocket Client
>    is not mandated to implement support of UDP and TCP.
> ----------------

Looks good.

Thanks
Suresh

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