Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remote recording - RTC-Web client acting as SIPREC session recording client]

Matthew Kaufman <matthew.kaufman@skype.net> Tue, 06 September 2011 21:14 UTC

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Date: Tue, 06 Sep 2011 14:16:24 -0700
From: Matthew Kaufman <matthew.kaufman@skype.net>
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To: Ravindran Parthasarathi <pravindran@sonusnet.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remote recording - RTC-Web client acting as SIPREC session recording client]
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On 9/6/11 12:02 PM, Ravindran Parthasarathi wrote:
> Matthew,
>
> <snip>>>  And there's some security reasons that I'd rather you tunnel it
> over
>> HTTP rather than opening up your ability to send UDP packets to port
>> 5060.
>> SIP runs on TCP and TLS ports too.</snip>
> In case one port opening (80) will not create security issue but how
> opening two ports (80&  5060) will cause security issue? Could you
> please explain the security reason between port 80 for HTTP and 5060 for
> SIP

There has been a whole lot of work regarding what browsers do and don't 
do over HTTP. (Think: cross site access, what headers can be controlled 
from script, etc.) We would be forced to reinvent all of this for a 
reduced-functionality SIP in order to get the same protections. (Should 
a browser that runs Javascript loaded from www.evil.com be able to send 
SIP INVITE to addresses inside your firewall?)

And there is a whole lot of infrastructure that allows organizations to 
safely allow HTTP to pass in and out of their infrastructure.
>
> Please read inline for further comments.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: Olle E. Johansson [mailto:oej@edvina.net]
>> Sent: Wednesday, September 07, 2011 12:17 AM
>> To: Matthew Kaufman
>> Cc: Ravindran Parthasarathi; rtcweb@ietf.org
>> Subject: Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remote
>> recording - RTC-Web client acting as SIPREC session recording client]
>>
>>
>> 6 sep 2011 kl. 20:40 skrev Matthew Kaufman:
>>
>>> On 9/6/11 11:36 AM, Ravindran Parthasarathi wrote:
>>>> Matthew,
>>>>
>>>> Even in case of SIP, there is no need of standardization in case it
>> is within single webserver(skype). SIP supports x-header or proprietary
>> header to extend the way you want it for providing innovative
>> functionality.
>>> Not if SIP is baked in to the browser it doesn't. If the browser
>> implements a SIP phone in native code, I have no way of adding "x-
>> header" or anything else. I live with the functionality provided by the
>> browser.
>> That's an implementation detail. One can easily add an API call to add
>> headers on the outbound INVITE.
> [Partha] My understanding is same as olle. Javascript API MUST provide
> the mechanism to add proprietary SIP headers or else I agree with you
> that solution is useless.

Not nearly sufficient as I stated before. The problem is the semantics 
of all the existing SIP messages and headers, not whether or not you can 
add more.

Matthew Kaufman