Re: [rtcweb] Requiring ICE for RTC calls

Neither Google Voice nor Skype (both fairly popular services) send raw RTP
directly from the client to a PSTN terminator. I can't speak for Skype, but
Google Voice uses a media gateway for the quality-related reasons I
mentioned earlier.

Since these large-scale services can deploy media gateways, it's clear that
this is not a significant impediment. PSTN calling is a paid-for service,
after all.

On Tue, Sep 27, 2011 at 11:12 AM, Roman Shpount <roman@telurix.com> wrote:

> I am sorry, but can you name "robust telephony services" that currently
> provide ICE and SRTP support in US? AFAK Verizon, Qwest, Level3, Global
> Crossing, Paetec, XO do not provide this.
> _____________
> Roman Shpount
>
>
>
> On Tue, Sep 27, 2011 at 10:55 AM, Justin Uberti <juberti@google.com>wrote:
>
>> We can still talk to them via a media gateway, which robust telephony
>> services will already provide for the reasons I mentioned previously.
>>
>> On Tue, Sep 27, 2011 at 10:49 AM, Roman Shpount <roman@telurix.com>wrote:
>>
>>> Sorry, correcting myself, there are over 5 billion mobile phones
>>> worldwide. Are you sure we don't need to talk to them.
>>> _____________
>>> Roman Shpount
>>>
>>>
>>>
>>> On Tue, Sep 27, 2011 at 10:40 AM, Roman Shpount <roman@telurix.com>wrote:
>>>
>>>> Bernard,
>>>>
>>>> What about 500M Skype users? Skype for SIP does not support SRTP or
>>>> ICE.What about 2 billion wireless subscribers? Despite what you think number
>>>> of phones and phone numbers is growing, and a wast majority of these phones
>>>> are dumb wireless phones.
>>>> _____________
>>>> Roman Shpount
>>>>
>>>>
>>>> On Mon, Sep 26, 2011 at 11:44 PM, Bernard Aboba <
>>>> bernard_aboba@hotmail.com> wrote:
>>>>
>>>>>
>>>>> > And "interoperability with SIP-PSTN providers" is only relevant if
>>>>> you
>>>>> > are trying to turn the browser into another phone. We have enough
>>>>> > phones. What we don't have are new real-time communication
>>>>> experiences
>>>>> > that can only be created within this environment.
>>>>>
>>>>> [BA] I think you may be on to something here.  POTS lines are dropping
>>>>> so rapidly
>>>>> that wireline service won't even exist in some countries within a few
>>>>> years.  While
>>>>> PSTN interop may have been a key scenario in the early days of SIP, I
>>>>> have serious
>>>>> doubts as to how much attention we should pay to this in RTCWEB.
>>>>> Ability to
>>>>> make a call to an E.164 number?  Probably.  Support for every potential
>>>>> corner
>>>>> case?  Not necessarily.
>>>>>
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>