It is really hard to avoid this discussion, but one cannot let such nonsense
stand, especially in the IETF.
With the work on SIP participants very early on realized that there are other
organizations interested in real-time communication.
This was not really seen as a problem given that
+1
Thanks, Henry
On 3/10/11 12:24 PM, Ted Hardie ted.i...@gmail.com wrote:
On Thu, Mar 10, 2011 at 9:29 AM, Ed Juskevicius edj@gmail.com wrote:
I also recall a Plenary presentation during IETF 57 in Vienna which
suggested a reversal in the IETF's previous stance on this topic.
Brian,
Having running code only as a guideline has not served the IETF well lately,
since it is largely ignored.
I am still cringing during the IETF SIMPLE meetings when we use Jabber IM that
has the code free and available.
Would the SIMPLE WG have had the mandatory requirement of running
The I-D ³P2P Status and Requirements² is a useful start, though it points to
many items that have to be discussed and corrected, such as:
* This is a bandwidth view only of the (elephant) problem. The correct title
should specify this and could be for example: ³P2P Bandwidth Issues: Status
and
mentioned?
Thanks, Henry
-Original Message-
From: Rémi Denis-Courmont [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 27, 2008 2:16 AM
To: Henry Sinnreich
Cc: Magnus Westerlund; Cullen Jennings; Peterson, Jon; Gonzalo Camarillo; Jeff
Goldberg (jgoldber); Philip Matthews; Lars Eggert
interoperability.
Henry Sinnreich
-Original Message-
From: Elwell, John [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 08, 2007 7:59 AM
To: ietf@ietf.org; IETF-Announce
Cc: [EMAIL PROTECTED]
Subject: RE: [Sipping] Last Call: draft-ietf-sipping-service-examples
(SessionInitiation Protocol
Folks,
Thats what RAQMON doesin a media gnostioc fashion.
RAQMON is an excellent approach for network elements
The media statistics for SIP calls are intended at the application level in
endpoints, such as when using an Instant Messenger client for voice or a
soft phone, or a SIP phone, or a
mobility and integration of communications with applications
based on the SIP events architecture and it seems quite clear that H.323 is
not even in the same class as SIP.
Thanks, Henry
Henry Sinnreich
MCI
1201 E. Arapaho Rd./Office 1020
Richardson, Texas 75081
USA
-Original Message-
From
Is the sender anonymous or could we know
name and affiliation?
Thanks,
Henry Sinnreich
MCI
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Tuesday, September 02, 2003
11:10 PM
To: [EMAIL PROTECTED];
[EMAIL PROTECTED]
Subject: Re
Is the sender anonymous or could we know name and affiliation?
Thanks,
Henry Sinnreich
MCI
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, September 02, 2003 11:10 PM
To: [EMAIL PROTECTED]; [EMAIL
Could one use the NAPTR concept to create a new identifier space with
specific dynamics? It would take two lookups: one to DNS to get the NAPTR
and one to resolve the NAPTR identifier into an IP address.
We will be soon able to test the speed of such a mechanism with the NAPTR
client built
systems?
Thanks, Henry
-Original Message-
From: Keith Moore [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 21, 2003 6:01 PM
To: Henry Sinnreich
Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED];
[EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED];
[EMAIL PROTECTED
]
Sent: Thursday, August 21, 2003 9:55 AM
To: Henry Sinnreich
Cc: 'vinton g. cerf'; 'Marshall Rose'; 'Peterson, Jon'; [EMAIL PROTECTED];
Alan Johnston; Robert Sparks
Subject: Re: WG Review: Centralized Conferencing (xcon)
Greetings.
Henry, If we wanted to do a SIP-only conferencing
requirements for
conferencing?
If this was an emotional response, you need not reply.
Henry
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Keith
Moore
Sent: Thursday, August 21, 2003 12:36 PM
To: Henry Sinnreich
Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED
of millions of desktops: The IETF SIP work is already the
de facto standard. Let's just stay focused.
It is thus entirely appropriate XCON should be a SIP oriented WG for
centralized conferencing.
Thanks,
Henry Sinnreich
MCI
-Original Message-
From: vinton g. cerf [mailto:[EMAIL PROTECTED
So pure Internet SIP won't work for all of us any time soon.
Glad to clear up the confusion on this point. People on the PSTN can
dial in and can be called from the SIP conferencing server by using a
service provider that has standard PSTN-SIP gateways. The typical SIP
voice conference has both
There are excellent SIP voice conferencing bridges available, such as
from snom AG and eDial. They can be used with various soft clients such
as the Windows Messenger, HotSIP Active Contacts or the Pingtel instant
expressa, or any SIP phone.
I have taken the liberty of copying here the contacts
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