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Today's Topics:
1. Re: JTAPI (Eric Burger)
2. CASE SENSITIVE (Jigar Dani)
3. Diversion Header (Jigar Dani)
4. Important update on SIPconnect 1.1 effort. (Richard
Shockey)
5. Re: CASE SENSITIVE (kanumuri Sreeram)
------------------------------------------------------------
----------
Message: 1
Date: Wed, 07 May 2008 09:06:44 -0400
From: Eric Burger <eburgersipforum.org>
Subject: Re: [SIPForum-discussion] JTAPI
To: <discussionsipforum.org>
Message-ID: <C44721A4.1B718%eburgersipforum.org>
Content-Type: text/plain; charset="US-ASCII"
Of course you can use JTAPI to create an application. You
can also write
applications in assembly language.
The question is not, "can I do it?" the question
is "how long will it take
me to do it?"
JTAPI assumes, for the most part, the computer is the
endpoint for the
media.
JAIN and SIP Servlet lets you use network resources.
Furthermore, you will have to write a lot of support code
for a JTAPI or
JAIN app that you get with SIP Servlet.
It all depends on when you want to be done. If the average
programmer
delivers 50 lines of code per day, I would want those lines
of code to be
for the application, not for support code.
> From: Joel Silva <joel.silvanovabase.pt>
>
> I?m getting a little confused with this subject.
> Can JTapi be used instead of Jain Sip and/or Sip
Servlet to build Sip
> Applications?
> I'm thinking in applications like dual ringing, call
forward, block calls,
> etc.. Is it possible?
> Thanks,
>
>
> Joel Silva
> Analyst
>
>
............................................................
..................
> ...............................
>
> Novabase
> Av. D Jo?o II, lote 1.03.2.3, Parque das Na??es,
1998-031, Lisboa - Portugal
> mailto:joel.silvanovabase.pt
>
www.novabase.pt<blocked::BLOCKED::www.novabase.pt>
------------------------------
Message: 2
Date: Wed, 7 May 2008 10:38:22 -0400
From: "Jigar Dani" <jigar.danidialogic.com>
Subject: [SIPForum-discussion] CASE SENSITIVE
To: <discussionsipforum.org>, <mmusicietf.org>
Message-ID:
<E096369D937D3A4DB271B382850C5C5F0646AD6Dpysmail.eicon.com>
Content-Type: text/plain; charset="us-ascii"
Hi,
One of my interop issues is failing, because of the
Content:
application/sdp line in the SIP header. What I get is
Content:
application/SDP. Is this allowed? Does it matter?
Thanks & Regards,
Jigar
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Message: 3
Date: Wed, 7 May 2008 11:12:36 -0400
From: "Jigar Dani" <jigar.danidialogic.com>
Subject: [SIPForum-discussion] Diversion Header
To: <mmusicietf.org>, <discussionsipforum.org>
Message-ID:
<E096369D937D3A4DB271B382850C5C5F0646AEB9pysmail.eicon.com>
Content-Type: text/plain; charset="us-ascii"
Hi all,
Another quick question:
Can the Diversion header in be split across multiple lines,
with the
username on one line and the uri on the other?
Diversion: "User 1"
Diversion:
<sip:1000192.168.0.1>;reason=unconditional;privacy=off;scre
en=yes
Thanks & Regards,
Jigar
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Message: 4
Date: Wed, 7 May 2008 11:13:07 -0400
From: "Richard Shockey" <richardshockey.us>
Subject: [SIPForum-discussion] Important update on
SIPconnect 1.1
effort.
To: <discussionsipforum.org>
Message-ID: <384201c8b054$dc951b80$95bf5280$us>
Content-Type: text/plain; charset="us-ascii"
It is understandable that some of you might be asking what
the status of the
SIPconnect
1.1 effort is since my last note to the list on Jan 25,
2008, noted below.
There has not been a great deal of discussion on this list
for a number of
reasons.
Typically getting any new work of this complexity off the
ground, even in
the IETF context, requires considerable background
discussions and
negotiation to scope the work and decide on what are the
next proper steps.
The SIP Forum Board of Directors and the SIP Forum Technical
Working Group
Chairs want to take this opportunity to make a number of
important
announcements.
1. We are committed to a SIPconnect 1.1 Specification and we
hereby reissue
our calls for contributions to that end.
The concept of SIP trunking has received a great deal of
industry interest
over the last several months and many vendors and service
providers are
ramping up their efforts. Many of them have come to see the
SIP Forum
SIPconnect and SIPconnect Compliant effort as a valid forum
to help resolve
many of the ambiguities existent in the IETF SIP
specifications and create a
unified "profile" that both PBX vendors and
service providers can rally
around.
The value of this work is self-evident. IP PBX vendors do
not want to have
to go through separate certification processes to connect
with each and
every service provider. More importantly, enterprises want
to know that the
products and services they buy will work together.
2. Microsoft has Joined the SIP Forum and has committed to
actively
contributing to the SIPConnect work.
Notably, Microsoft has formally joined the SIP Forum as a
Full Member
company, and has released, for general distribution, their
view of how
SIPconnect could be constructed. This is based on their own
extensive
experience with their OCS product and extensive discussions
with Enterprises
and Service Providers.
The SIP Forum has created a depository of SIPconnect 1.1
Scoping Documents,
which currently includes the Microsoft document mentioned
above, and will be
the location of other member contributions as they are made
available.
The URL for this material is:
http://www.s
ipforum.org/component/option,com_docman/task,cat_view/gi
d,45/Ite
mid,75/
All members of this list are encouraged to make comments on
the Microsoft
document. The first order of business is to bound the scope
of what
SIPconnect 1.1 should cover.
3. CableLabs has also Joined the SIPForum as a full member
and CableLabs and
its member companies are committed to contributing to the
SipConnect 1.1
effort
As many of you probably already know, CableLabs is the joint
industry
standards body of the North American Cable operators.
More information about CableLabs can be found here:
http://www.cablelabs.com/
It should be no surprise to anyone that Cable Operators want
to offer
Enterprise-oriented SIP trunking services within their
service boundaries.
CableLabs and its member companies believe that the SIP
Forum SIPconnect
process is the best way forward to create a comprehensive
industry-wide
specification for Enterprise IP-PBX to MSO interconnection.
CableLabs has a extensive PacketCable internal working group
devoted to
delivering an Enterprise PBX to MSO specification to
CableLabs partners.
They have agreed to coordinate their efforts with the SIP
Forum SIPconnect
effort as well as make significant contributions to that
end.
4. Broadsoft is also contributing extensive documentation
Related to their
SIP Trunking experience
Broadsoft, a current full member of the SIP Forum, has also
agreed to make
available their extensive documentation of experience with
SIP trunking and
the current implementation of SIPconnect 1.0 as well.
5. World-Class Technical Support
The SIP Forum Technical Working Group Chairs are in the
process of
recruiting a well-known SIP technical expert to actually
edit the SIPconnect
1.1 specification.
The Time to Start the Work is Now!
Members of this list are encouraged to make comments and
propose
contributions to begin this important effort.
The SIP Forum document depository for relevant SIPconnect
1.1 contributions
is located here:
http://www.s
ipforum.org/component/option,com_docman/task,cat_view/gi
d,45/Ite
mid,75/
List Information....
_______________________________________________
techwg mailing list
Send mail to: techwgsipforum.org
Unsubscribe or edit options at:
http://si
pforum.org/mailman/listinfo/techwg
################
SIP Forum Announces Ratification of Version 1.0 of the
SIPconnect Technical
Recommendation
Important Milestone Reached as Leading IP Communications
Industry
Association Formally Adopts Version 1.0 of the SIPconnect
Technical
Recommendation and Announces the Formation of the SIPconnect
v.1.1 Task
Group
Miami Beach, FL, January 23, 2008 - The SIP Forum announced
today that it
has ratified version 1.0 of the SIPconnect Technical
Recommendation with the
unanimous approval of the SIP Forum Board of Directors.
The SIPconnect Technical Recommendation v.1.0, an important
initiative of
the SIP Forum, is a standards-based approach to direct IP
peering between
SIP-enabled IP PBXs and VoIP service provider networks, and
specifies a
reference architecture, required protocols and features, and
implementation
rules necessary for seamless interconnection between IP PBXs
and VoIP
service providers. As SIPconnect was designed to directly
tie enterprise
customers to VoIP service providers using an end-to-end IP
connection, it
eliminates the need for a VoIP gateway, improves voice
quality and creates a
stable foundation for personalized applications and rich
media services.
The formal adoption by the SIP Forum Board of the SIPconnect
1.0 Technical
Recommendation is based on recognition that the
recommendation has been
through credible peer review, including broad membership and
significant
community review, that it is stable and is well-understood,
and that it is
believed to have resolved known design choices.
"I want to thank all of the participants in the
SIPconnect Task Group for
their selfless efforts and valuable time necessary to reach
consensus and
complete the ratification process," said Richard
Shockey, Technical Working
Group Chair of the SIP Forum. "This was truly an
industry-wide effort that
we can continue to build on."
Eric Burger, Chairman of the Board of the SIP Forum,
commented, "You can
expect more work like this from the SIP Forum as we continue
to grow our
community and revitalize our Technical Working Group to
tackle additional
interoperability issues."
The SIPconnect Technical Recommendation has recently enjoyed
a number of
independent implementations, as witnessed by the rapid
adoption of the
SIPconnect Compliant Program. The SIPconnect Compliant
Program offers
equipment manufacturers, software providers and service
providers the means
to attain SIPconnect Certification by the SIP Forum, as well
as enjoy the
use of a unique certification mark created by the Forum to
ensure
interoperability among IP communications equipment, software
and services.
The SIPconnect Compliant logo serves as the "Seal of
Compatibility" for
companies that have agreed to abide by the terms of the
licensing agreement,
and is permitted only on compliant equipment and services
that has been
rigorously reviewed by the SIP Forum SIPconnect Compliant
committee. Several
industry leaders and SIP Forum members have recently been
awarded the
SIPconnect Compliant mark, including Acme Packet, Avaya,
BroadSoft, Cbeyond,
Digium, Ingate Systems and McLeodUSA.
The SIPconnect v.1.1 Task Group
Due to the rapid adoption of SIPconnect and the SIPconnect
Compliant
Program, there has been substantial interest in enhancing
the work through
an updated version of the recommendation - SIPconnect 1.1.
To that end, the
SIP Forum has announced the formation of the SIPconnect 1.1
Task Group.
As with all SIP Forum recommendations, SIPconnect 1.1 will
not attempt to
create new standards, since that is the work of standards
bodies such as the
IETF. The goal of SIP Forum specifications is to develop
workable and
implementable technical recommendations or profiles based on
existing
industry standards.
The newly formed SIPconnect 1.1 task group will follow the
same IETF-like
procedures that were adopted by the SIP Forum Technology
Working Group and
successfully used by the SIPconnect 1.0 task group.
In order assess the requirements, goals and potential
milestones for the
task group charter, it will be necessary to assess from the
SIP community
what is required in the next version of SIPconnect. To that
end, the SIP
Forum Board of Directors wishes to formally open a
discussion of
requirements for SIPconnect 1.1 at this time.
Those interested in participating in the SIPconnect v1.1
development effort
and being part of the Task Group must be either a
Participant or Full Member
of the SIP Forum. To join the SIP Forum, please visit the
SIP Forum website
at http://www.sipforum.org.
About The SIP Forum
The SIP Forum is an IP communications industry association
that engages in
numerous activities that advance and promote SIP technology,
such as the
development of industry recommendations, the SIPit
interoperability and
testing events, special interoperability workshops, and
general promotion of
SIP in the industry. One of the Forum's recent technical
activities is the
development of the SIPconnect Technical Recommendation -- a
standards-based
recommendation that provides detailed guidelines for direct
IP peering and
interoperability between IP PBXs and VoIP service provider
networks, and the
SIPconnect Compliant Certification Program through which
eligible companies
can gain SIPconnect validation and the right to license the
use of the SIP
Forum's 'SIPconnect Compliant' certification mark -- the
official brand of
the leading standard for SIP Trunking products and services.
For more
information, please visit: http://www.sipforum.org.
Richard Shockey
Director, Member of the Technical Staff
NeuStar
46000 Center Oak Plaza - Sterling, VA 20166
PSTN Office +1 571.434.5651
PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
<mailto:richard.shockey(at)neustar.biz>
------------------------------
Message: 5
Date: Wed, 7 May 2008 08:33:49 -0700 (PDT)
From: kanumuri Sreeram <skanumuryyahoo.com>
Subject: Re: [SIPForum-discussion] CASE SENSITIVE
To: Jigar Dani <jigar.danidialogic.com>,
discussionsipforum.org,
mmusicietf.org
Message-ID: <612730.29352.qmweb56915.mail.re3.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi Jigar,
Content-type: application/sdp? and
Content-type: application/SDP
Both are same.
Refer? RFC 2045, Section 5.1
content := "Content-Type" ":" type
"/" subtype
??????????????? *(";" parameter)
??????????????? ; Matching of media type and subtype
??????????????? ; is ALWAYS case-insensitive.
I can see as content , not as content-type. may be that is
the problem.
HTH,
Sreeram
----- Original Message ----
From: Jigar Dani <jigar.danidialogic.com>
To: discussionsipforum.org; mmusicietf.org
Sent: Wednesday, May 7, 2008 10:38:22 AM
Subject: [SIPForum-discussion] CASE SENSITIVE
Hi,
?
One of my interop issues is failing, because of the Content:
application/sdp line in the SIP header. What I get is
Content: application/SDP. Is this allowed? Does it matter?
?
Thanks & Regards,
Jigar
____________________________________________________________
________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9
tAcJ
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