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Technology Stocks : Voice-on-the-net (VON), VoIP, Internet (IP) Telephony -- Ignore unavailable to you. Want to Upgrade?


To: elk who wrote (555)5/18/1998 1:31:00 AM
From: Frank A. Coluccio  Read Replies (1) | Respond to of 3178
 
Evan,

>I was trying to help Biz for you! I figured you could throw out some teasers,
maybe get a few folks in need of this sort of advice interested, lay out some
parameters, but they got to pay the bread for the meat!<

Thanks for the thoughts, but I perceived such to be the case, and I considered the perceived backlash, i.e., the old criticism about consultants holding out the tease. So I refrained from coming across in a perceived solicitous manner in the thread.

By offering even the best solution to a single set of circumstances would actually, indeed, be a tease, because like I stated, it is a matter of what ails the patient, and no two patients' disorders are exactly alike. In fact, most are vastly different.

That's the beauty of the Internet Protocol. Once the older, restrictive platform is removed altogether, and the new IP provisions are put in place (and standards are adhered to), network elements lose their fickleness about who will talk to whom, and where. Provided, of course, that security and policy measures allow, and permissions are granted.

If you like, I will do a full-blown treatment using the parameters you cited in your earlier post, if you describe the actual set of givens in a particular networking situation. Now it's *your* turn to go and do some homework. No fair, if you get Charlie Slater to help you, though. He knows where all of the gremlins are. <g>

The rest of your post is real good stuff that we could work with. I would like to address those tomorrow, sometime, as it is bloody late here, and I'm going to get some shut eye.

Good Night to All from Narrows Manor in Bay Ridge, Brooklyn.

Frank C.



To: elk who wrote (555)5/19/1998 5:22:00 PM
From: Frank A. Coluccio  Read Replies (1) | Respond to of 3178
 
Evan, you asked,

>>Could you expound a bit on the Backwards compatibility, as I have not heard it really mentioned before.<<

You already know this, but the name may have thrown you. Nothing exotic, although hard to achieve sometimes, backwards compatibility exists when a new version of any generic object, be it hardware or software, is able to properly work in substitution of (or with) existing or even previous versions of that same generic which was in place before it. Difficulties arise when orchestration must be maintained in a larger environment among many different devices working together in a rev-defined set of processes.

My 5.0 version of Visio Software, for example, backwards compatibly reads the files of the drawings I created earlier with Version 4.5. Version 4.5 software, however, is not able to read 5.o. You are familiar with this joy of automation, aren't you? <g>

Often times a vendor will have proprietary code in a "standards based" application in a line of hardware, software, whatever, and they will release new products that will work backwards with their previous versions, as well as forwards. Potential problems sometimes occur between different vendor lines when a new release from Brand X does not backwards compat with the environment in which its predecessor was placed, or with a previous release of Brand Y's in that environment. It is said to be backwards incompatible then, and the vendors need to duke it out, often resulting in a temporary <cough> patch or fix, or a new version release.

>>On a somewhat related tangent, I read in a recent issue of IP Telephony on some of the issues, outside of the existing codec standards, that affect interoperability. They expressed that there was a need for standards in Bandwidth management, call setup and signaling, among other issues before true compatibility would be capable between the different vendors.If perhaps you or Mr Pulver, and it is great to see him post here, could discuss some of these aspects and if they are being addressed by the ITU or similar..<<

Guess Jeff is off to some show somewhere...

Bandwidth management. Vague concept based on the misuse of a term, often. I'd have to read the actual article, but from what I infer, they are talking about reserving channel resources in some way, perhaps through Resource reSerVation Protocol, or RSVP; real time protocol, or RTP; and real time control protocol, RTCP.

Sometimes altering packet lengths is performed by auto-sensing and compensation capabilities like AGC, to properly compensate for longer-term latency trending. This could result in noticeable differences in quality at the extreme conditions, consequently.

Since VoIP will prevail in a number of diverse environments characterized by varying levels of hostility, i.e., the Public Internet, VPNs, Intranets/Extranets, LANs, you name it, it sometimes will reside in ideal conditions [where traffic management and shaping could be easily controlled and bandwidth is allocated with predictability ], and sometimes not, as in the case of the Public sector.

These dissimilar environments must be negotiated dynamically and the protocol must be able to sense (or set to defaults in the more controlled nets) in order to properly compensate for the adverse effects of variable packet delays and lost UDP packets.

That's an over-simplification to a relatively complex problem. I hope that it adequately addresses your question.

[Curtis, I would be honored if you jumped in here added to this, if you would...]

Call setup and signaling is effectively the equivalent of the same basic sequences that exist when you dial an ordinary telephone number using a black phone. Whereas the PSTN uses SS7 to do look-ahead surveillance and link reservations prior to the connection being made, which could either be prior to or after the determination of the status of the called party's line (if it is busy you get a locally generated busy signal, if it is not, the link is created), on the Internet the hand-shake is done using TCP/IP in combination with the signals of "other origins." The latter is dependent on the nature of the type of interfacing taking place and what it interfaces to.

When you go beyond the simple call setup, other database lookups are required that are beyond a simplified response here.

If you are interested, I could give you the titles and ISBNs of some TCP/IP, ITU-IMTC, CTI and SS7 textbooks that will drive you to distraction and beyond with explanations. <g>

Keep in mind that all of the above, with the possible exception of the backwards compatibility issue, are gross over-simplifications. If things were as simple as I stated, there'd be no stock play over the technology.

Best Regards, Frank Coluccio




To: elk who wrote (555)5/20/1998 7:54:00 AM
From: Frank A. Coluccio  Read Replies (2) | Respond to of 3178
 
||: List of VoIP Standards Considerations :||

Evan,

My previous reply to your questions concerning VoIP QoS-related issues and call set-up procedures was not only overly simplistic, but in an effort to be concise I overlooked many of the parameters, and even some of the more prominent points associated with the new model.

By viewing the url below you will get a fuller sense of the issues that were cited earlier on as the key elements needed to incorporate into the new architecture. It's a document included in the European Telecommunications Standards Institute (ETSI) TIPHON Project proceedings on the study of VoIP standards issues.

Of particular interest to some may be the cast of characters that makes up this cadre of the undertaking, and the inclusion early on of ISDN and GSM Cellular in the blueprint. From the document, it is intended to:

"...establish the state of the art concerning voice interworking between terminals over IP based networks and PSTN/ISDN/GSM terminals, to understand the issues involved and the experience required and to prepare a proposal for further work."

Below, I've included some of the more salient points that may illuminate what those issues were, and still are.

Enjoy, and Regards,
Frank Coluccio
=================(Edited for thread)
From:
iihe.ac.be

It was made a condition that the project will concentrate on the following terms:

ú interoperability between PSTN, N-ISDN, GSM and IP-based networks,

ú the ETSI members will be informed on a wide basis about the progress of the project,

ú to avoid overlap with other ETSI groups,

ú to make subcontracts if necessary only,

ú Internet-specific work concerning TCP and IP protocols is belonging to the IETF and is no part of the TIPHON project.

The scope of the project focuses on voice communication and the project is divided into three phases:

1. Deliverables required for ensuring voice communication from an IP based network voice user to voice users in PSTN/ISDN/GSM.

2. Deliverables required for ensuring voice communication from users in PSTN/ISDN/GSM to voice users in IP based networks.

3. Deliverables required for ensuring voice communication between users of PSTN/ISDN/GSM using IP based networks for the connection/trunking between the involved PSTN/ISDN/GSM

Two "guardian angels", Mr Phil Davidson (BT) and Mr Dieter Kaiser (Siemens) were appointed by the ETSI Board, in order to observe the adherence to these terms.

<delete>

Introduction : The first real TIPHON meeting took place, during three days in Sophia Antipolis, on 20-22 May 1997. This meeting was attended by 48 participants from 13 countries including the United States. Delegates from ITU-T SG 16 (John Magill, Lucent Technologies) and the IMTC VoIP Forum (Scott Petrack, VocalTec) attended the meeting. At least five attendees were regular IETF meetings participants. Many North-American companies were represented (Lucent Technologies, 3Com, Cisco, Motorola, AT&T, Digital, Nortel, Unisys,...).

<delete>

Contributions :

On the second day, six contributions were presented and discussed. These contributions were proposed by Lucent Technologies (TD 15), Ericsson (TD 23), Nokia (TD 14), 3Com (TD 17), VocalTec (TD 18) and Lucent Technologies (TD 19).

ú TD 15 proposed requirements for Internet Telephony Naming and presented an overview of some naming systems.
ú TD23 discussed some basic mechanisms to ensure IP based Networks and PSTN/ISDN/GSM service interoperability.

ú TD14 presented a routing protocol approach for address mapping for IP voice/ISDN interworking.

ú TD17 proposed to use a combination of RSVP, IEEE 801.1p and IEEE 801.1q in order to support Quality of Service (QOS) in LAN and IP environments.

ú TD18 stated several topics in the area of addressing, numbering and multimedia Intelligent Network (IN) services. It suggested the creation of an IN/IP working group.

ú TD19 presented a report about the PINT BOF held at the last IETF meeting. PINT deals with signaling features like click-to-dial, click-to-fax and click-to-fax-back. The purpose of PINT is not to deal with voice over IP and no gateway issues are covered by this project. In our opinion, it can be viewed as a transition phase between the use of two separated voice and IP networks and the use of an integrated voice over IP network. It was agreed at the meeting that no gateway discussion will be initiated in the IETF by TIPHON members.

Working groups :
Following these presentations, a discussion occurred on the project work items and the creation of working groups. It is of prime importance to note that a consensus seems to be established to consider only ITU-T H.323 terminals as the base for all further work in the TIPHON project. It doesn't seem, at this time, that other terminal types will be considered, i.e. architectures defined by other standardization bodies such as DAVIC, MMCF or even the IETF. In particular, the architecture being defined by the IETF (based on RTP but using its own draft-signaling) is not considered at all. Moreover, it should be observed that interworking between multimedia terminals is outside the scope of this ETSI project. On the contrary, the interworking between IETF and ITU multimedia terminals is being studied at the IETF.
<delete>

1. Requirements for service interoperability, technical aspects of charging/billing and security (chair: Mohammad Mahloujian, Ericsson).
2. Architecture and reference configurations (chair: Jozef Vandenameele, Alcatel).

3. Call control procedures, information flows and protocols (chair: Scott Petrack, VocalTec).

4. Naming, numbering and addressing (chair: Louise Spergel, Lucent Technologies).

5. End-to-end quality of service aspects (chair: Barrymore Castle, 3Com).

6. Verification and Demonstration Implementation (interim chair: Michael Blaschitz, Infonova).



ú Web site : etsi.fr