April 1999 - sg16-avd - lists.packetizer.com

Event reporting (was audio call Thursday: Minutes)
by Christian Huitema 08 Apr '99

08 Apr '99

> 5) event handling > - currently an event is identified to the MGC with an eventId - Paul > Sijben suggested that it would be better to use a terminationId Sorry for missing the call, but let's comment on that specific point. We discussed the event reporting model at length during the IETF meetings and in the following week, and I don't beleive that there is any advantage in opening the debate again. Termination Id and event names just do not belong to the same space. Terminations are defined on a per MG basis; events are defined independently of the MG, and in many cases independently of the termination class. The current notifications identify both the termination on which the event was observed and the name of the event. There is a lot of experience to show that this is a very efficient decomposition, and I don't see any reason whatsoever for ditching the experience in favor of an unproven theoretical design. -- Christian Huitema ------------------------------ Please note my new address: huitema(a)research.telcordia.com http://www.telcordia.com/

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Transport issue (was: audio call Thursday: Minutes)
by Christian Huitema 08 Apr '99

08 Apr '99

> 7) transport issues > There was discussion on what to use for interoperability purposes until the > actual transport is agreed on. Some people wanted to mandate TCP for the MG, > others said UDP + the message acks, + the addition of timers would be > better. Are there MG manufacturers that do not want to put TCP in their MGs, > even if only for demonstration purposes? If there is opposition to TCP even > for demonstration purposes, then we would go with UDP + message acks, + > timers. Chip Sharp reminded us that SGCP worked just fine with UPD, acks, > and timers. ===> An issue for the mailing list. Using TCP would be a serious problem in large configurations. Our experience is that UDP+timers has a wider applicability and is not in practice harder to implement than TCP -- you have to manage timers, but then you don't have to manage connections. -- Christian Huitema ------------------------------ Please note my new address: huitema(a)research.telcordia.com http://www.telcordia.com/

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audio call Thursday: Minutes
by Nancy-M Greene 08 Apr '99

08 Apr '99

Minutes of the IETF/ITU Megaco/H.GCP audio call held April 8, 1999, 11am-1pm EDT 22 people participated, with participants from IETF Megaco, and from ITU-T SG16. Topics discussed: a) issues with the current Megaco protocol internet-draft (draft-ietf-megaco-protocol-00.txt) b) status on folding this work into H.gcp c) report on US Study Group D meeting April 6th/99 d) miscellaneous Action items: i) Next version of documents: April 9th: Bryan Hill will release the current version of H.gcp April 16th: Brian Rosen will release an new version of the Megaco protocol internet-draft. He will coordinate with Bryan Hill, to also get out a new version of H.gcp at that time that is as identical as possible to the new internet-draft. ii) Tom Taylor volunteered to produce write-up explaining two approaches for handling H.320 gateways - input will come from email msgs sent out on this topic by Brian Rosen and Paul Sijben. iii) Tom Taylor will put together an Issues List, where each issue is described, zero or more resolutions proposed, and there will be a tag marking the issue as open or closed. All editor's notes will be pulled out of the body of the internet-draft/H.gcp, and put in the issues list, with references back to the internet-draft. The issues list will be included in each revision of the internet-draft/H.gcp. Next meeting: - no date was set for an audio call. There was some discussion on having one some time before April 21st. If one is scheduled, it will be announced on the ITU-SG16 and megaco reflectors. Detailed report: Brian Hill opened the meeting. a) issues with the current Megaco protocol internet-draft (draft-ietf-megaco-protocol-00.txt) As stated above, Tom Taylor will put together an Issues List, where each issue is described, zero or more resolutions proposed, and there will be a tag marking the issue as open or closed. All editor's notes will be pulled out of the body of the internet-draft/H.gcp, and put in the issues list, with references back to the internet-draft. The issues list will be included in each revision of the internet-draft/H.gcp. Issues discussed in the audio call today: 1) association of contexts and terminations - is the context model, with one media type per context, adequate for multimedia? Tom Taylor says he believes the current context model with one media type per context is adequate. Brian Rosen explained that keeping it this way simplifies things for the common case where there is just one media type per call (i.e. audio). Paul Sijben would still prefer to allow multiple media types per context. When there was just one media type, the fields identifying media type, or separate media streams would be optional. Brian Rosen is still not convinced. Conclusion: Tom Taylor volunteered to produce a write-up explaining the two approaches for handling H.320 gateways - input will come from email msgs sent out on this topic by Brian Rosen and Paul Sijben. We agreed that figures for this discussion can be distributed in pdf instead of ascii. Discussion on what you would need to synchronize between contexts: - lip synch - issue - do you link contexts, or do you link a particular termination in a context to a particular termination in another context? How do you know that the video stream going out is the one that you want to lip sync with the audio stream going out? If you were lipsynching the current speaker and you knew that both the audio and video were of the current speaker, you could lipsync the contexts. But what if the video stream going out was NOT the one related to the audio? - synchronize closed captioning - synchronize the switching of streams being sent based on current speaker 2) H.245 capability negotiation Tom Taylor explained that the capability negotiaton stage is done between MGCs, and thus does not need to be expressable in the Megaco protocol - this negotiation is done BEFORE connection setup. Audit allows the MGC to discover what capabilities the MG has, but it does not give information on mutually exclusive choices based on processing power availability, for example. Need to walk through the example of Test Case 1 that Tom put out before the last IETF meeting, where the terminating MG may choose G.711 or G.729 - does the originating MG reserve both G.711 and G.729 for the termination, or just one of the two, and then modify it based on what the terminating MG chose? The test case allows G.711 in one direction and G.729 in the other. 3) attributes per termination class - these need more detail 4) how to extend the protocol - this needs to be spelled out 5) event handling - currently an event is identified to the MGC with an eventId - Paul Sijben suggested that it would be better to use a terminationId 6) terminationIds - there was a question whether terminationIds were logical or physical, i.e., would you put the word audio in a terminationId to make it a logical name? The answer was NO, a terminationId is a name of a physical termination, not a logical one. 7) transport issues There was discussion on what to use for interoperability purposes until the actual transport is agreed on. Some people wanted to mandate TCP for the MG, others said UDP + the message acks, + the addition of timers would be better. Are there MG manufacturers that do not want to put TCP in their MGs, even if only for demonstration purposes? If there is opposition to TCP even for demonstration purposes, then we would go with UDP + message acks, + timers. Chip Sharp reminded us that SGCP worked just fine with UPD, acks, and timers. ===> An issue for the mailing list. 8) security issues - no one had issues with security as it is described in the Megaco protocol internet-draft 9) syntax issues - discussion on this issue was deferred 10) Handling of NAS - can the protocol handle dial-out? this should be no problem - indication that the NAS wishes to dial out would be an event from the NAS (MG) to the MGC (NAS Controller), then the MGC issues an Add into a new context. - if the MGC does directory lookup to an AAA server, and there are Radius attributes to pass to the NAS, these need to be spelled out ===> need to make sure the foundations are in place to handle NAS. If termination classes, etc. for NAS are not ready for this internet-draft, they can go in as a separate RFC. But we will try to get them ready in time. b) status on folding this work into H.gcp As mentioned in the Action Items list above, Brian Hill, the H.gcp editor, will be releasing the current version of H.gcp tommorrow. It has much of the Megaco internet-draft folded into it. Everyone agreed that we want to keep these two protocols aligned, and identical if at all possible. When Brian Rosen releases a new version of the internet-draft by April 16th, he will coordinate with Bryan Hill to get this version into a new version of H.gcp as well. A question was asked on what a determined document means in the ITU? Tom Taylor answered that it means the technical directions are firmly set. A document that is to be decided at an ITU meeting must be ready four months before the actual meeting. In this case, it means it must be ready by Nov/99 (white paper deadline) to be decided at the Feb/2000 SG16 meeting. If interoperability testing of the protocol starts in early fall, there will be time to feed any corrections needed into the protocol before the Nov/99 white paper deadline. With all this cooperation, there is a feeling that there is a good chance to get one common protocol document between the IETF and the ITU. c) report on US Study Group D meeting April 6th/99 It seems there is still an issue that the US may oppose determining a document they haven't seen yet, unless there is "overriding industry interest" in having the document determined. So, any US company interested in having H.gcp determined should make their interest in determining H.gcp known. d) miscellaneous Chip Sharp asked who is using H.226. No one on the call was. Meeting Conclusion: See above for action items. -end of the meeting- Nancy -------------------------------------------------------------------------- Nancy M. Greene Internet & Service Provider Networks, Nortel Networks T:514-271-7221 (internal:ESN853-1077) E:ngreene@nortelnetworks.com

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PER ASN Encode of Open type elements
by Manoj Paul 07 Apr '99

07 Apr '99

We would like to seek opinion about the PER ASN encoding of open type elements (such as extension additions in sequence/choice etc.). Such an encoding requires unconstrained length "len" to be appended before the actual field-list of open type element, where "len" is the number of octets in open type element field list encoding. There seem to be the following ways to achieve this: a) Keep encoding the open type element and when it's encoding is finished, count the number of octets in it's encoded value and add that before the encoded value as unconstrained length. b) Just parse through open type element encode rules and find the number of octets it will need. Encode this number as unconstrained length and then do the acutal encode of the open type element value. Implementing (a), on most systems might be expensive, as it will need writing "len" value somewhere in the middle of the message. (b) is also not desirable from performance point of view as it needs double parse. Encoding "len" value less than 16K (which it mostly will) as uncostrained length requires two bytes. But X-691 also says that this length should be encoded in one byte if it is less than 127. Is it conforming to X-691, if length value less than 127 is also encoded in two bytes? That way, we know that whatever be the overall number of octets in encoded open type element field list, we will encode "len" in two octets and keep a place holder for these twp bytes before actually encoding the open type value. These "len" octets can then be written to contain unconstrained length of open type value. This procedure will save the inefficiencies mentioned in (a) and (b) above. But then the catch being if length less than 127 can be encoded in two octets. best regards Manoj Paul. +------------------------------+--------------------------------------+ |Manoj Paul | Trillium Digital Systems, Inc | |Member of Technical Staff | 12100 Wilshire Blvd #1800 | |E-mail:mpaul@trillium.com | Los Angeles, California 90025 | |Tel: +1(310)-442-9222 | U.S.A | +------------------------------+--------------------------------------+

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Re: ASN.1 vs Text
by Paul Long 07 Apr '99

07 Apr '99

The experiment should implement the entire protocol and include a large set of messages. The syntax should then be extended in various ways. Text encoding is seductive. At first, text looks easier than ASN.1, but as more complex structures are needed and the syntax grows, it becomes cumbersome and error-prone. Paul Long Smith Micro Software, Inc. -----Original Message----- From: Tom-PT Taylor [SMTP:taylor@NORTELNETWORKS.COM] Sent: Wednesday, March 31, 1999 9:17 PM To: ITU-SG16(a)MAILBAG.INTEL.COM Subject: ASN.1 vs Text Since the other responses seem to have gone off into a discussion of firewalls, let me respond to this more directly by recalling a bit of past discussion. 1) On the Megaco list, we finally agreed that the text vs. binary question is something to be resolved by experiment. 2) In Turin (I think), the general opinion was that the protocol would have to be binary to achieve the required performance. Interestingly enough, however, no one that I recall felt that it should be PER ASN.1 (or even, I think, BER ASN.1). Good reasons were given at the time. It would be good if someone can remember them, but I think it has something to do with processing performance and less risk of fragmentation than call signalling. 3) We have had an occasionally recurring suggestion on the Megaco list that the Media Gateway control protocol should conform to the format of other messaging out of the MGC, for the reason you give (i.e. to minimize transcoding). Interestingly, the person advocating this was talking about billing data, not signalling. The counter-argument on the list is that the Media Gateway control protocol must work with different signalling protocols (for example in H.GCP's case, with any of the H-series systems), so there is no point in optimizing it for just one. The real sticking point of debate is going to be whether the media description structures will be those defined for H.245 OLC etc., or SDP. Text vs. binary is a closely related but broader debate. I'd suggest we start thinking about experimental design. > -----Original Message----- > From: Ami Amir [SMTP:amir@RADVISION.RAD.CO.IL] > Sent: Wednesday, March 31, 1999 4:15 AM > To: ITU-SG16(a)MAILBAG.INTEL.COM > Subject: H.320 gateways a MEGACO / ITU > > Hi > > During the last MEGACO conference call - Tom Taylor asked what could > hinder > ITU acceptance of the MEGACO work. It is obviously to everybody's interest > that IETF and ITU standards merge. As an example - it would be great if > the > MEGACO work could become an ITU SG 16 contribution. > > An obvious item is efficient multimedia support (as reflected also in > John's > mail). > > However, I think that one of the even more major difficulties facing us > will > be the encoding scheme (ASN.1 vs Text). > > There are many who feel that ASN.1 is too heavy and complex for simple > devices, and should be avoided. This was one of the major reasons for the > emergence of SIP. > > On the other hand, experience in the ITU H.323 work has shown that since > ASN.1 is the encoding scheme on the PSTN side, the use of ASN cleared the > way for PSTN to IP interoperability. This feature will be extremely > important in hybrid networks that need to provide Intelligent Network (IN) > services (e.g. "800"), while retaining the investment in existing IN, > billing and directory services (411). > > Another problem is that if ASN.1 is not chosen, every device that will > have > to connect between a MEGACO component and H.323 will need to dis-assemble > and re-assemble (transcode) messages, and hence network performance will > suffer, and those devices will be more complex. A prime example - MGC to > GK > communications. > > I am not promoting a specific approach. I just think that this complex > issue > needs to be addressed if we want to be able to be able accept a universal > protocol. > > Do you think this is really a problem? > If so - any ideas on how to bridge the gap? > > Ami > > -----Original Message----- > From: John Segers [SMTP:jsegers@lucent.com] > Sent: Tuesday, March 30, 1999 6:27 PM > To: ITU-SG16(a)mailbag.cps.intel.com > Subject: H.320 gateways > > People, > > In yesterday's conference call, the subject of H.320 GWs was > raised > briefly. In my opinion, the connection model and protocol should > be > able > to deal with H.320. I would like to continue discussion on it on > the > mailing list. > > H.320 allows a user to have a session with both audio and video on > a > single 64 kbit/s channel such as an ISDN B-channel. The same > channel > carries some signalling information (frame alignment, bitrate > allocation). To a MG supporting H.320, this means that on a > single > endpoint, three streams can come in, carrying different types of > media. > The current connection model of megaco/H.gcp does not cater to > this. > I > see two possible solutions: > > The first is to allow multiple media in one context and to > describe > for > terminations the logical streams they carry. In a picture: > > +----------+ > | | > | +--------- signalling (FAS, BAS) > | | > B-channel ==========+ +--------- audio (16 kbit/s) > | | > | +--------- video (46.4 kbit/s) > | | > +----------+ > > The second solution is to have separate terminations for the > different > streams. They would all "connect to" the same physical endpoint. > In > order to properly identify the terminations, it is necessary to > have > logical names for them. The physical endpoint they connect may > have > the > hierarchical name proposed in the megaco document. > > Another example of a H.320 session is the case of two B-channels > being > used for an audiovisual call. The following frame structure is > then > possible. > > +--------------------------++-----------------------+ > | Channel 1 || Channel 2 | > +-----+--+--+--+--+--+--+--++--+--+--+--+--+--+--+--+ > |Bit 1|B2|B3|B4|B5|B6|B7|B8||B1|B2|B3|B4|B5|B6|B7|B8| > +-----+--+--+--+--+--+--+--++--+--+--+--+--+--+--+--+ > 1 | a1 |a2|a3|a4|a5|a6|v1|F ||v2|v3|v4|v5|v6|v7|v8|F | > 2 | a7 |a8|a9|a |a |a |v9|F ||v |v |v |v |v |v |v |F | > 3 | a |a |a |a |a |a |v |F ||v |v |v |v |v |v |v |F | > 4 | a | | | | |a |v |F ||v | |v |F | > 5 | a | | | | |a |v |F ||v | |v |F | > 6 | a | | | | |a |v |F ||v | |v |F | > 7 | a | | | | |a |v |F ||v | |v |F | > 8 | a | | | | |a |v |F ||v | |v |F | > +---------------------------------------------------+ > 9 | a | | | | |a |v |B ||v | |v |B | > 10 | a | | | | |a |v |B ||v | |v |B | > 11 | a | | | | |a |v |B ||v | |v |B | > 12 | a | | | | |a |v |B ||v | |v |B | > 13 | a | | | | |a |v |B ||v | |v |B | > 14 | a | | | | |a |v |B ||v | |v |B | > 15 | a | | | | |a |v |B ||v | |v |B | > 16 | a | | | | |a |v |B ||v | |v |B | > +---------------------------------------------------+ > 17 | a | | | | |a |v |v ||v | |v |v | > . > . > . > 80 | a | | | | |a |v |v ||v | |v |v | > +---------------------------------------------------+ > > (a=audio, v=video, F=FAS, B=BAS). > > We see that the video stream is split up over two channels. In > order to > cater to this, it seems we have to allow terminations to receive > media > from and send it to multiple physical endpoints. The two > approaches > outlined above can both be extended to allow this. Both extensions > will > lead to the introduction of logical names for terminations. In > the > first approach there will be one termination "containing" two > B-channels > on one side and three logical streams on the other. In the second > approach there will be three terminations, the one for the video > stream > referencing both B-channels, the ones for signalling and audio > referencing only channel 1. > > The second approach allows us to keep separate contexts for > different > media types. It is then easy to delete, for instance, the video > part of > a session (session used loosely to desribe the contexts for the > audio > and video). > > The first approach groups the streams coming from/going to one > user, > making it possible to remove a user from a context more easily. > > > Personally, I can't decide which approach I would prefer. How do > others > feel about these ideas? > > Regards, > > John Segers > -- > John Segers email: > jsegers(a)lucent.com > Lucent Technologies Room HE > 344 > Dept. Forward Looking Work phone: +31 35 687 > 4724 > P.O. Box 18, 1270 AA Huizen fax: +31 35 687 > 5954

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joining the megaco list
by Nancy-M Greene 07 Apr '99

07 Apr '99

Some of you have been asking how to join the IETF Megaco working group mailing list. Here is how: Send a message to majordomo(a)baynetworks.com with subscribe megaco end in the message body. Nancy -------------------------------------------------------------------------- Nancy M. Greene Internet & Service Provider Networks, Nortel Networks T:514-271-7221 (internal:ESN853-1077) E:ngreene@nortelnetworks.com

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Re: Firewalls [was: H.320 gateways a MEGACO / ITU]
by Melinda Shore 06 Apr '99

06 Apr '99

There's an awful lot of overlap between HFCI and SOCKS. Additionally, HFCI is still rather incomplete yet has some enormously useful stuff in it, so I'm hopeful that w'll be able to resolve these two over the next 3-6 months within the IETF or elsewhere. Melinda At 12:09 PM 4/6/99 +0100, Iain Barker wrote: >One solution would be not to expect the firewall to 'find' >which ports to open, but instead for the gateway to inform >the firewall of the nature of the underlying connection. > >FYI, SIEMENS have a submission to TIPHON which disusses how a >media gateway [H.323 in their example] can instruct the firewall >which IP ports to allow through. Melinda Shore Member of the Scientific Staff Nokia IP Telephony 127 West State Street Ithaca, New York 14850 +1 607 273 0724 x81 (office) +1 607 275 3610 (fax) +1 607 280 0010 (mobile) shore(a)ithaca-viennasys.com

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audio call Thursday
by Bryan Hill 06 Apr '99

06 Apr '99

We decided at the last audio call March 29th/99 (minutes went out the same day), to hold the next call Thursday April 8th/99. The goal of the call is to move towards agreement for content of H.GCP, based on the draft Megaco protocol (draft-ietf-megaco-protocol-00.txt). As well, the call is a place where issues with the protocol can be discussed, and questions answered. As before, no "decisions" can be made on the call. Minutes will be posted to both itu-sg16(a)mailbag.intel.com <mailto:itu-sg16@mailbag.intel.com> and megaco(a)baynetworks.com <mailto:megaco@baynetworks.com> . Content discussion of the Megaco protocol, by convention, is taking place on the megaco(a)baynetworks.com <mailto:megaco@baynetworks.com> list. Here are the call details: Date: April 8th, 1999 Time: 10amEDT - 12pmEDT (we switched to EDT on Apr 4th) Phone #: 1-613-763-6338 PassCode: 131313# (same as last time) ***Problems getting on? please call 1-613-765-CONF and ask for the conference chaired by Nancy Greene. ***Please share ports with colleagues if possible. 60 ports have been booked. ***Because of the large # of participants, no tones will announce people entering or leaving the conference. ***Please mute your phone when you are not speaking (use either your mute button or press 63 to mute and 66 to unmute) Nancy Greene Bryan Hill ngreene(a)nortelnetworks.com <mailto:ngreene@nortelnetworks.com> bhill(a)videoserver.com <mailto:bhill@videoserver.com>

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Re: Issues Around Support of Multimedia
by Christian Huitema 06 Apr '99

06 Apr '99

Tom, I agree with your analysis of requirements. We need to support multuimedia conference, yet we should not loose sight that H.320 is only a very small fraction of the initial market, and that we should not unduly increase the complexity of an audio only system. I also agree that we have to support multicast, and that we have to provide for fancy control of (at least some) conferences. The architecture we had in mind, as Brian explained, is to have one context per media. This brings in the notion of a "context type" -- an audio bridge, a video bridge, a T.120 bridge all behave somewhat differently. In fact, the particular realm of video bridging is prone to divisions such as "single image", "mosaic", "incrustations", which could be defined as context level parameters. In fact, a common practice on the MBONE is to carry several video streams on the same RTP association, using the SSRC as the stream identifier, and letting individual recipients select which video is displayed and which one is iconized. Short of that, conference control does not appear to be a real problem with the current spec. The mode parameter allows the MGC to specify the terminations that "contribute" to the mix (allowed to send) and the ones that only receive information. In a video bridge, the contributing terminations are sent in the mosaic, or multiplexed as RTP streams. In a T.120 bridge, the "master" termination is authorized to contribute, the other one remain passive. Synchronization of multiple context can be performed by the MGC: event reporting allows it to detect the last speaker, or video activity, or T.120 activity, and to issue MODIFY command that manage the conference accordingly. If the event/command chain is deemed to slow, then we will have to resort to a local script, a surrogate for the MGC. But there is more. Support for H.320 requires, in particular, the support for H.221 multiplexing of several streams onto a single carrier. That support is somewhat similar to the support of several DS0 inside a single DS1, or several B-Channels, albeit there are some obvious differences -- multiplexing parameters vary on a call per call basis, or even within a call. We have to express the notion that a specific termination, for the duration of a call, is "exploded" into a set of ephemeral terminations, each of which is then connected to a media specific context. This requires an additional mechanism, something like a 'demultiplexing context'. Demultiplexing has a symmetric, inverse multiplexing. This happens when an H.320 conference spans several B channels. It can also happen for data calls. In order to at least obtain the "warm and fuzzy feeling" that Fernando mentions, we have at a minimum to sketch out our solution to inverse multiplexing and demultiplexing, and maybe to propose one way to handle H.320. -- Christian Huitema ------------------------------ Please note my new address: huitema(a)research.telcordia.com http://www.telcordia.com/

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FW: AW: Call hold and transfer in H.323 AnnexF. Too limited??
by Chris Purvis 06 Apr '99

06 Apr '99

All, I find it hard to believe that we can't do better (from the user's point of view) than the old black-phone view of the world whereby people have to type in various combinations of *s and digits to mean various different things to the local exchange. Most endpoints, after all, have pretty UIs. Now at the messaging level it may well make sense to have things APPEAR as if that (ie the typing of arcane sequences of digits) is what is happening - and it's possible that all we need to do to make that work in an interoperable way is to define what sequences of characters in the destination address in a setup message mean what sort of service. Do we start from a point such as "Setup with a single destination address which is e164 starting with '*' means arcane supplementary service", or define another alias type for "arcane supplementary service" or what? My aim is simply that an endpoint vendor should be ABLE to write things that do not require the user to understand bizarre control sequences, and will be able to access supplementary services wherever, and by whatever vendor, they are provided. Regards, Chris -- Dr Chris Purvis - Senior Development Engineer, WAVE CC Software Madge Networks Ltd, Wexham Springs, Framewood Road, Wexham, Berks. ENGLAND Phone: +44 1753 661 359 email: cpurvis(a)madge.com > -----Original Message----- > From: Espen Skjæran [mailto:Espen.Skjaeran@ERICSSON.COM] > Sent: 26 March 1999 6:42 > To: ITU-SG16(a)MAILBAG.INTEL.COM > Subject: Re: AW: Call hold and transfer in H.323 AnnexF. Too limited?? > > > Hei, > > I do not think it is a bad thing if ISPs choose to control service > profiles/configuration outside of H.450. There are a lot > more flexible ways to transport data. Proven technologies > exist today which are > very flexible and extendible (HTTP, WAP, IVRs, etc). User > oriented services > might also include non-H.323 equipment (mobile, email, even a > SIP terminal!), so > confining it to H.450 and H.323 would limit the possibillity > for "fancy" > services significantly, thereby also limiting the > distribution of H.323. > > If a Call Forward service is put in the network, it turns > more into a user > oriented than an endpoint oriented service, and the > possibilities for extensions are huge. As soon as the "Call > Forward Unreachable" > is standardized, I might come up with a proposal for call > forward based on > week/time, and this will hardly ever stop. We need a flexible > mechanism for > controlling these services, not new detailed service specifications. > > As the mechanisms for a call forward done in the network does > not require the > same level of interactions as a terminal > oriented (or in-call) service, we do not need to follow the > same approach as > H.450, where services are specified in detail. > The H.450 approach makes the creation (and extension) of a > service a very > lenghty process, and is actually a great > obstacle to creativity and development of new user oriented > services. (Though > for in-call services I don't see any way around it...) > > Espen > > > "Derks, Frank" wrote: > > > Paul, > > > > you got my points exactly right. The reason that I weakened > by arguments by > > hinting at the ease of upgrade in an IP environment was to > avoid people > > coming back with this line of argument. Although using > "DTMF digits" as the > > means to access services may not be the most elegant one, > it is probably the > > simplest and it maps well onto a simple "blackphone" type > of terminal. If we > > do not provide for something fairly simple (both for the > users and the > > service providers) then the chances are that service > providers will come up > > with "(IP) Telephony unrelated" means to access services. > E.g. they could > > provide Web pages that allow customers to specify their > service behaviour. > > > > Frank > > > > -----Original Message----- > > From: Paul E. Jones [mailto:paul.jones@TIES.ITU.INT] > > Sent: 22 March 1999 19:11 > > To: ITU-SG16(a)MAILBAG.INTEL.COM > > Subject: Re: AW: Call hold and transfer in H.323 AnnexF. > Too limited?? > > > > Frank, > > > > I do agree that two mechanisms for accomplishing the same > task is a bad > > idea. I, too, would rather see one mechanism employed-- we > do want to > > create interoperable equipment, after all. Unfortunately, > we already have > > two ways of doing "call hold"-- H.450.4 and "empty > capability sets" (see > > 7.6.2 of Annex F). > > > > The issues you raise with supplementary services echoes the > concerns of > > those also participating in the TIPHON work. Essentially, > service providers > > would like to be able to add new services without upgrading > software in the > > endpoints. Although it may be possible to upgrade IP phone > devices, I can > > assure you that the average person would never do that-- > once the phone is > > plugged in, it will stay there until it stops working. > More importantly, > > why would one want to require somebody who purchased a > hardware phone device > > to upgrade periodically? > > > > We need to engineer a solution so that the telephony > service providers can > > introduce new services without requiring software upgrades > in the endpoints. > > I would like to see SET device to take advantage of those > newly introduced > > services without software or hardware upgrades. > > > > Paul > > > > -----Original Message----- > > From: Derks, Frank <F.Derks(a)PBC.BE.PHILIPS.COM> > > To: ITU-SG16(a)MAILBAG.INTEL.COM <ITU-SG16(a)MAILBAG.INTEL.COM> > > Date: Monday, March 22, 1999 5:45 AM > > Subject: Re: AW: Call hold and transfer in H.323 AnnexF. > Too limited?? > > > > >Folks, > > > > > >when talking about a Simple Endpoint Type, I think we > should aim for it to > > >be something that closely resembles a black phone. This > way it becomes a > > lot > > >easier to define what its capabilities are and it makes > life easy on the > > >users and on those companies that will actually make > (physical) IP-phones. > > >These phones should probably look and act like the normal > phones that are > > >currently being used. Looking at how most supplementary > services are > > >accessed in both the public and the private (PBX) > networks, I think it is > > >safe to say that in most cases we are talking about > "stimulus protocols". > > >I.e. DTMF digits are sent to an exchange and the exchange > interprets > > certain > > >digit sequences as being the invocation of some service > rather than a > > number > > >to be dialled. The big advantage over functional protocols > (like H.450.x) > > >being that services can be added from the exchange side, > without the > > >terminal having to be modified as well. > > > > > >Functional protocols never became a success in the ISDN > world and this may > > >end up to be the same in the IP world. However, having > said this, there is > > a > > >lot more potential for easy upgrading of e.g. terminal > software in this > > >domain, which reduces the side effects of functional protocols. > > > > > >It does not seem to make sense to define "alternative" > mechanisms to > > provide > > >services, so I would strongly opt for using H.450.x when > possible and using > > >a simple stimulus protocol otherwise. The latter would > allow service > > >providers to easily make services available and I see no > reason why this > > >should be standardised. In practice, today, we already see > that certain > > >digit sequences for service activation are identical in > several countries. > > > > > >Frank > > > > > >----------------------------------------------------- > > >Frank Derks |Tel +31 35 6893238 | > > >Advanced Development |Fax +31 35 6891030 | > > >Philip Business Communications |P.O. Box 32 | > > > |1200 JD Hilversum | > > > |The Netherlands | > > >----------------------------------------------------| > > >E-mail: mailto:f.derks@pbc.be.philips.com | > > >WWW: http://www.business-comms.be.philips.com | > > >----------------------------------------------------- > > > > > > > > > > > >-----Original Message----- > > >From: Klaghofer Karl ICN IB NL IP 7 > > >[mailto:Karl.Klaghofer@ICN.SIEMENS.DE] > > >Sent: 18 March 1999 22:36 > > >To: ITU-SG16(a)MAILBAG.INTEL.COM > > >Subject: AW: AW: Call hold and transfer in H.323 AnnexF. > Too limited?? > > > > > > > > >See comment below. > > > > > >Karl > > > > > >> -----Ursprüngliche Nachricht----- > > >> Von: Paul E. Jones [SMTP:paul.jones@TIES.ITU.INT] > > >> Gesendet am: Donnerstag, 18. März 1999 18:57 > > >> An: ITU-SG16(a)mailbag.cps.intel.com > > >> Betreff: Re: AW: Call hold and transfer in H.323 AnnexF. Too > > >> limited?? > > >> > > >> Karl, > > >> > > >> Unfortunately, I will have to disagree with your > comments. While it is > > >> true > > >> that the H.450 supplementary services could be utilized > in a SET device, > > I > > >> believe that introducing H.450 into a SET breaks the > spirit of that work. > > >> > > >> The goal of Annex F is to define a "Simple Endpoint > Type". There are > > >> simpler ways to put a call on hold and to transfer a > call. Introducing > > >> H.450 introduces a lot more complexity that I believe we > want to have. > > If > > >> Annex F is not sufficiently clear on how to simply > transfer a call or put > > >> a > > >> call on hold, we should work on that text-- I will > absolutely disagree > > >> with > > >> introducing H.450 into a SET device. > > > [Klaghofer, Karl PN VS LP3] Whatever you mean with > > "introducing" - > > >H.450 as I sayd in my previous mail is a way of providing > supplementary > > >services like call hold and call transfer to a SET device. > It IS already > > >part of the H.323 Annex F! > > >> Paul > > >> > > >> -----Original Message----- > > >> From: Klaghofer Karl ICN IB NL IP 7 > <Karl.Klaghofer(a)ICN.SIEMENS.DE> > > >> To: ITU-SG16(a)MAILBAG.INTEL.COM <ITU-SG16(a)MAILBAG.INTEL.COM> > > >> Date: Wednesday, March 17, 1999 3:27 PM > > >> Subject: AW: Call hold and transfer in H.323 AnnexF. Too > limited?? > > >> > > >> > > >> >Gunnar, > > >> > > > >> >You are referring to call hold and transfer in > conjunction with H.323 > > >> Annex > > >> >F SETs (Audio or Text) and clause 7.6 of H.323 Annex F. > > >> > > > >> >Talking about call hold, clause 7.6 of H.323 Annex F is > not needed for a > > >> SET > > >> >at all. Call Hold works for a SET as it is defined in H.450.4. > > >> > > > >> >Talking about Call Transfer, clause 7.6 of H.323 Annex > F is not needed > > >> for > > >> a > > >> >SET, if the transfer is executed by the endpoints as > defined in H.450.2. > > >> >Codec re-negotiation you are referring to is no problem > and takes place > > >> >between the transferred and the transferred-to > endpoint. This may cover > > >> your > > >> >case with wireless endpoints being involved. > > >> > > > >> >For call transfer, section 7.6 of H.323 Annex F is only > needed if the > > >> >gatekeeper or a proxy acts on behalf of the transferred > SET endpoint B. > > >> >However, media re-negotiation also should work here as > part of the > > >> fastStart > > >> >method. > > >> > > > >> >Regards, > > >> >Karl > > >> >------------------------------------------------ > > >> >Karl Klaghofer, Siemens AG, Dpmt. ICN IB NL IP7 > > >> >Hofmannstr. 51, D-81359 Munich, Germany > > >> >Tel.: +49 89 722 31488, Fax.: +49 89 722 37629 > > >> >e-mail: karl.klaghofer(a)icn.siemens.de > > >> > > > >> > > > >> > > > >> >> -----Ursprüngliche Nachricht----- > > >> >> Von: Gunnar Hellstrom [SMTP:gunnar.hellstrom@OMNITOR.SE] > > >> >> Gesendet am: Dienstag, 16. März 1999 23:01 > > >> >> An: ITU-SG16(a)mailbag.cps.intel.com > > >> >> Betreff: Call hold and transfer in H.323 AnnexF. > Too limited?? > > >> >> > > >> >> Dear multimedia experts. > > >> >> > > >> >> In my efforts to establish the simple IP voice and > text telephone Text > > >> >> SET, > > >> >> I came across a section in H.323 Annex F (Simple > Endpoint Type, TD11 > > in > > >> >> Monterey) that I feel is causing a functional > obstacle also to the > > >> voice > > >> >> users. Can anyone clarify if I am correct and why it > is specified the > > >> way > > >> >> it is. > > >> >> > > >> >> In section 7.6.1 and 7.6.2 it is specified:" The > Audio SET device > > >> shall > > >> >> then resume transmitting its media stream(s) to the transport > > >> address(es) > > >> >> newly indicated in the OpenLogicalChannel structures." > > >> >> I understand that this means that you cannot > re-negotiate audio > > coding, > > >> >> and > > >> >> you cannot add text conversation after rerouting the > call from a Voice > > >> >> only > > >> >> SET to a Text SET. > > >> >> > > >> >> Re-negotiating the audio coding will probably be a > desired function, > > >> e.g. > > >> >> when rerouting from a fixed to a wireless IP phone. > > >> >> Adding a data channel for text will also be a desired > function, after > > >> >> answering a call in an audio-only SET, and then > rerouting it to a > > >> >> text-capable SET. > > >> >> That action is very common in today's text telephone > usage, and I > > would > > >> >> expect it to be just as common in the IP telephony > world. You first > > >> >> receive > > >> >> the call in the terminal that is closest to you, and > then you get a > > >> reason > > >> >> to start text mode. Then you transfer the call to > another device with > > >> text > > >> >> capabilities, where you can switch mode. > > >> >> > > >> >> Questions: > > >> >> > > >> >> 1. Is that kind of call transfer that is handled by > the mechanisms in > > >> 7.61 > > >> >> and 7.6.2? > > >> >> > > >> >> 2. Are my conclusion right about the limitations? > > >> >> > > >> >> 3. Is this limitation a consequence of using Fast Connect? > > >> >> > > >> >> 4. Do you see any possibility to avoid the negative > effects of it - to > > >> >> make > > >> >> re-negotiation possible? > > >> >> > > >> >> 5. Is the specified functionality acceptable in the > voice world? If > > two > > >> >> devices have agreed on a voice coder, is it likely > that the third > > >> device > > >> >> supports it? Will this not create a lot of > unsuccessful call transfers > > >> >> where the users will have a no chance to understand > why they fail? > > >> >> > > >> >> ---- > > >> >> > > >> >> Another question area: > > >> >> > > >> >> 6. When selecting the transport protocol for the text > conversation, > > the > > >> >> current draft (APC 1504) specifies TCP or UDP. I > realize that there > > are > > >> >> situations where TCP must be avoided. One such situation is a > > >> sub-titled > > >> >> H.332 transmission. Also other multi-casting > situations is better off > > >> with > > >> >> a UDP based transport protocol. > > >> >> I am therfore now leaning towards using RTP as the > transport for text > > >> >> conversation. With RTP we can discover dropped frames > and possibly > > >> invent > > >> >> a > > >> >> mechanism to mark that event in the text stream for > T.140 to display. > > >> If > > >> >> we > > >> >> have less than 3 % dropped frames, I think the users > would accept it. > > >> >> > > >> >> 6.1 Do you agree that there are situations when TCP > should be avoided, > > >> and > > >> >> a UDP based protocol preferred? > > >> >> > > >> >> 6.2 Do you agree that RTP is a good alternative, with > a thin protocol > > >> for > > >> >> error indications to the user? > > >> >> > > >> >> 6.3 Most packets will carry only 1-4 characters . Can > anyone give me > > an > > >> >> indication on the expected packet loss rates in > different situations > > >> for > > >> >> such packets. Or a document giving such figures. Is > max 3% loss > > >> reachable? > > >> >> > > >> >> Please give your view on these questions. > > >> >> > > >> >> Best regards > > >> >> > > >> >> Gunnar Hellström > > >> >> ----------------------------------------------- > > >> >> Gunnar Hellstrom > > >> >> Representing Ericsson in ITU-T > > >> >> > > >> >> E-mail gunnar.hellstrom(a)omnitor.se > > >> >> Tel +46 751 100 501 > > >> >> fax +46 8 556 002 06 > ********************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This footnote also confirms that this email message has been swept by MIMEsweeper for the presence of computer viruses. www.mimesweeper.com **********************************************************************

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