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H.GCP / Megaco Audio call
by Bryan Hill 08 Jun '99

08 Jun '99

In order to continue the cooperation between SG16 and the IETF, there will be another Megaco/H.gcp conference call so that we have rough consensus on what Brian Rosen will put into megaco-protocol-02 (Internet-Draft cutoff date for IETF Oslo meeting is June 25/99). In Megaco we have agreed to produce a completed audio-only version of the megaco protocol for June 25th, but that this version would take into consideration the proposals for multimedia in SG16's H.gcpSantiago so that they can easily be added in a future version. We need to have an "implementable" version of the Megaco protocol by the July Oslo meeting, or we will have missed the boat. It is expected that the megaco-protocol-02 would be an implementable audio version of the Megaco/H.gcp protocol, and that SG16 would be able to add to the protocol syntax, in a backwards compatible way, all that is necessary for multimedia. Proposed discussion topics: 1) Termination naming - hierarchical or logical, or a combination with TerminationId + Bearer? 2) handling multplexing terminations - as in H.gcp? or as a new context type? 3) allowing one BearerDescriptor, and multiple MediaDescriptors per termination - acceptable? 4) decision on transport 5) proposal for tagging the MediaDescriptor with its encoding: SDP or H.GCP 6) proposal for text vs binary encoding of the protocol (draft-huitema-megaco-sdp-discuss-01.txt) 7) multimedia vs monomedia contexts (related to #3) Call Details Date: Wednesday, June 16th Time: 11amET for 2 hours Phone: 613-763-6338 PassCode: 131313# Problems getting on the bridge? call 613-765-CONF, and ask for the conference chaired by Nancy Greene. As before, minutes of the meeting will be posted ASAP after the call. Nancy Greene Bryan Hill ngreene(a)nortelnetworks.com <mailto:ngreene@nortelnetworks.com> bhill(a)videoserver.com <mailto:bhill@videoserver.com> -------------------------------------------------------------------------- Nancy M. Greene Internet & Service Provider Networks, Nortel Networks T:514-271-7221 (internal:ESN853-1077) E:ngreene@nortelnetworks.com

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H.GCP is about gateway recomposition, not decomposition.. (attached)
by Boaz Michaely 06 Jun '99

06 Jun '99

Sorry about that attachement. (don't you love NOTES too ?) Hope this works : (See attached file: H.GCP composition.doc) - Boaz "John Talbot" <jptalbot(a)nortelnetworks.com> on 03/06/99 07:08:47 PM To: Boaz Michaely cc: Subject: RE: H.GCP is about gateway recomposition, not decomposition.. Michaely, Did you realy mean to send the image of the icon of the document rather than the document itself with your email? Anyway I read your document and have included my comments :-) <<pic10782.pcx>> Cheers John > -----Original Message----- > From: Boaz Michaely [SMTP:Boaz_Michaely@VOCALTEC.COM] > Sent: 03 June 1999 13:46 > To: ITU-SG16(a)MAILBAG.INTEL.COM > Subject: H.GCP is about gateway recomposition, not decomposition.. > > Dear Brian, > I am documenting a piece on the "minimal interoperability" issue, which I > will forward to you next week. (if there is interest I can post it to the > list) > > The basic idea is that while the motivation for H.GCP was gateway > decomposition, the actual standard deals with composing rather than > decomposing. > > Thus I propose to introduce some terms, such as "composite gateway", while > adhering to H.323 current definitions, and work on the call setup > procedures, closely following the actual H.323 text. This will lead to the > resolution of transfering call signalling between the MGC and the MG in > the > various compositions. One thought that crossed my mind was to explicitly > deal with all the gateway categories that H.323 defines ( H.310 (H.320 on > B-ISDN), H.320 (ISDN), H.321 (ATM), H.322 (GQOS-LAN), H.324 (GSTN), H.324M > (Mobile), and V.70 (DSVD)). > > We may also need to modify H.323v3, at least to some extent, in order to > better cover gateway sub-components. > > A sample of the text is attached (Word 97) > > (Embedded image moved to file: pic10782.pcx) > > Boaz. > > Boaz Michaely > Senior Program Manager, Advanced Technologies > VocalTec Communications Ltd > e-mail boaz(a)vocaltec.com > Tel +972 (9) 970-7869

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Santiago meeting documents
by Sakae OKUBO 04 Jun '99

04 Jun '99

Dear Q.12-14/16 experts, All the files uploaded to the /Incoming directory of the PictureTel ftp site have been moved to the /9905_San directory. They include Mr. Ott's Annex F to H.323 and Mr. Duckworth's H.283. The index is given by the following: http://standard.pictel.com/ftp/avc-site/9905_San/9905_San.html For your convenience, the index file is attached to this message. Please Editors upload your Recommendations in charge as soon as your editing is completed. Best regards, Sakae OKUBO (Mr.) *********************************************************** Waseda Research Center Telecommunications Advancement Organization of Japan (TAO) 5th Floor, Nishi-Waseda Bldg. 1-21-1 Nishi-Waseda, Shinjuku-ku, Tokyo 169-0051 Japan Tel: +81 3 5286 3830 Fax: +81 3 5287 7287 e-mail: okubo(a)giti.or.jp ***********************************************************

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Text Telephone support in megaco and H.gcp
by Gunnar Hellstrom 04 Jun '99

04 Jun '99

It seems to be useful to structure Text Telephone support in gateways in three levels: 1. Simple text-only support. ---------------------------- There is a simple way to use a textphone. That is to connect with the intention from the beginning to get a pure text call established and stay in the text-only mode during the whole call. Many users use this mode only, and it is also used in many cases in connection to relay services and emergency services. (relay services are services where the users get help to translate between different modes in the conversation, most commonly between voice telephony and text telephony. Relay services are usually manned services.) 2. Alternating text and voice support. -------------------------------------- Text telephone suppport can be quite advanced, with switching back and forth between text and voice during the call. This mechanism is used with current textphones that were created long before any simultaneous voice-data technologies were invented. Ideally, this support would also be wanted on any gateway termination, monitoring the line during voice operation for textphone tones and at that moment initiate text mode. 3. Simultaneous text and voice call. ------------------------------------ Alternative 2 is quite cumbersome for the users. A simple way to use simultaneous voice and text is eagerly needed. It is likely that the H.323 Annex G Text Conversation and Text SET will make this happen as the normal thing in the IP environment. But will it also eventually evolve in the PSTN? Within V.18 there is an opportunity to use V.61 for text telephony with voice and text. H.324 (especially without video) is also a possible environment. Market forces may tell if this enhancement is wanted in the PSTN. The Total Conversation terminals with simultaneous Video, Text and Voice should be placed in the simultaneous voice-text level when discussing text telephone interworking. Since Total Conversation terminals already exist in the H.320 invironment in ISDN, we have them in SCN as well as in IP networks. ----------- Termination arrangement for text telephone support. --------------------------------------------------- Level 1. -------- Level 1 can be achieved with a simple termination. On the SCN side it would contain V.18 modem support and terminate a T.140 text media stream. ( see eartlier message in the lists) Level 2. -------- If level 2 - alternating voice - text is going to be supported, it is advisable to have a model for how it should be viewed in terms of terminations, contexts and signaling. Multiplexor model: One model would be to regard the termination to contain a kind of multiplexor. It would then terminate two media streams - one T.140 text stream and one audio stream. The mechanisms to connect either stream to the line would be an internal mechanism in the termination. The need for external signaling in this case is low. Another model would be to regard the termination to terminate only one type of stream at a time. It would need to report events, e.g. about the carrier dropping, so that the voice mode can be started by an MGC command. A whole set of detailed interactions seems to be needed, between the MGC and the MG for the purpose of controlling the termination to follow the user's expectations for mode changes. I prefer the multiplexor model for the alternating voice - text mode. Some thoughts should be given to it, to verify if decent support of user requirements can be achieved. Level 3. -------- Level 3, simultaneous voice and text, is seen on the IP side as one T.140 TCP termination and one RTP audio termination. On the SCN side it is seen as a termination according to the kind of multiplexor it uses. Conclusions: ------------ I see no problem in mapping this structure within the H.gcp . I just want the mechanisms to be agreed on and the details around specific termination parameters and MG behaviour. It may be useful to take part of the discussion above into the megaco requirement spec in order to clarify what is meant with text telephone support and text conversation support. Regards Gunnar Hellström At 07:51 1999-06-04 -0600, Don Stanwyck wrote: >Gunnar Hellstrom wrote: >> >> Text Telephone support is very similar to FAX. If we let FAX into gateway >> specifications, we shall also include text telephony and text conversation >> in the same move. It saves a lot of efforts. > >Actually, text telephone (e.g., TDD) is quite a bit different from FAX, and >in the US, at least, mandated to be supported in every network (ADA >requirements). TDD conversations can be one-way modem one-way voice, >two-way modem, part-time modem (what TDD users call voice-over mode), etc. >My daughter is a TDD user, and we have become quite familiar with it. The >difficulties it presents to IP-telephony are considerable, but can be >solved by allowing that any line, any time, can be instantly converted to a >modem communication line without any prior notice or any promise that modem >tones will continue on the line for any particular duration. > >Don >Attachment Converted: "c:\_Omnitor\Attach\stanwyck.vcf" > ----------------------------------------------- Gunnar Hellstrom LM Ericsson E-mail gunnar.hellstrom(a)omnitor.se Tel +46 751 100 501 fax +46 8 556 002 06

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Re: Decided and Determined texts
by Kumar, Vineet 04 Jun '99

04 Jun '99

I found H323AnnexF-decided.zip in http://standard.pictel.com/ftp/avc-site/9905_San/ vineet -----Original Message----- From: Kumar, Vineet [mailto:vineet.kumar@INTEL.COM] Sent: Friday, June 04, 1999 10:16 AM To: ITU-SG16(a)MAILBAG.INTEL.COM Subject: Re: Decided and Determined texts I was expecting to find H323AnnexF-decided.zip at http://standard.pictel.com/ftp/avc-site/Incoming/ but I don't see it there yet. vineet -----Original Message----- From: Greg Herlein [mailto:gherlein@HERLEIN.COM] Sent: Friday, June 04, 1999 2:12 AM To: ITU-SG16(a)MAILBAG.INTEL.COM Subject: Re: Decided and Determined texts > I have uploaded the decided clean text version of H.323 Annex F > into the Incoming directory of the PictureTel ftp site. The file > name is H323AnnexF-decided.zip Can someone give me a full URL for this document? I can't find it by browsing around... Thanks, Greg /************************************************************** Greg Herlein gherlein(a)herlein.com Herlein Engineering www.herlein.com Software Development - Communications - Networking - Linux/Unix ***************************************************************/

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Re: Decided and Determined texts
by Kumar, Vineet 04 Jun '99

04 Jun '99

I was expecting to find H323AnnexF-decided.zip at http://standard.pictel.com/ftp/avc-site/Incoming/ but I don't see it there yet. vineet -----Original Message----- From: Greg Herlein [mailto:gherlein@HERLEIN.COM] Sent: Friday, June 04, 1999 2:12 AM To: ITU-SG16(a)MAILBAG.INTEL.COM Subject: Re: Decided and Determined texts > I have uploaded the decided clean text version of H.323 Annex F > into the Incoming directory of the PictureTel ftp site. The file > name is H323AnnexF-decided.zip Can someone give me a full URL for this document? I can't find it by browsing around... Thanks, Greg /************************************************************** Greg Herlein gherlein(a)herlein.com Herlein Engineering www.herlein.com Software Development - Communications - Networking - Linux/Unix ***************************************************************/

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Decided and Determined texts
by Sakae OKUBO 04 Jun '99

04 Jun '99

Dear Editors, Please upload integrated texts for the subject Recommendations to the PictureTel ftp site Incoming directory at the earliest occasion. Best regards, Sakae OKUBO (Mr.) *********************************************************** Waseda Research Center Telecommunications Advancement Organization of Japan (TAO) 5th Floor, Nishi-Waseda Bldg. 1-21-1 Nishi-Waseda, Shinjuku-ku, Tokyo 169-0051 Japan Tel: +81 3 5286 3830 Fax: +81 3 5287 7287 e-mail: okubo(a)giti.or.jp ***********************************************************

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Re: unsubscribe
by NieXiuYing 03 Jun '99

03 Jun '99

unsubscribe

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Re: List Of Termination Prop... Text Telephone Support
by Gunnar Hellstrom 03 Jun '99

03 Jun '99

In order to give an example of why it is of importance for the MGC to know what type of modem connection that was activated, I provide a first draft of how text telephone support can be explained in H.gcp . In this case the modem will indicate V.18 usage, and that means that the media stream will be a T.140 text stream. In the T.38 case it will be some kind of fax data stream, while for the V.90 case, I hope that the idea would be to use it for a PPP session to access the Internet. So, these three modem types need quite different actions from the gateway and it need to be reflected in the MGC interface. Please look in the attached proposed new section to H.gcp for further discussion and explanation. Can anyone look into how to structure alternating between voice and text on the SCN side during a call and leading text to the text stream and voice to the audio stream on the IP side? ( called "voice over" in USA ) Can we support it? Regards Gunnar Hellström At 11:05 1999-06-03 -0700, you wrote: >At 12:19 PM 6/3/99 -0500, Nancy-M Greene wrote: >>> > >Modem type: none, V.90, T.38, V.18, ... as applicable. >>>> Why is this passed from MGC to MG? What does the MG do with it? Isn't >>>> this an MG capability? Maybe this is only useful for audit. >> >>>[[TomT]] This information is probably only useful for H.324 calls, although >>>maybe it also applies to NAS operation. The MGC knows which modem type to >>>assign based on dialled number, and passes the information to the MG to act >>>upon. >> >> >>Yes, this information is useful for NASes. In the SS7 case, the MGC knows >>which physical circuit the call will be completed on, and based on the >>dialled number, may have modem characteristics that need to be given to the >>NAS (MG) before that call is completed. > >As has been discussed quite a bit before, this is AAA info and NAS's >already have a protocol for that. We don't need to reinvent the wheel. > > ----------------------------------------------- Gunnar Hellstrom LM Ericsson E-mail gunnar.hellstrom(a)omnitor.se Tel +46 751 100 501 fax +46 8 556 002 06

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H.GCP is about gateway recomposition, not decomposition..
by Boaz Michaely 03 Jun '99

03 Jun '99

Dear Brian, I am documenting a piece on the "minimal interoperability" issue, which I will forward to you next week. (if there is interest I can post it to the list) The basic idea is that while the motivation for H.GCP was gateway decomposition, the actual standard deals with composing rather than decomposing. Thus I propose to introduce some terms, such as "composite gateway", while adhering to H.323 current definitions, and work on the call setup procedures, closely following the actual H.323 text. This will lead to the resolution of transfering call signalling between the MGC and the MG in the various compositions. One thought that crossed my mind was to explicitly deal with all the gateway categories that H.323 defines ( H.310 (H.320 on B-ISDN), H.320 (ISDN), H.321 (ATM), H.322 (GQOS-LAN), H.324 (GSTN), H.324M (Mobile), and V.70 (DSVD)). We may also need to modify H.323v3, at least to some extent, in order to better cover gateway sub-components. A sample of the text is attached (Word 97) (Embedded image moved to file: pic10782.pcx) Boaz. Boaz Michaely Senior Program Manager, Advanced Technologies VocalTec Communications Ltd e-mail boaz(a)vocaltec.com Tel +972 (9) 970-7869

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