Subject: PSTN emergency calls Was [Sipping] ToIP Requirements

From: "A vWijk" <[email-masked]>
Date: Mon, 08 Mar 2004 12:03:52 +0100

Hello Eric,
About the recap, see comments inline: (marked with **)

>So, to recap:

>PSTN TDD User gatewaying into SIP Network:

>PSTN TDD User 
   dials 600-911 
       ----> GW 
         GW issues INVITE sip:sos@mumble;caller_pref=tdd
            ----> PSAP ESP routes appropriately

** yes, the prefix 600 tells the PSTN that it is a TDD calling and that the text gateway must be used for T.140/RTP transcoding.
the ESP routes are then exactly for IP telephony with RFC 2793 interactive text as media. (instead of voice for most users, unless placing a call while burglars are in the room next to yours :-)).
Any other solution that allows the PSTN to separate a voice call and a TDD/analogue textphone call can alo be used instead the prefix. That is also why the prefix is in this example and has no other function.

>IP TDD User gatewaying into PSTN:

>IP TDD User 
   INVITE sip:sos@mumble;caller_pref=tdd 
      ----> Proxy 
         Proxy issues INVITE tel:600911@gw 
             ----> GW
                GW dials 600-911

** first of all, it is not an IP TDD User but an Interacitive text user, it can be any user who prefers text here. It can be a deaf person who uses Interactive text replacing a TDD.
Second, why would you when being on IP move back to the PSTN to place an analogue 600-911 call? 
And from IP to PSTN you do NOT need a prefix number. The text gateway knows that text is needed thanks to the m and a lines in SDP. (showing T.140).

** The great thing that this allows is that every interactive text emergency call that comes in at the emergency center (911 in USA) arrives as T.140/RTP regardless it is then a native IP caller or through a gateway at the PSTN to transcode TDD into Interactive text. There is no longer a need for a special analogue text telephony emergency call system. 

greetz

Arnoud

> -----Original Message-----
> From: A vWijk [mailto:[email-masked]] 
> Sent: Monday, March 01, 2004 1:00 AM
> To: [email-masked]; [email-masked];
> [email-masked]; [email-masked]; [email-masked] 
> Cc: [email-masked]; [email-masked]; 
> [email-masked];
> [email-masked]; [email-masked] 
> Subject: Betr.: RE: Betr.: RE: Betr.: [Sipping] ToIP Requirements
> 
> 
> Hi Brian,
> I am only talking to analogue text telephony to IP telephony 
> on the PSTN.
> In the PSTN, you do not use a SIP URI for placing a call. You 
> would dial 911 if it was a voice phone or perhaps prefix 600-911. 
> 
> The text gateway will send the Text media announcement in the 
> SDP. And send it to the sos@localdomain URI. I agree with you 
> on the sip side. The text gateway acts then as the "virtual" 
> SIP version of the analogue text telephone, using RFC2793 text.
> 
> Perhaps we are miscommunicating what we are talking about. :-)
> 
> greetz
> 
> Arnoud
> 
> 
> <<< "Rosen, Brian" <[email-masked]> 01-03 06:03  >>>
> Arnoud
> 
> With respect to prefixes:
> In SIP calls, there are no numbers, and thus no prefixes.
> You send the Text media announcement in the SDP.
> You send it to the sos@localdomain URI.  We don't need, nor want
> prefixes
> 
> In the existing dial plan, some jurisdictions have special
> numbers for text, some don't.  We can't change that.
> We have to work with both ways, but we don't advocate,
> support or discourage them; we simple deal with the
> possibility that a user might use those numbers.
> 
> If you want prefixes for every call, speak to your
> local number adminstration agency; it has nothing
> to do with SIP.
> 
> Brian
> 
> > -----Original Message-----
> > From: A vWijk [mailto:[email-masked]] 
> > Sent: Sunday, February 29, 2004 11:47 PM
> > To: [email-masked]; [email-masked];
> > [email-masked]; [email-masked]; 
> [email-masked] 
> > Cc: [email-masked]; [email-masked]; 
> > [email-masked];
> > [email-masked]; [email-masked] 
> > Subject: Betr.: RE: Betr.: [Sipping] ToIP Requirements
> > 
> > 
> > And that is why those 2 solutions would be best IMHO:
> > 1. using a prefix number when using a textphone.
> > This prefix is for every textphone call you place! Thus not 
> > emergency only but any call made by the textphone.
> > 600-911, 600-555-123456, 600-800-cheapshoe.
> > 
> > 2. Label the phone number you use for the textphone as text 
> > telephone user.
> > 
> > In both cases the PSTN will immediately invoke the text 
> > gateway. Which then sends the V.18 reception/boudot reception 
> > signalling.
> > 
> > I know that it is not perfect, but the analogue textphones 
> > will disappear quickly. 2-3 years for most users and up to 
> > 10-15 years for the last few users.
> > 
> > I and many deaf people I know will throw the analogue 
> > textphone out of the window as soon as possible.
> > We will use mobile textphones via the textgateway and SIP 
> > phones with RFC2793 (and our pcs with an interactive text client).
> > 
> > greetz
> > 
> > Arnoud
> > 
> > 
> > <<< "Gunnar Hellstrom" <[email-masked]> 01-03 
> 03:27  >>>
> > 1. How it is in PSTN:
> > A calling V.18 sends a specific data pattern for text 
> > telephony in the V.8
> > signal CI.
> > It is sent in the connected voice channel.
> > So, as with all calls from PSTN, there is no indication 
> > before the call is
> > set up about what kind of media will be used.
> > 
> > But it is far more easy to identify it as a textphone call 
> > than it is for
> > some of the old systems. E.G the US system "Baudot" is silent 
> > when calling,
> > and the answering side must guess that it is a textphone.
> > 
> > There is not yet any really clear trend to move to V.18 for 
> > users in USA.
> > Therefore I agree that the mechanisms for emergency calls 
> > must for a long
> > time accomplish for the tedious task to try to respond in 
> > text on PSTN to
> > all incoming silent calls to discriminate the text callers. 
> > That response
> > should be done with V.18 signals, so that it at least is 
> > prepared for more
> > convenient handling when people have moved to using V.18 textphones.
> > 
> > 2. In SIP.
> > In SIP, we have text in a separate media stream in the call. 
> > An emergency
> > centre should have capability to use both voice and text 
> > simultane
> 
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Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands.
Mobile: +31651921948
International text telephone: +31735588408

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use [email-masked] for questions on current sip
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