Subject: RE: [Sipping] SIP Support for Hearing and Speech Impaired Users

From: "Arnoud van Wijk (ELN)" <[email-masked]>
Date: Tue, 31 Jul 2001 11:17:35 +0200

Hello Gonzalo.

Thanks for your feedback. When this draft was written, the call flows were probably not completely correct.
That is not a problem, the first goal was to give examples how such services could be implemented. Now we can continue and work out better call flows depending on what the best proposals and solutions are.

We have to start somewhere. :-)

I read the comments. And it is indeed a good point. In the call flows, it is seen as an send-only RTP stream. Thus this would mean that the caller in figure 2 would send only the audio to the relay TTS and receive the receive only RTP audio stream from the hearing impaired user. Perhaps I should make it more clear that the RTP streams are unidirectional and not bidirectional.

The question on what proposal to implement is something we need to look into it. I guess it depends on which proposal is now selected for conference bridges and on the effect on the relay services.

One question though, I asume that in figure 2, in case of 2 m= lines. That the hearing impaired user is actually sending the 2 invites? And the Hearing impaired user is actually sending one invite for himself (or herself) and one invite that is a copy of the SDP content sent by the Caller. 
maybe I am mixing up things, but if above is correct, then it must use 2 seperate invites and not one invite with 2 m= lines since each m= line is meant for a different user (and thus 2 different IP addresses)?

Cheers

Arnoud

_______________________________________________________________
ERICSSON EUROLABS NETHERLANDS BV
Arnoud van Wijk
Innovation Unit
Research & Development (ELN/V)
Fax: +31-161 247569
_______________________________________________________________

-----Original Message-----
From: Gonzalo Camarillo [mailto:[email-masked]]
Sent: maandag 30 jul 2001 23:15
To: Arnoud van Wijk (ELN)
Cc: [email-masked]'; Jonathan Rosemberg; Henning Schulzrinne; Cathy
Gearhart (EUS); Henry Sinnreich
Subject: Re: [Sipping] SIP Support for Hearing and Speech Impaired Users

Hello,

Here you have some feedback on draft-vanwijk-sipping-deaf-req-00.txt:

This draft contains some call flows that show how different services
required by hearing impaired people can be implemented. The call flow of
section 6.2 shows how to implement a one-way speech to text translation
service.

In this example the voice from the caller (A) has to be sent to the
human relay (speech to text translation S2T). The S2T will send text
over RTP to the callee (B). B, however, sends audio directly to A (B can
speak perfectly).

Besides the text over RTP session between S2T and B, this example
assumes one single audio session. A sends audio to S2T and B sends audio
to A over the same RTP session.

This would not work, because S2T does not have any address to send RTCP
reports. Besides, the RTCP reports sent from B to A would indicate that
there were no packets received, since A is sending RTP packets to S2T
instead.

This example should contain two audio sessions: one sendonly from A to
S2T and another reconly (from A's point of view) from B to A.

The same bug can be found in other sections of the draft. Section 6.3
also assumes that the SDP in the INVITE from B to the relay only
contains an m line (video stream) while it should contain the video line
and the audio line (for RTCP).

The same thing with section 6.5.

Once we agree that both lines are needed, let's see how we are going to
provide the relay with these two m lines:

Let's assume that we are a speech to text relay. We need to receive two
m lines: 1) an outgoing text over RTP m line
2) an incoming voice over RTP line

I made some time ago a proposal for this. It consisted of using the same
format as in a conference bridge. That is, several users INVITE the
conference bridge. This mixes all the incoming audio and sends it out.

Applying the same principle, I INVITE the S2T with an SDP containing one
RTP text line, and then I send another INVITE with the audio line. Both
INVITEs belong to the same conference (they are typically related using
the request-URI). Therefore, the S2T acts effectively as a conference
bridge between both streams.

Jonathan made another proposal. He suggested that everything could be
done in a single INVITE. The relay would know that it had to translate
the media received over one media line to the other one. While this
approach is less general (it just works for translators/transcoders), it
saves messages.

I guess you will choose the second approach, right? (any of them is OK
with me)

Best regards,

Gonzalo

"Arnoud van Wijk (ELN)" wrote:
> 
> I submitted the following draft:
> SIP Support for Hearing and Speech Impaired Users
> 
> SIP is attracting more and more attention as a valuable tool to enable and support voice and multimedia communications over the Internet. However, some users of SIP-based services will be unable or severely restricted to use plain voice communication. In particular, people who are hearing impaired often cannot send and/or receive voice. Its goal is to lay a foundation for design and implementation of SIP based services for Hearing and Speech Impaired users. These services are generally enabled by baseline SIP, or through the use of the caller preferences specification. No additional extensions are proposed here in order to support universal access.
> 
> It can be found at:
> 
> http://search.ietf.org/internet-drafts/draft-vanwijk-sipping-deaf-req-00.txt
> 
> Let me know what ye all think of it! And ofcourse everybody is welcome to bring additional requirements and possibilities.
> 
> Greetz
> 
> Arnoud van Wijk
> _______________________________________________________________
> ERICSSON EUROLABS NETHERLANDS BV
> Arnoud van Wijk
> Innovation Unit
> Research & Development (ELN/V)
> Fax: +31-161 247569
> _______________________________________________________________
> 
> _______________________________________________
> Sipping mailing list  http://www.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use [email-masked] for questions on current sip
> Use [email-masked] for new developments of core SIP

-- 
Gonzalo Camarillo                    Phone :   +1 212 939 71 71
Columbia University                  Mobile:  +358 40 702 35 35
472 Computer Science Building        Fax   :  +358  9 299 30 52
1214 Amsterdam Ave., Mail Code 0401  http://www.hut.fi/~gonzalo
New York, NY 10027                   
USA                              [email-masked]

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