Internet DRAFT - draft-hellstrom-txtgwy

draft-hellstrom-txtgwy






Network Working Group                                       G. Hellstrom
Internet-Draft                                                   Omnitor
Intended status: BCP                                           B. Dingle
Expires: September 7, 2010
                                                             A. van Wijk
                                         Real-Time Text Taskforce (R3TF)
                                                               G. Gybels
                                                 RNID, Department of New
                                                            Technologies
                                                           March 8, 2010


        Real-time text interworking between PSTN and IP networks
                       draft-hellstrom-txtgwy-02

Abstract

   IP networks can support real-time text communication.  SIP-based
   real- time text is called Text-over-IP or ToIP.  PSTN networks
   support real-time text using textphones (or TTYs).  When real-time
   text is supported by different networks, gateways are needed to
   provide interoperability.  Real-time text capable gateways may also
   support real-time voice.

   This specification describes procedures for interworking between ToIP
   and PSTN textphones using a real-time text capable gateway (RTT
   gateway).  It also describes ways to route calls to RTT gateways for
   several call scenarios.

   Procedures that support the phased introduction of RTT gateways and
   procedures that support the invocation of text channels at any time
   during the call are included.  Interworking of PSTN textphones that
   do not support simultaneity of voice and text with IP User Agents
   that support simultaneous voice and text is also described.

Status of this Memo

   This Internet-Draft is submitted to IETF in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
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   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference



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   material or to cite them other than as "work in progress."

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   This Internet-Draft will expire on September 7, 2010.

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   described in the BSD License.



























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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.1.  Abbreviations  . . . . . . . . . . . . . . . . . . . . . .  5
     2.2.  Definitions  . . . . . . . . . . . . . . . . . . . . . . .  5
   3.  Functional goals of the procedures . . . . . . . . . . . . . .  5
   4.  Scope  . . . . . . . . . . . . . . . . . . . . . . . . . . . .  5
   5.  Locating RTT gateways  . . . . . . . . . . . . . . . . . . . .  5
     5.1.  Types of RTT gateways  . . . . . . . . . . . . . . . . . .  5
     5.2.  Locating a gateway . . . . . . . . . . . . . . . . . . . .  6
       5.2.1.  From the IP side . . . . . . . . . . . . . . . . . . .  6
       5.2.2.  From the PSTN side . . . . . . . . . . . . . . . . . .  6
   6.  SIP Call control . . . . . . . . . . . . . . . . . . . . . . .  7
     6.1.  SIP and SDP text indicators  . . . . . . . . . . . . . . .  7
     6.2.  SIP/SDP Call procedures  . . . . . . . . . . . . . . . . .  8
       6.2.1.  Calls from the PSTN  . . . . . . . . . . . . . . . . .  8
       6.2.2.  Call from a ToIP User Agent  . . . . . . . . . . . . . 10
     6.3.  Procedure for RFC 4103 and other real-time text
           transports . . . . . . . . . . . . . . . . . . . . . . . . 12
   7.  Text medium interworking . . . . . . . . . . . . . . . . . . . 12
     7.1.  Handling of alternating text and audio . . . . . . . . . . 13
       7.1.1.  Non-continuous carrier PSTN textphones . . . . . . . . 14
       7.1.2.  Continuous-carrier PSTN textphones . . . . . . . . . . 14
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 14
   9.  Security Considerations  . . . . . . . . . . . . . . . . . . . 14
   10. Normative References . . . . . . . . . . . . . . . . . . . . . 14
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16























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1.  Introduction

   Real-time text can be a component in IP multimedia telephony and
   total conversation.  Real-time text can be transported in IP networks
   using standard IP protocols and be used as a medium in a
   conversational service.  IP devices such as multimedia telephones,
   voicemail systems, IP phones, IVR systems or other devices, may
   support the transmission of real-time text over IP networks.  An IP
   User Agent that supports real-time text over IP is called a ToIP User
   Agent.  A ToIP User Agent may also support text and voice.

   The control of IP real-time text calls is defined in SIP RFC 3261
   [RFC3261] and SDP RFC 4566 [RFC4566].  RFC 4103 [RFC4103] specifies
   the carriage of real-time text in RTP packets between ToIP devices in
   IP networks.  RFC 5194 [RFC5194] describes the implementation aspects
   of ToIP using SIP.

   PSTN networks can support the transport of real-time text by
   representing the text as audio.  PSTN textphones (or TTYs) use modem
   technology to carry the text encoded as tones.  Some PSTN textphones
   can support the transport of both voice and real-time text, but
   usually not both at the same time.  Some PSTN textphone protocols do
   not include signalling to indicate whether or not the device is in
   text or voice mode and therefore it is unclear if the audio signal is
   to be treated as text or as voice.

   Interworking between the different forms of real-time text transport
   and the different call control protocols requires a real-time text
   capable gateway (RTT gateway).  It supports ToIP (and possibly VoIP)
   protocols on the IP side and textphone protocols on the PSTN side.

   PSTN textphones and multimedia IP User Agents usually support the
   transport of both voice and real-time text.  For calls that support
   both voice and text media, the gateway needs to consider the lack of
   media simultaneity imposed by some PSTN textphone protocols and
   include procedures to support medium interworking.

   The case where ToIP support is not provided at all PSTN/IP gateways
   that could be selected to convey the call is also considered.  It
   requires specific routing actions for calls that may contain text.


2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].




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2.1.  Abbreviations

   RTT real-time text

   ToIP text over IP

   PSTN public switched telephone network

2.2.  Definitions

   TTY: term used in some countries for PSTN textphone.


3.  Functional goals of the procedures

   The procedures described in this specification are designed to meet
   the following functional requirements.
   o  The real-time text transport in the IP network shall use ToIP as
      specified by RFC 5194 [RFC5194].
   o  The text medium shall be able to be established at any time during
      the call for gateways that support text and voice.
   o  The voice medium shall be able to be established at any time
      during the call for gateways that support text and voice.


4.  Scope

   This specification describes the procedures for call routing, call
   control and media transport of real-time text calls between Text-
   over- IP (ToIP) User Agents and PSTN textphones (or TTYs) using real-
   time text capable (RTT) gateways.  It specifies how to discover the
   RTT gateways from the PSTN and IP sides and how to invoke the text
   capability in them.  It also specifies how to control media
   interaction between PSTN and IP for calls that involve both real-time
   text and voice.


5.  Locating RTT gateways

5.1.  Types of RTT gateways

   Text capable gateways can be located in at least the following
   distinctively different locations:

   1.  a residential text capable gateway with PSTN terminals directly
       connected to it





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   2.  a network text capable gateway that has PSTN technology on one
       side and IP technology on the other
   3.  a network text capable gateway that is in the IP network that has
       text transported to it from the PSTN that is still in audio
       (modem tones) format but carried by IP transport (e.g.  G.711
       [RFC3551] A-law encoded audio) and with high QoS.

5.2.  Locating a gateway

5.2.1.  From the IP side

   A call to a PSTN number will require the use of a gateway.  If not
   all PSTN/IP gateways are text capable, then some means of routing to
   a RTT gateway is required.  The methods resulting in one step calling
   should be preferred.  The following mechanisms are possible:

   Option 1: Indications are used in the SIP header to indicate that
   text capability may be required.  RFC 3840 [RFC3840] and RFC 3841
   [RFC3841] provide a means of doing that.

   Option 2: ENUM RFC 3761 [RFC3761] procedures may be used to identify
   and route to RTT gateways.

   Option 3: Addressing is used consisting of the PSTN number as the
   user name and the RTT gateway as the SIP domain address on the form
   number@gateway_domain.

   Option 4: Dial a special RTT gateway address.  The gateway answers in
   text mode.  Then enter the destination number when requested by the
   RTT gateway using text.

5.2.2.  From the PSTN side

   If a PSTN textphone is not directly connected to a RTT gateway or if
   not all PSTN/IP gateways that are candidates for handling the call
   are text capable, then some means of routing to a RTT gateway is
   required.  Most PSTN textphone calls appear as ordinary audio
   telephone calls and do not provide an indication that a call could
   involve text.  A means of indicating the possible need to support
   real-time text to the PSTN routing procedures is required.  Several
   options are available, the ones resulting in a one-step procedure
   should be preferred

   Option 1 - Use a prefix to destination number (e.g. 18001 +
   'destination number')

   Option 2 - Dial a special RTT gateway address.  The gateway answers
   in text mode.  Then enter the destination number or address when



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   requested by the RTT gateway using text.

   Option 3 - A PSTN line marking of 'text' that is carried by the
   signalling

   Option 4 - The call routing made after number analysis that results
   in that a call to a real-time text user is routed through the text
   gateway located in the IP network.

   Option 5 - The destination SIP subscriber has an indication on SIP
   registration level or call control level that text support is
   required.


6.  SIP Call control

6.1.  SIP and SDP text indicators

   SIP User Agents and RTT gateways SHALL announce support for real-time
   text in the Contact field according to RFC 3840 [RFC3840] by sending
   the following in an INVITE:

      Contact: <uri>;text

   A calling ToIP User Agent that does not want to give priority to text
   can indicate an interest to use text according to RFC 3841 [RFC3841]
   by sending the following in a Response to an INVITE:

      Accept-Contact: *;text

   A calling ToIP User Agent should indicate priority to establishing a
   text connection according to RFC 3841 [RFC3841] by sending the
   following in a Response to an INVITE:

      Accept-Contact: *;text;require

   PSTN textphones can support alternating text and audio during the
   call.  If this capability is known, the RTT gateway can indicate it
   according to RFC 3840 [RFC3840] by sending an indication in the call
   setup as specified in draft-hellstrom-text-turntaking
   [I-D.hellstrom-text-turntaking].

   This indication will allow ToIP User Agents to inform users of the
   need to communicate in a turn-taking manner.







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6.2.  SIP/SDP Call procedures

   The decision to include text when offering the call may be because:

   a. textphone tones have already been detected by the gateway on the
   PSTN line.

   b. the gateway may be configured to be a dedicated RTT gateway.

   c. it is known by subscription or other external means that the PSTN
   user has preference for text

   d. it is simpler to offer text initially.

   Note - the examples below do not include all SIP and SDP fields.
   They concentrate on the fields needed for ToIP operation and VoIP
   operation (if required).

   The audio medium description in the examples assumes ITU-T G.711
   A-law encoding.

6.2.1.  Calls from the PSTN

6.2.1.1.  Text-only call

   When a RTT gateway has information that indicates that only a text
   medium is required, it SHALL include specifications in line with this
   example in an INVITE:

      Contact: <sip:gw-uri>;text
      ...
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98

   The answering ToIP User Agent SHALL accept the text medium by
   including specifications according to this example in its Response:

      Contact: <sip:term-uri>;text
      ...
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98






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6.2.1.2.  Call with text and voice

   If the RTT gateway has an indication that voice and text media are
   required in the call, it SHALL include a media line for audio and a
   media line for text in an INVITE in line with the following example:

      Accept-Contact: *;text;require
      Contact: <sip:gw-uri>;text
      ...
      m=audio 7200 RTP/AVP 0
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98

   The answering ToIP device can accept the audio and the text media by
   sending the following in the Response:

      Contact: <sip:term-uri>;text
      ...
      m=audio 7200 RTP/AVP 0
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98

6.2.1.3.  Voice call with text added later

   A RTT gateway that wishes to establish an audio medium and indicate
   support for text but does not wish to establish a text medium
   immediately e.g. in order to avoid spending resources for text
   transport on calls that do not carry text at all, SHOULD send an
   INVITE with contents in line with the following example:

      Contact: <sip:gw-uri>;text
      ....
      m=audio 7200 RTP/AVP 0

   The answering ToIP device can indicate text support and accept the
   audio medium by including lines in line with the following example in
   the Response to the INVITE:

      Contact: <sip:term-uri>;text
      .....
      m=audio 7200 RTP/AVP 0

   Then the text medium can be added later in the call.




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6.2.2.  Call from a ToIP User Agent

6.2.2.1.  Text-only call

   A calling ToIP User Agent can indicate that a text medium is required
   in the call by sending an INVITE based on the following example:

      Contact: <sip:term-uri>;text
      Accept-Contact: *;text;require
      ...
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=fmtp:98 cps=20
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98
   The network will route the call to a RTT gateway if the destination
   address is in the PSTN and the Contact indicates 'text'.  The RTT
   gateway should then commence call establishment procedures on the
   PSTN side.

   The RTT gateway can accept the text request by sending the following
   in the Response:

      Contact: <sip:gw-uri>;text
      ....
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=fmtp:98 cps=20
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98

6.2.2.2.  Text and voice call

   A calling ToIP User Agent can request a text medium and a voice
   medium in a call by including a media line for audio and a media line
   for text in an INVITE as follows:

      Contact: <sip:term-uri>;text
      Accept-Contact: *;text;require
      ...
      m=audio 7200 RTP/AVP 0
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=fmtp:98 cps=20
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98

   The network will route the call to a RTT gateway if the destination



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   address is in the PSTN.

   The answering RTT gateway can accept the text medium and the audio
   medium by sending the following in the Response:

      Contact: <sip:gw-uri>;text
      Accept-Contact: *;text;require
      ...
      m=audio 7200 RTP/AVP 0
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98

6.2.2.3.  Voice call with text added later

   If a calling ToIP device user is not depending on text, but wants to
   have it available for occasional use, the INVITE that is sent could
   include the following:

      Contact: <sip:term-uri>;text
      Accept-Contact: *;text
      ...
      m=audio 7200 RTP/AVP 0
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=fmtp:98 cps=20
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98

   The RTT gateway may decide to not establish the text channel
   initially, but should indicate its text capability in the Contact
   header of the Response as follows:

      Contact: <gwy-uri>;text
      Accept-Contact: *;text
      ....
      m=audio 7200 RTP/AVP 0
      m=text 0 RTP/AVP 99 98

   During a call, if either party starts text activity, a text channel
   can be added to the session by sending a re-Invite according to RFC
   3261 [RFC3261].  This procedure can be executed even if no text
   capability was expressed from the ToIP User Agent initially.  A re-
   INVITE from the gateway could include the following:






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      Contact: <uri>;text
      ...
      m=audio 7200 RTP/AVP 0
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=fmtp:98 cps=20
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98

   The receiving device should answer with a Response that includes the
   following:

      Contact: <uri>;text
      ...
      m=audio 7200 RTP/AVP 0
      m=text 7202 RTP/AVP 99 98
      a=rtpmap:98 t140/1000
      a=fmtp:98 cps=20
      a=rtpmap:99 red/1000
      a=fmtp:99 98/98/98

6.3.  Procedure for RFC 4103 and other real-time text transports

   In the case where a User Agent implements both RFC 4103 [RFC4103] and
   other codecs for text transport, the procedures in this specification
   can be complemented with the other transport establishment.  When
   establishing a call, a ToIP device may advertise support for real-
   time text in accordance with other transport methods, and for real-
   time Text over IP according to the procedures described above.

   The capabilities of the other party in the call setup will lead to
   the establishment of text transport according to either RFC 4103
   [RFC4103] or the other transport method, or both.


7.  Text medium interworking

   Many PSTN textphones can support both voice and text on the one
   analog 'voice' channel.  The voice and the text make use of the
   channel in an alternating fashion.  This direction dependence is
   however a usage convention, while the transmission can be made in
   either direction.  In most cases, voice is used in one direction and
   text in the other direction, for example a hearing impaired user
   talks to an relay operator, who then replies in text.

   ITU-T V.18 V.18 [V.18] specifies a range of PSTN textphone protocols
   that could be supported.  On the IP side, text and voice are carried
   in separate text and voice media.



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   The RTT gateway must be able to:

   a) indicate to the ToIP User Agent if the PSTN textphone operates in
   an alternating text and voice mode so that the IP User can be
   informed and communicate appropriately, and

   b) indicate to the ToIP User Agent if turntaking of text in one
   direction and text in the other direction is required by the PSTN
   textphone.  The RTT gateway shall decode the characters received and
   transmit them to the ToIP User Agent.

   Determining the type of PSTN textphone device in use is the
   responsibility of the RTT gateway.  The ToIP User Agent need not
   concern itself with what kind of PSTN textphone device is connected
   to the RTT gateway.

   When the ToIP User Agent first has characters to send to the RTT
   gateway the ToIP device shall open the text channel if it was not
   opened before.  The ToIP device shall then transmit the characters to
   the RTT gateway.  The RTT gateway shall perform any required
   signaling on the PSTN termination if the type of PSTN textphone s
   needed to be known by the RTT gateway.  While connecting, the RTT
   gateway shall buffer any characters received from the ToIP User Agent
   and transmit them when the RTT gateway has connected to the PSTN
   textphone.  The size of the character buffer should be sufficient to
   hold the characters that may come from one side in a connection
   before a response is expected.

   If the RTT gateway does not support the modulation used by the PSTN
   textphone device, the RTT gateway may:

   a) transmit the received textphone signals via Voiceband Data (VBD)
   or the audio stream, depending on the capabilities of the ToIP User
   Agent.

   b) discard characters received from the ToIP User Agent.

   c) transmit the received characters to the PSTN textphone based on a
   pre-provisioned indication of the modulation.

7.1.  Handling of alternating text and audio

   Many PSTN textphones support alternating voice and real-time text.
   Some PSTN textphones can only handle text transmission in one
   direction at a time.  Most ToIP User Agents can send text and voice
   simultaneously and in both directions at the same time.  When
   bridging between these two environments, turn-taking schemes must be
   introduced both by the human users and by the RTT gateways.  The



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   following procedures should be applied:

7.1.1.  Non-continuous carrier PSTN textphones

   For these PSTN textphones, audio is transmitted through the RTT
   gateway between the PSTN circuit and the audio stream of the ToIP
   User Agent.  This is done as long as there is no text to transmit or
   receive on the PSTN side.  Text transmission and reception has
   priority over audio transmission.

7.1.2.  Continuous-carrier PSTN textphones

   For these PSTN textphones, the recommended procedure is:

   As long as carrier is maintained from the PSTN textphone, it is
   maintained from the RTT gateway, and text transmission can occur.  If
   carrier is dropped from the PSTN textphone, the RTT gateway shall
   transmit any remaining characters to the PSTN textphone and then drop
   the carrier and transmit audio.

   When carrier is received again from the PSTN textphone and if text is
   to be transmitted to the PSTN textphone, carrier transmission will be
   resumed from the gateway and text may be transmitted.

   If, during a period of audio transmission, text is received from the
   ToIP device, then audio transmission will be interrupted, carrier re-
   established and the text transmitted.

   If, during a period of carrier or text transmission, the Interrupt
   command (INT) is received in the T.140 [T.140] stream from the ToIP
   User Agent, the carrier should be dropped and transmission of audio
   through the gateway established.  In most cases the control of the
   carrier can be left to the PSTN textphone user.


8.  IANA Considerations

   TBD.


9.  Security Considerations

   TBD


10.  Normative References

   [T.140]    ITU-T, "Recommendation T.140, Protocol for Multimedia



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              Application Text Conversation and Addendum 1",
              February 2000.

   [V.18]     ITU-T, "Operational and interworking requirements for DCEs
              operating in the text telephone mode", November 2000.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC3761]  Faltstrom, P. and M. Mealling, "The E.164 to Uniform
              Resource Identifiers (URI) Dynamic Delegation Discovery
              System (DDDS) Application (ENUM)", RFC 3761, April 2004.

   [RFC3840]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
              "Indicating User Agent Capabilities in the Session
              Initiation Protocol (SIP)", RFC 3840, August 2004.

   [RFC3841]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller
              Preferences for the Session Initiation Protocol (SIP)",
              RFC 3841, August 2004.

   [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
              Conversation", RFC 4103, June 2005.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4975]  Campbell, B., Mahy, R., and C. Jennings, "The Message
              Session Relay Protocol (MSRP)", RFC 4975, September 2007.

   [RFC5194]  van Wijk, A. and G. Gybels, "Framework for Real-Time Text
              over IP Using the Session Initiation Protocol (SIP)",
              RFC 5194, June 2008.

   [I-D.hellstrom-text-turntaking]
              Hellstrom, G. and A. Wijk, "Registration of the Real-time-
              text Media Feature Tag",
              draft-hellstrom-text-turntaking-02 (work in progress),
              July 2009.



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Internet-Draft     Real-time text in SIP gateway calls        March 2010


Authors' Addresses

   Gunnar Hellstrom
   Omnitor
   Box 92054
   Stockholm,   120 06
   SE

   Phone: +46-8-58900056
   Email: gunnar.hellstrom@omnitor.se


   Barry Dingle
   3 Cosmo Court
   Kilsyth,   3137
   AU

   Phone: +61-3-9725-3937
   Email: btdingle@gmail.com


   Arnoud van Wijk
   Real-Time Text Taskforce (R3TF)
   NL

   Phone:
   Email: arnoud@realtimetext.org


   Guido Gybels
   RNID, Department of New Technologies
   19-23 Featherstone Street
   London,   EC1Y 8SL
   UK

   Phone: +44-20-7294 3713
   Email: guido.gybels@rnid.org














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