Speechsc D. Oran Internet-Draft Cisco Systems, Inc. Expires: December 5, 2003 June 6, 2003 Requirements for Distributed Control of ASR, SI/SV and TTS Resources draft-ietf-speechsc-reqts-04 Status of this Memo This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet-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 material or to cite them other than as "work in progress." The list of current Internet-Drafts can be accessed at http:// www.ietf.org/ietf/1id-abstracts.txt. The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on December 5, 2003. Copyright Notice Copyright (C) The Internet Society (2003). All Rights Reserved. Abstract This document outlines the needs and requirements for a protocol to control distributed speech processing of audio streams. By speech processing, this document specifically means automatic speech recognition (ASR), speaker recognition - which includes both speaker identification (SI) and speaker verification (SV) - and text-to-speech (TTS). Other IETF protocols, such as SIP and RTSP, address rendezvous and control for generalized media streams. However, speech processing presents additional requirements that none of the extant IETF protocols address. Discussion of this and related documents is on the speechsc mailing list. To subscribe, send the message "subscribe speechsc" to speechsc-request@ietf.org. The public archive is at http:// Oran Expires December 5, 2003 [Page 1] Internet-Draft Speech Services Control Requirements June 2003 www.ietf.org/mail-archive/workinggroups/speechsc/current/ maillist.html 1. Introduction There are multiple IETF protocols for establishment and termination of media sessions (SIP [5]), low-level media control (MGCP [6] and MEGACO [7]), and media record and playback (RTSP [8]). This document focuses on requirements for one or more protocols to support the control of network elements that perform Automated Speech Recognition (ASR), speaker identification or verification (SI/SV), and rendering text into audio, also known as Text-to-Speech (TTS). Many multimedia applications can benefit from having automatic speech recognition (ASR) and text-to-speech (TTS) processing available as a distributed, network resource. This requirements document limits its focus to the distributed control of ASR, SI/SV and TTS servers. There are a broad range of systems which can benefit from a unified approach to control of TTS, ASR, and SI/SV. These include environments such as VoIP gateways to the PSTN, IP Telephones, media servers, and wireless mobile devices who obtain speech services via servers on the network. To date, there are a number of proprietary ASR and TTS API's, as well as two IETF drafts that address this problem [12], [13]. However, there are serious deficiencies to the existing drafts. In particular, they mix the semantics of existing protocols yet are close enough to other protocols as to be confusing to the implementer. This document sets forth requirements for protocols to support distributed speech processing of audio streams. For simplicity, and to remove confusion with existing protocol proposals, this document presents the requirements as being for a "framework" that addresses the distributed control of speech resources It refers to such a framework as "SPEECHSC", for Speech Services Control. 2. SPEECHSC Framework Figure 1 below shows the SPEECHSC framework for speech processing. +-------------+ | Application | | Server |\ +-------------+ \ SPEECHSC SIP, VoiceXML, / \ etc. / \ +------------+ / \ +-------------+ Oran Expires December 5, 2003 [Page 2] Internet-Draft Speech Services Control Requirements June 2003 | Media |/ SPEECHSC \---| ASR, SI/SV | | Processing |-------------------------| and/or TTS | RTP | Entity | RTP | Server | =====| |=========================| | +------------+ +-------------+ Figure 1: Figure 1: SPEECHSC Framework The "Media Processing Entity" is a network element that processes media. It may be either a pure media handler, or also have an associated SIP user agent, VoiceXML browser or other control entity. The "ASR, SI/SV and/or TTS Server" is a network element which performs the back-end speech processing. It may generate an RTP stream as output based on text input (TTS) or return recognition results in response to an RTP stream as input (ASR, SI/SV). The "Application Server" is a network element that instructs the Media Processing Entity on what transformations to make to the media stream. Those instructions may be established via a session protocol such as SIP, or provided via a client/server exchange such as VoiceXML. The framework allows either the Media Processing Entity or the Application Server to control the ASR or TTS Server using SPEECHSC as a control protocol, which accounts for the speechsc protocol appearing twice in the diagram. Physical embodiments of the entities can reside in one physical instance per entity, or some combination of entities. For example, a VoiceXML [10] Gateway may combine the ASR and TTS functions on the same platform as the Media Processing Entity. Note that VoiceXML Gateways themselves are outside the scope of this protocol. Likewise, one can combine the Application Server and Media Processing Entity, as would be the case in an interactive voice response (IVR) platform. One can also decompose the Media Processing Entity into an entity that controls media endpoints and entities that process media directly. Such would be the case with a decomposed gateway using MGCP or megaco. However, this decomposition is again orthogonal to the scope of SPEECHSC. The following subsections provide a number of example use cases the SPEECHSC, one each for TTS, ASR and SI/SV. They are intended to be illustrative only, and not to imply any restriction on the scope of the framework or to limit the decompostion or configuration to that shown in the example. 2.1 TTS Example This example illustrates a simple usage of SPEECHSC to provide a Text to Speech service for playing announcements to a user on a phone with no display for textual error messages. The example scenario is shown below in figure 2. In the figure, the VoIP gateway acts as both the Oran Expires December 5, 2003 [Page 3] Internet-Draft Speech Services Control Requirements June 2003 Media Processing Entity and the Application Server of the SPEECHSC framework in figure 1. +---------+ _| SIP | _/ | Server | +-----------+ SIP/ +---------+ | | _/ +-------+ | VoIP |_/ | POTS |___| Gateway | RTP +---------+ | Phone | | (SIP UA) |=========| | +-------+ | |\_ | SPEECHSC| +-----------+ \ | TTS | \__ | Server | SPEECHSC | | \_| | +---------+ Figure 2: Figure 2: Text to speech example of SPEECHSC The POTS phone on the left attempts to make a phone call. The VoIP gateway, acting as a SIP UA, tries to establish a SIP session to complete the call, but gets an error, such as a SIP "486 Busy Here" response. Without SPEECHSC the gateway would most likely just output a busy signal to the POTS phone, However, with SPEECHSC access to a TTS server it can provide a spoken error message. The VoIP Gateway therefore constructs a text error string using information from the SIP messages, such as "Your call to 978-555-1212 did not go through because the called party was busy". It then can use SPEECHSC to establish an association with a SPEECHSC server, open an RTP stream between itself and the server, and issue a TTS request for the error message, which will be played to the user on the POTS phone. 2.2 Automatic speech recognition example This example illustrates a VXML-enabled media processing entity and associated application server using the SPEECHSC framework to supply an ASR-based user interface through an Interactive Voice Response (IVR) system. The example scenario is shown below in figure 3. The VXML-client corresponds to the "media processing entity", while the IVR application server corresponds to the "application server" of the SPEECHSC framework of figure 1. +------------+ | IVR | _|Application | VXML_/ +------------+ +-----------+ __/ Oran Expires December 5, 2003 [Page 4] Internet-Draft Speech Services Control Requirements June 2003 | |_/ +------------+ PSTN Trunk | VoIP | SPEECHSC| | =============| Gateway |---------| SPEECHSC | |(VXML voice| | ASR | | browser) |=========| Server | +-----------+ RTP +------------+ Figure 3: Figure 3: Automatic speech recognition example In this example, users call into the service in order to obtain stock quotes. The VoIP gateway answers their PSTN call. An IVR application feeds VXML scripts to the gateway to drive the user interaction. The VXML interpreter on the gateway directs the user's media stream to the SPEECHSC ASR server and uses SPEECHSC to control the ASR server. When, for example, the user speaks the name of a stock in response to an IVR prompt, the SPEECHSC ASR server attempts recognition of the name, and returns the results to the VXML gateway. The VXML gateway, following standard VXML mechanisms, informs the IVR Application of the recognized result. The IVR Application can then do the appropriate information lookup. The answer, of course, can be sent back to the user using text-to-speech. This example does not show this scenario, but it would work analogously to the scenario shown in section Section 2.1. 2.3 Speaker Identification example This example illustrates using speaker identification to allow voice-actuated login to an IP phone. The example scenario is shown below in figure 4. In the figure, the IP Phone acts as both the "Media Processing Entity" and the Application Server" of the SPEECHSC framework in figure 1. +-----------+ +---------+ | | RTP | | | IP |=========| SPEECHSC| | Phone | | TTS | | |_________| Server | | | SPEECHSC| | +-----------+ +---------+ Figure 4: Figure 4: Speaker identification example In this example, a user speaks into a SIP phone in order to get "logged in" to that phone to make and receive phone calls using his identity and preferences. The IP phone uses the SPEECHSC framework to Oran Expires December 5, 2003 [Page 5] Internet-Draft Speech Services Control Requirements June 2003 set up an RTP stream between the phone and the SPEECHSC SI/SV server and to request verification. The SV server verifies the user's identity and returns the result, including the necessary login credentials, to the phone via SPEECHSC. The IP Phone may either use the identity directly to identify the user in outgoing calls, to fetch the user's preferences from a configuration server, request authorization from a AAA server, in any combination. Since this example uses SPEECHSC to perform a security-related function, be sure to note the associated material in Section 9 3. General Requirements 3.1 Reuse Existing Protocols To the extent feasible, the SPEECHSC framework SHOULD use existing protocols. 3.2 Maintain Existing Protocol Integrity In meeting the requirement of Section 3.1, the SPEECHSC framework MUST NOT redefine the semantics of an existing protocol. Said differently, we will not break existing protocols or cause backward compatibility problems. 3.3 Avoid Duplicating Existing Protocols To the extent feasible, SPEECHSC SHOULD NOT duplicate the functionality of existing protocols. For example, network announcements using SIP [11] and RTSP [8] already define how to request playback of audio. The focus of SPEECHSC is new functionality not addressed by existing protocols or extending existing protocols within the strictures of the requirement in Section 3.2. Where an existing protocol can be gracefully extended to support SPEECHSC requirements, such extensions are acceptable alternatives for meeting the requirements. As a corollary to this, the SPEECHSC should not require a separate protocol to perform functions that could be easily added into the SPEECHSC protocol (like redirecting media streams, or discovering capabilities), unless it is similarly easy to embed that protocol directly into the SPEECHSC framework. 3.4 Efficiency The SPEECHSC framework SHOULD employ protocol elements known to result in efficient operation. Techniques to be considered include: o Re-use of transport connections across sessions Oran Expires December 5, 2003 [Page 6] Internet-Draft Speech Services Control Requirements June 2003 o Piggybacking of responses on requests in the reverse direction o Caching of state across requests 3.5 Invocation of services The SPEECHSC framework MUST be compliant with the IAB OPES [3] framework. The applicability of the SPEECHSC protocol will therefore be specified as occurring between clients and servers at least one of which is operating directly on behalf of the user requesting the service. 3.6 Location and Load Balancing To the extent feasible, the SPEECHSC framework SHOULD exploit existing schemes for supporting service location and load balancing, such as the Service Location Protocol [12] or DNS SRV records [13]. Where such facilities are not deemed adequate, the SPEECHSC framework MAY define additional load balancing techniques. 3.7 Multiple services The SPEECHSC framework MUST permit multiple services to operate on a single media stream so that either the same or different servers may be performing speech recognition, speaker identification or verification, etc. in parallel. 3.8 Multiple media sessions The SPEECHSC framework MUST allow a 1:N mapping between session and RTP channels. For example, a single session may include an outbound RTP channel for TTS, an inbound for ASR and a different inbound for SI/SV (e.g. if processed by different elements on the Media Resource Element). Note: All of these can be described via SDP, so if SDP is utilized for media channel description, this requirement is met "for free". 3.9 Handicapped Users The SPEECHSC framework must have sufficient capabilities to address the critical needs of handicapped users. In particular, the set of requirements set forth in RFC3351 [4] MUST be taken into account by the framework. It is also important that implementers of SPEECHSC clients and servers be cognizant that some interaction modalities of SPEECHSC may be inconvenient, or simply inappropriate for handicapped users. Hearing-impaired users may find TTS of limited utility. Spech-impaired users may be unable to make use of ASR or SI/SV Oran Expires December 5, 2003 [Page 7] Internet-Draft Speech Services Control Requirements June 2003 capabilities. Therefore, systems employing SPEECHSC need to, where feasible, provide alternative interaction modes or avoid the use of speech processing entirely. 3.10 Identification of process which produced media or control output The client of a SPEECHSC operation SHOULD be able to ascertain via the SPEECHSC framework what speech process produced the output. For example, an RTP stream containing the spoken output of TTS should be identifiable as TTS output, and the recognized utterance of ASR be identifiable as having been produced by ASR processing. 4. TTS Requirements 4.1 Requesting Text Playback The SPEECHSC framework MUST allow a Media Processing Entity or Application Server, using a control protocol, to request the TTS Server to playback text as voice in an RTP stream. 4.2 Text Formats 4.2.1 Plain Text The SPEECHSC framework MAY assume that all TTS servers are capable of reading plain text. For reading plain text, framework MUST allow the language and voicing to be indicated via session parameters. For finer control over such properties, see Section 4.2.2. 4.2.2 SSML The SPEECHSC framework MUST support SSML[3] basics, and SHOULD support other SSML tags. The framework assumes all TTS servers are capable of reading SSML formatted text. Internationalization of TTS in the SPEECHSC framework, including multi-lingual output within a single utterance, is accomplished via SSML xml:lang tags. 4.2.3 Text in Control Channel The Speechsc framework assumes all TTS servers accept text over the SPEECHSC connection for reading over the RTP connection. The framework assumes the server can accept text either "by value" (embedded in the protocol), or "by reference" (e.g. by de-referencing a URI embedded in the protocol). 4.2.4 Document Type Indication A document type specifies to the syntax in which the text to be read Oran Expires December 5, 2003 [Page 8] Internet-Draft Speech Services Control Requirements June 2003 is encoded. The SPEECHSC framework MUST be capable of explicitly indicating the document type of the text to be processed, as opposed to forcing the server to infer the content by other means. 4.3 Control Channel The SPEECHSC framework MUST be capable of establishing the control channel between the client and server on a per-session basis, where a session is loosely defined to be associated with a single "call" or "dialog". The protocol SHOULD be capable of maintaining a long-lived control channel for multiple sessions serially, and MAY be capable of shorter time horizons as well, including as short as for the processing of a single utterance. 4.4 Media origination/termination by control elements The SPEECHSC framework MUST NOT require the controlling element (application server, media processing entity) to accept or originate media streams. Media streams MAY source & sink from the controlled element (ASR, TTS, etc.). 4.5 Playback Controls The Speechsc framework MUST support "VCR controls" for controlling the playout of streaming media output from SPEECHSC processing, and MUST allow for servers with varying capabilities to accommodate such controls. The protocol SHOULD allow clients to state what controls they wish to use, and for servers to report which ones they honor. These capabilities include: o The ability to jump in time to the location of a specific marker. o The ability to jump in time, forwards or backwards, by a specified amount of time. Valid time units MUST include seconds, words, paragraphs, sentences, and markers. o The ability to increase and decrease playout speed. o The ability to fast-forward and fast-rewind the audio, where snippets of audio are played as the server moves forwards or backwards in time. o The ability to pause and resume playout. o The ability to increase and decrease playout volume. These controls SHOULD be made easily available to users through the client user interface and through per-user customization capabilities Oran Expires December 5, 2003 [Page 9] Internet-Draft Speech Services Control Requirements June 2003 of the client. This is particularly important for hearing-impaired users, who will likely desire settings and control regimes different from those that would be acceptable for non-impaired users. 4.6 Session Parameters The SPEECHSC framework MUST support the specification of session parameters, such as language, prosody and voicing. 4.7 Speech Markers The SPEECHSC framework MUST accommodate speech markers, with capability at least as flexible as that provided in SSML [1]. The framework MUST further provide an efficient mechanism for reporting that a marker has been reached during playout. 5. ASR Requirements 5.1 Requesting Automatic Speech Recognition The SPEECHSC framework MUST allow a Media Processing Entity or Application Server to request the ASR Server to perform automatic speech recognition on an RTP stream, returning the results over SPEECHSC. 5.2 XML The Speechsc framework assumes that all ASR servers support the VoiceXML speech recognition grammar specification (SRGS) for speech recognition [2]. 5.3 Grammar Requirements 5.3.1 Grammar Specification The SPEECHSC framework assumes all ASR servers are capable of accepting grammar specifications either "by value" (embedded in the protocol), or "by reference" (e.g. by de-referencing a URI embedded in the protocol). The latter MUST allow the indication of a grammar already known to, or otherwise "built in" to the server. The framework and protocol further SHOULD exploit the ability to store and later retrieve by reference large grammars which were originally supplied by the client. 5.3.2 Explicit Indication of Grammar Format The SPEECHSC framework protocol MUST be able to explicitly convey the grammar format in which the grammar is encoded and MUST be extensible Oran Expires December 5, 2003 [Page 10] Internet-Draft Speech Services Control Requirements June 2003 to allow for conveying new grammar formats as they are defined. 5.3.3 Grammar Sharing The SPEECHSC framework SHOULD exploit sharing grammars across sessions for servers which are capable of doing so. This supports applications with large grammars for which it is unrealistic to dynamically load. An example is a city-country grammar for a weather service. 5.4 Session Parameters The SPEECHSC framework MUST accommodate at a minimum all of the protocol parameters currently defined in MRCP [9] In addition there SHOULD be a capability to reset parameters within a session. 5.5 Input Capture The SPEECHSC framework MUST support a method directing the ASR Server to capture the input media stream for later analysis and tuning of the ASR engine. 6. Speaker Identification and Verification Requirements 6.1 Requesting SI/SV The SPEECHSC framework MUST allow a Media Processing Entity to request the SI/SV Server to perform speaker identification or verification on an RTP stream, returning the results over SPEECHSC. 6.2 Identifiers for SI/SV The SPEECHSC framework MUST accommodate an identifier for each verification resource and permit control of that resource by ID, because voiceprint format and contents are vendor specific. 6.3 State for multiple utterances The SPEECHSC framework MUST work with SI/SV servers which maintain state to handle multi-utterance verification. 6.4 Input Capture The SPEECHSC framework MUST support a method for capturing the input media stream for later analysis and tuning of the SI/SV engine. The framework may assume all servers are capable of doing so. In addition the framework assumes that the captured stream contains enough timestamp context (e.g. the NTP time range from the RTCP packets Oran Expires December 5, 2003 [Page 11] Internet-Draft Speech Services Control Requirements June 2003 which corresponds to the RTP timestamps of the captured input) to ascertain after the fact exactly when the verification was requested. 6.5 SI/SV functional extensibility The SPEECHSC framework SHOULD be extensible to additional functions associated with SI/SV, such as prompting, utterance verification, and retraining. 7. Duplexing and Parallel Operation Requirements One very important requirement for an interactive speech-driven system is that user perception of the quality of the interaction depends strongly on the ability of the user to interrupt a prompt or rendered TTS with speech. Interrupting, or barging, the speech output requires more than energy detection from the user's direction. Many advanced systems halt the media towards the user by employing the ASR engine to decide if an utterance is likely to be real speech, as opposed to a cough, for example. 7.1 Full Duplex operation To achieve low latency between utterance detection and halting of playback, many implementations combine the speaking and ASR functions. The SPEECHSC framework MUST support such full-duplex implementations. 7.2 Multiple services in parallel Good spoken user interfaces typically depend upon the ease with which the user can accomplish his or her task. When making use of Speaker Identification or Verification technologies, user interface improvements often come from the combination of the different technologies: simultaneous identity claim and verification (on the same utterance), simultaneous knowledge and voice verification (using ASR and verification simultaneously). Using ASR and verification on the same utterance is in fact the only way to support rolling or dynamically-generated challenge phrases (e.g., "say 51723"). The SPEECHSC framework MUST support such parallel service implementations. 7.3 Combination of services It is optionally of interest that the SPEECHSC framework support more complex remote combination and controls of speech engines: o Combination in series of engines that may then act on the input or output of ASR, TTS or Speaker recognition engines. The control MAY Oran Expires December 5, 2003 [Page 12] Internet-Draft Speech Services Control Requirements June 2003 then extend beyond such engines to include other audio input and output processing and natural language processing. o Intermediate exchanges and coordination between engines o Remote specification of flows between engines. These capabilities MAY benefit from service discovery mechanisms (e.g. engines, properties and states discovery). 8. Additional Considerations (non-normative) The framework assumes that SDP will be used to describe media sessions and streams. The framework further assumes RTP carriage of media, however since SDP can be used to describe other media transport schemes (e.g. ATM) these could be used if they provide the necessary elements (e.g. explicit timestamps). The working group will not be defining distributed speech recognition methods (DSR), as exemplified by the ETSI Aurora project. The working group will not be recreating functionality available in other protocols, such as SIP or SDP. TTS looks very much like playing back a file. Extending RTSP looks promising for when one requires VCR controls or markers in the text to be spoken. When one does not require VCR controls, SIP in a framework such as Network Announcements [11] works directly without modification. ASR has an entirely different set of characteristics. For barge-in support, ASR requires real-time return of intermediate results. Barring the discovery of a good reuse model for an existing protocol, this will most likely become the focus of SPEECHSC. 9. Security Considerations Protocols relating to speech processing must take security into account. Many applications of speech technology deal with sensitive information, such as the use of TTS to read financial information. Likewise, popular uses for ASR include executing financial transactions and shopping. There are at least three aspects of speech processing security which intersect with the SPEECHSC requirements - securing the speechsc protocol itself, implementing and deploying the servers which run the protocol, and ensuring utilization of the technology for providing security functions is appropriate. Each of these aspects in discussed in the following sub-sections. While some of these considerations are Oran Expires December 5, 2003 [Page 13] Internet-Draft Speech Services Control Requirements June 2003 strictly speaking out of scope of the protocol itself, they will be carefully considered and accommodated during protocol design, and will be called out as part of the applicability statement accompanying the protocol specification(s). 9.1 SPEECHSC protocol security We envision that rather than providing application-specific security mechanisms in SPEECHSC itself, the resulting protocol will employ security machinery of either containing protocols or the transport on which it runs. For example, we will consider solutions such as using TLS for securing the control channel, and SRTP for securing the media channel. Third-party dependencies necessitating transitive trust will be minimized or explicitly dealt with through the authentication and authorization aspects of the protocol design. 9.2 Client and server implementation and deployment Given the possibly sensitive nature of the information carried, SPEECHSC clients and servers need to take steps to ensure confidentiality and integrity of the data and its transformations to and from spoken form. In addition to these general considerations, certain SPEECHSC functions, such as speaker verification and identification, employ voiceprints whose privacy and integrity must be maintained. Similarly, the requirement to support input capture for analysis and tuning can represent a privacy vulnerability because user utterances are recorded and could be either revealed or re-played inappropriately. 9.3 Use of SPEECHSC for security functions Speaker identification/verification can be used as an authentication and/or authorization technology. When so employed, there are additional vulnerabilties that may need to be addressed through the use of security measures for both the protocol and the clients and servers. For example, the ability of manipulate the media stream of a speaker verification request could inappropriately permit or deny access based on impersonation, or simple garbling via noise injection. 10. Acknowledgements Eric Burger wrote the original draft of these requirements and has continued to contribute actively throughout their development. He is a co-author in all but formal authorship, and is instead acknowledged here as it is preferable that working group co-chairs have non-conflicting roles with respect to the progression of documents. Oran Expires December 5, 2003 [Page 14] Internet-Draft Speech Services Control Requirements June 2003 Normative References [1] World Wide Web Consortium, "Speech Synthesis Markup Language Version 1.0", W3C Working Draft , December 2002, . [2] World Wide Web Consortium, "Speech Recognition Grammar Specification Version 1.0", W3C Candidate Recommendation , June 2002, . [3] Floyd, S. and L. Daigle, "IAB Architectural and Policy Considerations for Open Pluggable Edge Services", RFC 3238, January 2002. [4] Charlton, N., Gasson, M., Gybels, G., Spanner, M. and A. van Wijk, "User Requirements for the Session Initiation Protocol (SIP) in Support of Deaf, Hard of Hearing and Speech-impaired Individuals", RFC 3351, August 2002. Informative References [5] 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. [6] Arango, M., Dugan, A., Elliott, I., Huitema, C. and S. Pickett, "Media Gateway Control Protocol (MGCP) Version 1.0", RFC 2705, October 1999. [7] Cuervo, F., Greene, N., Rayhan, A., Huitema, C., Rosen, B. and J. Segers, "Megaco Protocol Version 1.0", RFC 3015, November 2000. [8] Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, April 1998. [9] Shanmugham, S., Monaco, P. and B. Eberman, "MRCP: Media Resource Control Protocol", Internet Draft draft-shanmugham-mrcp-04.txt, May 2003. [10] World Wide Web Consortium, "Voice Extensible Markup Language (VoiceXML) Version 2.0", W3C Working Draft , April 2002, . [11] Burger, E., Van Dyke, J., O'Connor, W. and A. Spitzer, "Basic Network Media Services with SIP", draft-burger-sipping-netann-05 (work in progress), March 2003. Oran Expires December 5, 2003 [Page 15] Internet-Draft Speech Services Control Requirements June 2003 [12] Guttman, E., Perkins, C., Veizades, J. and M. Day, "Service Location Protocol, Version 2", RFC 2608, June 1999. [13] Gulbrandsen, A., Vixie, P. and L. Esibov, "A DNS RR for specifying the location of services (DNS SRV)", RFC 2782, February 2000. Author's Address David R Oran Cisco Systems, Inc. 7 Ladyslipper Lane Acton, MA USA EMail: oran@cisco.com Oran Expires December 5, 2003 [Page 16] Internet-Draft Speech Services Control Requirements June 2003 Intellectual Property Statement The IETF takes no position regarding the validity or scope of any intellectual property or other rights that might be claimed to pertain to the implementation or use of the technology described in this document or the extent to which any license under such rights might or might not be available; neither does it represent that it has made any effort to identify any such rights. Information on the IETF's procedures with respect to rights in standards-track and standards-related documentation can be found in BCP-11. 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