Network Working Group G. Camarillo
Request for Comments: 4117 Ericsson
Category: Informational E. Burger
Brooktrout
H. Schulzrinne
Columbia University
A. van Wijk
Viataal
June 2005
Transcoding Services Invocation in
the Session Initiation Protocol (SIP)
Using Third Party Call Control (3pcc)
Status of This Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2005).
Abstract
This document describes how to invoke transcoding services using
Session Initiation Protocol (SIP) and third party call control. This
way of invocation meets the requirements for SIP regarding
transcoding services invocation to support deaf, hard of hearing and
speech-impaired individuals.
Table of Contents
1. Introduction ....................................................2
2. General Overview ................................................2
3. Third Party Call Control Flows ..................................2
3.1. Terminology ................................................3
3.2. Callee's Invocation ........................................3
3.3. Caller's Invocation ........................................8
3.4. Receiving the Original Stream ..............................8
3.5. Transcoding Services in Parallel ..........................10
3.6. Multiple Transcoding Services in Series ...................14
4. Security Considerations ........................................16
5. Normative References ...........................................17
6. Informative References .........................................17
Camarillo, et al. Informational [Page 1]
RFC 4117 3pcc Transcoding in SIP June 2005 1. Introduction The framework for transcoding with SIP [4] describes how two SIP [1] UAs (User Agents) can discover incompatibilities that prevent them from establishing a session (e.g., lack of support for a common codec or common media type). When such incompatibilities are found, the UAs need to invoke transcoding services to successfully establish the session. 3pcc (third party call control) [2] is one way to perform such invocation. 2. General Overview In the 3pcc model for transcoding invocation, a transcoding server that provides a particular transcoding service (e.g., speech-to-text) is identified by a URI. A UA that wishes to invoke that service sends an INVITE request to that URI establishing a number of media streams. The way the transcoder manipulates and manages the contents of those media streams (e.g., the text received over the text stream is transformed into speech and sent over the audio stream) is service specific. All the call flows in this document use SDP. The same call flows could be used with another session description protocol that provides similar session description capabilities. 3. Third Party Call Control Flows Given two UAs (A and B) and a transcoding server (T), the invocation of a transcoding service consists of establishing two sessions; A-T and T-B. How these sessions are established depends on which party, the caller (A) or the callee (B), invokes the transcoding services. Section 3.2 deals with callee invocation and Section 3.3 deals with caller invocation. In all our 3pcc flows we have followed the general principle that a 200 (OK) response from the transcoding service has to be received before contacting the callee. This tries to ensure that the transcoding service will be available when the callee accepts the session. Still, the transcoding service does not know the exact type of transcoding it will be performing until the callee accepts the session. So, there is always the chance of failing to provide transcoding services after the callee has accepted the session. A system with more stringent requirements could use preconditions to avoid this situation. When preconditions are used, the callee is not alerted until everything is ready for the session. Camarillo, et al. Informational [Page 2]
RFC 4117 3pcc Transcoding in SIP June 2005 3.1. Terminology All the flows in this document follow the naming convention below: SDP A: A session description generated by A. It contains, among other things, the transport address/es (IP address and port number) where A wants to receive media for each particular stream. SDP B: A session description generated by B. It contains, among other things, the transport address/es where B wants to receive media for each particular stream. SDP A+B: A session description that contains, among other things, the transport address/es where A wants to receive media and the transport address/es where B wants to receive media. SDP TA: A session description generated by T and intended for A. It contains, among other things, the transport address/es where T wants to receive media from A. SDP TB: A session description generated by T and intended for B. It contains, among other things, the transport address/es where T wants to receive media from B. SDP TA+TB: A session description generated by T that contains, among other things, the transport address/es where T wants to receive media from A and the transport address/es where T wants to receive media from B. 3.2. Callee's Invocation In this scenario, B receives an INVITE from A, and B decides to introduce T in the session. Figure 1 shows the call flow for this scenario. In Figure 1, A can both hear and speak, and B is a deaf user with a speech impairment. A proposes to establish a session that consists of an audio stream (1). B wants to send and receive only text, so it invokes a transcoding service T that will perform both speech-to-text and text-to-speech conversions (2). The session descriptions of Figure 1 are partially shown below. Camarillo, et al. Informational [Page 3]
RFC 4117 3pcc Transcoding in SIP June 2005 A T B | | | |--------------------(1) INVITE SDP A-------------------->| | | | | |<---(2) INVITE SDP A+B------| | | | | |---(3) 200 OK SDP TA+TB---->| | | | | |<---------(4) ACK-----------| | | | |<-------------------(5) 200 OK SDP TA--------------------| | | | |------------------------(6) ACK------------------------->| | | | | ************************** | ************************** | |* MEDIA *|* MEDIA *| | ************************** | ************************** | | | | Figure 1: Callee's Invocation of a Transcoding Service (1) INVITE SDP A m=audio 20000 RTP/AVP 0 c=IN IP4 A.example.com (2) INVITE SDP A+B m=audio 20000 RTP/AVP 0 c=IN IP4 A.example.com m=text 40000 RTP/AVP 96 c=IN IP4 B.example.com a=rtpmap:96 t140/1000 (3) 200 OK SDP TA+TB m=audio 30000 RTP/AVP 0 c=IN IP4 T.example.com m=text 30002 RTP/AVP 96 c=IN IP4 T.example.com a=rtpmap:96 t140/1000 (5) 200 OK SDP TA m=audio 30000 RTP/AVP 0 c=IN IP4 T.example.com Camarillo, et al. Informational [Page 4]
RFC 4117 3pcc Transcoding in SIP June 2005 Four media streams (i.e., two bi-directional streams) have been established at this point: 1. Audio from A to T.example.com:30000 2. Text from T to B.example.com:40000 3. Text from B to T.example.com:30002 4. Audio from T to A.example.com:20000 When either A or B decides to terminate the session, it sends a BYE indicating that the session is over. If the first INVITE (1) received by B is empty (no session description), the call flow is slightly different. Figure 2 shows the messages involved. B may have different reasons for invoking T before knowing A's session description. B may want to hide its lack of native capabilities, and therefore wants to return a session description with all the codecs that B supports, plus all the codecs that T supports. Or T may provide recording services (besides transcoding), and B wants T to record the conversation, regardless of whether transcoding is needed. This scenario (Figure 2) is a bit more complex than the previous one. In INVITE (2), B still does not have SDP A, so it cannot provide T with that information. When B finally receives SDP A in (6), it has to send it to T. B sends an empty INVITE to T (7) and gets a 200 OK with SDP TA+TB (8). In general, this SDP TA+TB can be different than the one sent in (3). That is why B needs to send the updated SDP TA to A in (9). A then sends a possibly updated SDP A (10) and B sends it to T in (12). On the other hand, if T happens to return the same SDP TA+TB in (8) as in (3), B can skip messages (9), (10), and (11). So, implementors of transcoding services are encouraged to return the same session description in (8) as in (3) in this type of scenario. The session descriptions of this flow are shown below: Camarillo, et al. Informational [Page 5]
RFC 4117 3pcc Transcoding in SIP June 2005 A T B | | | |----------------------(1) INVITE------------------------>| | | | | |<-----(2) INVITE SDP B------| | | | | |---(3) 200 OK SDP TA+TB---->| | | | | |<---------(4) ACK-----------| | | | |<-------------------(5) 200 OK SDP TA--------------------| | | | |-----------------------(6) ACK SDP A-------------------->| | | | | |<-------(7) INVITE----------| | | | | |---(8) 200 OK SDP TA+TB---->| | | | |<-----------------(9) INVITE SDP TA----------------------| | | | |------------------(10) 200 OK SDP A--------------------->| | | | |<-----------------------(11) ACK-------------------------| | | | | |<-----(12) ACK SDP A+B------| | | | | ************************** | ************************** | |* MEDIA *|* MEDIA *| | ************************** | ************************** | Figure 2: Callee's invocation after initial INVITE without SDP (2) INVITE SDP A+B m=audio 20000 RTP/AVP 0 c=IN IP4 0.0.0.0 m=text 40000 RTP/AVP 96 c=IN IP4 B.example.com a=rtpmap:96 t140/1000 (3) 200 OK SDP TA+TB m=audio 30000 RTP/AVP 0 c=IN IP4 T.example.com m=text 30002 RTP/AVP 96 c=IN IP4 T.example.com a=rtpmap:96 t140/1000 Camarillo, et al. Informational [Page 6]
RFC 4117 3pcc Transcoding in SIP June 2005 (5) 200 OK SDP TA m=audio 30000 RTP/AVP 0 c=IN IP4 T.example.com (6) ACK SDP A m=audio 20000 RTP/AVP 0 c=IN IP4 A.example.com (8) 200 OK SDP TA+TB m=audio 30004 RTP/AVP 0 c=IN IP4 T.example.com m=text 30006 RTP/AVP 96 c=IN IP4 T.example.com a=rtpmap:96 t140/1000 (9) INVITE SDP TA m=audio 30004 RTP/AVP 0 c=IN IP4 T.example.com (10) 200 OK SDP A m=audio 20002 RTP/AVP 0 c=IN IP4 A.example.com (12) ACK SDP A+B m=audio 20002 RTP/AVP 0 c=IN IP4 A.example.com m=text 40000 RTP/AVP 96 c=IN IP4 B.example.com a=rtpmap:96 t140/1000 Camarillo, et al. Informational [Page 7]
RFC 4117 3pcc Transcoding in SIP June 2005 Four media streams (i.e., two bi-directional streams) have been established at this point: 1. Audio from A to T.example.com:30004 2. Text from T to B.example.com:40000 3. Text from B to T.example.com:30006 4. Audio from T to A.example.com:20002 3.3. Caller's Invocation In this scenario, A wishes to establish a session with B using a transcoding service. A uses 3pcc to set up the session between T and B. The call flow we provide here is slightly different than the ones in [2]. In [2], the controller establishes a session between two user agents, which are the ones deciding the characteristics of the streams. Here, A wants to establish a session between T and B, but A wants to decide how many and which types of streams are established. That is why A sends its session description in the first INVITE (1) to T, as opposed to the media-less initial INVITE recommended by [2]. Figure 3 shows the call flow for this scenario. We do not include the session descriptions of this flow, since they are very similar to those in Figure 2. In this flow, if T returns the same SDP TA+TB in (8) as in (2), messages (9), (10), and (11) can be skipped. 3.4. Receiving the Original Stream Sometimes, as pointed out in the requirements for SIP in support of deaf, hard of hearing, and speech-impaired individuals [5], a user wants to receive both the original stream (e.g., audio) and the transcoded stream (e.g., the output of the speech-to-text conversion). There are various possible solutions for this problem. One solution consists of using the SDP group attribute with Flow Identification (FID) semantics [3]. FID allows requesting that a stream is sent to two different transport addresses in parallel, as shown below: Camarillo, et al. Informational [Page 8]
RFC 4117 3pcc Transcoding in SIP June 2005 A T B | | | |-------(1) INVITE SDP A---->| | | | | |<----(2) 200 OK SDP TA+TB---| | | | | |----------(3) ACK---------->| | | | | |--------------------(4) INVITE SDP TA------------------->| | | | |<--------------------(5) 200 OK SDP B--------------------| | | | |-------------------------(6) ACK------------------------>| | | | |--------(7) INVITE--------->| | | | | |<---(8) 200 OK SDP TA+TB --| | | | | |--------------------(9) INVITE SDP TA------------------->| | | | |<-------------------(10) 200 OK SDP B--------------------| | | | |-------------------------(11) ACK----------------------->| | | | |------(12) ACK SDP A+B----->| | | | | | ************************** | ************************** | |* MEDIA *|* MEDIA *| | ************************** | ************************** | | | | Figure 3: Caller's invocation of a transcoding service a=group:FID 1 2 m=audio 20000 RTP/AVP 0 c=IN IP4 A.example.com a=mid:1 m=audio 30000 RTP/AVP 0 c=IN IP4 T.example.com a=mid:2 The problem with this solution is that the majority of the SIP user agents do not support FID. Moreover, only a small fraction of the few UAs that support FID, also support sending simultaneous copies of the same media stream at the same time. In addition, FID forces both copies of the stream to use the same codec. Camarillo, et al. Informational [Page 9]
RFC 4117 3pcc Transcoding in SIP June 2005 Therefore, we recommend that T (instead of a user agent) replicates the media stream. The transcoder T receiving the following session description performs speech-to-text and text-to-speech conversions between the first audio stream and the text stream. In addition, T copies the first audio stream to the second audio stream and sends it to A. m=audio 40000 RTP/AVP 0 c=IN IP4 B.example.com m=audio 20000 RTP/AVP 0 c=IN IP4 A.example.com a=recvonly m=text 20002 RTP/AVP 96 c=IN IP4 A.example.com a=rtpmap:96 t140/1000 3.5. Transcoding Services in Parallel Transcoding services sometimes consist of human relays (e.g., a person performing speech-to-text and text-to-speech conversions for a session). If the same person is involved in both conversions (i.e., from A to B and from B to A), he or she has access to all of the conversation. In order to provide some degree of privacy, sometimes two different persons are allocated to do the job (i.e., one person handles A->B and the other B->A). This type of disposition is also useful for automated transcoding services, where one machine converts text to synthetic speech (text-to-speech) and another performs voice recognition (speech-to-text). The scenario described above involves four different sessions: A-T1, T1-B, B-T2 and T2-A. Figure 4 shows the call flow where A invokes T1 and T2. Note this example uses unidirectional media streams (i.e., sendonly or recvonly) to clearly identify which transcoder handles media in which direction. Nevertheless, nothing precludes the use of bidirectional streams in this scenario. They could be used, for example, by a human relay to ask for clarifications (e.g., I did not get that, could you repeat, please?) to the party he or she is receiving media from. Camarillo, et al. Informational [Page 10]
RFC 4117 3pcc Transcoding in SIP June 2005 (1) INVITE SDP AT1 m=text 20000 RTP/AVP 96 c=IN IP4 A.example.com a=rtpmap:96 t140/1000 a=sendonly m=audio 20000 RTP/AVP 0 c=IN IP4 0.0.0.0 a=recvonly (2) INVITE SDP AT2 m=text 20002 RTP/AVP 96 c=IN IP4 A.example.com a=rtpmap:96 t140/1000 a=recvonly m=audio 20000 RTP/AVP 0 c=IN IP4 0.0.0.0 a=sendonly (3) 200 OK SDP T1A+T1B m=text 30000 RTP/AVP 96 c=IN IP4 T1.example.com a=rtpmap:96 t140/1000 a=recvonly m=audio 30002 RTP/AVP 0 c=IN IP4 T1.example.com a=sendonly (5) 200 OK SDP T2A+T2B m=text 40000 RTP/AVP 96 c=IN IP4 T2.example.com a=rtpmap:96 t140/1000 a=sendonly m=audio 40002 RTP/AVP 0 c=IN IP4 T2.example.com a=recvonly (7) INVITE SDP T1B+T2B m=audio 30002 RTP/AVP 0 c=IN IP4 T1.example.com a=sendonly m=audio 40002 RTP/AVP 0 c=IN IP4 T2.example.com a=recvonly Camarillo, et al. Informational [Page 11]
RFC 4117 3pcc Transcoding in SIP June 2005 A T1 T2 B | | | | |----(1) INVITE SDP AT1--->| | | | | | | |----------------(2) INVITE SDP AT2-------------->| | | | | | |<-(3) 200 OK SDP T1A+T1B--| | | | | | | |---------(4) ACK--------->| | | | | | | |<---------------(5) 200 OK SDP T2A+T2B-----------| | | | | | |----------------------(6) ACK------------------->| | | | | | |-----------------------(7) INVITE SDP T1B+T2B----------------->| | | | | |<----------------------(8) 200 OK SDP BT1+BT2------------------| | | | | |------(9) INVITE--------->| | | | | | | |-------------------(10) INVITE------------------>| | | | | | |<-(11) 200 OK SDP T1A+T1B-| | | | | | | |<------------(12) 200 OK SDP T2A+T2B-------------| | | | | | |------------------(13) INVITE SDP T1B+T2B--------------------->| | | | | |<-----------------(14) 200 OK SDP BT1+BT2----------------------| | | | | |--------------------------(15) ACK---------------------------->| | | | | |---(16) ACK SDP AT1+BT1-->| | | | | | | |------------(17) ACK SDP AT2+BT2---------------->| | | | | | | ************************ | ********************************** | |* MEDIA *|* MEDIA *| | ************************ | ********************************** | | | | | | *********************************************** *********** |* MEDIA *|* MEDIA *| | *********************************************** | *********** | | | | | Figure 4: Transcoding services in parallel Camarillo, et al. Informational [Page 12]
RFC 4117 3pcc Transcoding in SIP June 2005 (8) 200 OK SDP BT1+BT2 m=audio 50000 RTP/AVP 0 c=IN IP4 B.example.com a=recvonly m=audio 50002 RTP/AVP 0 c=IN IP4 B.example.com a=sendonly (11) 200 OK SDP T1A+T1B m=text 30000 RTP/AVP 96 c=IN IP4 T1.example.com a=rtpmap:96 t140/1000 a=recvonly m=audio 30002 RTP/AVP 0 c=IN IP4 T1.example.com a=sendonly (12) 200 OK SDP T2A+T2B m=text 40000 RTP/AVP 96 c=IN IP4 T2.example.com a=rtpmap:96 t140/1000 a=sendonly m=audio 40002 RTP/AVP 0 c=IN IP4 T2.example.com a=recvonly Since T1 have returned the same SDP in (11) as in (3), and T2 has returned the same SDP in (12) as in (5), messages (13), (14) and (15) can be skipped. (16) ACK SDP AT1+BT1 m=text 20000 RTP/AVP 96 c=IN IP4 A.example.com a=rtpmap:96 t140/1000 a=sendonly m=audio 50000 RTP/AVP 0 c=IN IP4 B.example.com a=recvonly Camarillo, et al. Informational [Page 13]
RFC 4117 3pcc Transcoding in SIP June 2005 (17) ACK SDP AT2+BT2 m=text 20002 RTP/AVP 96 c=IN IP4 A.example.com a=rtpmap:96 t140/1000 a=recvonly m=audio 50002 RTP/AVP 0 c=IN IP4 B.example.com a=sendonly Four media streams have been established at this point: 1. Text from A to T1.example.com:30000 2. Audio from T1 to B.example.com:50000 3. Audio from B to T2.example.com:40002 4. Text from T2 to A.example.com:20002 Note that B, the user agent server, needs to support two media streams: sendonly and recvonly. At present, some user agents, although they support a single sendrecv media stream, do not support a different media line per direction. Implementers are encouraged to build support for this feature. 3.6. Multiple Transcoding Services in Series In a distributed environment, a complex transcoding service (e.g., English text to Spanish speech) is often provided by several servers. For example, one server performs English text to Spanish text translation, and its output is fed into a server that performs text- to-speech conversion. The flow in Figure 5 shows how A invokes T1 and T2. Camarillo, et al. Informational [Page 14]
RFC 4117 3pcc Transcoding in SIP June 2005 A T1 T2 B | | | | |----(1) INVITE SDP A-----> | | | | | | | |<-(2) 200 OK SDP T1A+T1T2- | | | | | | | |----------(3) ACK--------> | | | | | | | |-----------(4) INVITE SDP T1T2------------------>| | | | | | |<-----------(5) 200 OK SDP T2T1+T2B--------------| | | | | | |---------------------(6) ACK-------------------->| | | | | | |---------------------------(7) INVITE SDP T2B----------------->| | | | | |<--------------------------(8) 200 OK SDP B--------------------| | | | | |--------------------------------(9) ACK----------------------->| | | | | |---(10) INVITE-----------> | | | | | | | |------------------(11) INVITE------------------->| | | | | | |<-(12) 200 OK SDP T1A+T1T2-| | | | | | | |<-------------(13) 200 OK SDP T2T1+T2B-----------| | | | | | |---(14) ACK SDP T1T2+B---> | | | | | | | |-----------------------(15) INVITE SDP T2B-------------------->| | | | | |<----------------------(16) 200 OK SDP B-----------------------| | | | | |----------------(17) ACK SDP T1T2+B------------->| | | | | | |----------------------------(18) ACK-------------------------->| | | | | | ************************* | ******************* *********** | |* MEDIA *|* MEDIA *|* MEDIA *| | ************************* | ******************* | *********** | | | | | Figure 5: Transcoding services in serial Camarillo, et al. Informational [Page 15]
RFC 4117 3pcc Transcoding in SIP June 2005 4. Security Considerations RFC 3725 [2] discusses security considerations which relate to the use of third party call control in SIP. These considerations apply to this document, since it describes how to use third party call control to invoke transcoding service. In particular, RFC 3725 states that end-to-end media security is based on the exchange of keying material within SDP and depends on the controller behaving properly. That is, the controller should not try to disable the security mechanisms offered by the other parties. As a result, it is trivially possible for the controller to insert itself as an intermediary on the media exchange, if it should so desire. In this document, the controller is the UA invoking the transcoder, and there is a media session established using third party call control between the remote UA and the transcoder. Consequently, the attack described in RFC 3725 does not constitute a threat because the controller is the UA invoking the transcoding service and it has access to the media anyway by definition. So, it seems unlikely that a UA would attempt to launch an attack against its own session by disabling security between the transcoder and the remote UA. Regarding end-to-end media security from the UAs' point of view, the transcoder needs access to the media in order to perform its function. So, by definition, the transcoder behaves as a man in the middle. UAs that do not want a particular transcoder to have access to all the media exchanged between them can use a different transcoder for each direction. In addition, UAs can use different transcoders for different media types. Camarillo, et al. Informational [Page 16]
RFC 4117 3pcc Transcoding in SIP June 2005 5. Normative References [1] 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. [2] Rosenberg, J., Peterson, J., Schulzrinne, H., and G. Camarillo, "Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)", BCP 85, RFC 3725, April 2004. [3] Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne, "Grouping of Media Lines in the Session Description Protocol (SDP)", RFC 3388, December 2002. 6. Informative References [4] Camarillo, G., "Framework for transcoding with the session initiation protocol", August 2003, Work in Progress. [5] 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. Camarillo, et al. Informational [Page 17]
RFC 4117 3pcc Transcoding in SIP June 2005 Authors' Addresses Gonzalo Camarillo Ericsson Advanced Signalling Research Lab. FIN-02420 Jorvas Finland EMail: Gonzalo.Camarillo@ericsson.com Eric Burger Brooktrout Technology, Inc. 18 Keewaydin Way Salem, NH 03079 USA EMail: eburger@brooktrout.com Henning Schulzrinne Dept. of Computer Science Columbia University 1214 Amsterdam Avenue, MC 0401 New York, NY 10027 USA EMail: schulzrinne@cs.columbia.edu Arnoud van Wijk Viataal Research & Development Afdeling RDS Theerestraat 42 5271 GD Sint-Michielsgestel The Netherlands EMail: a.vwijk@viataal.nl Camarillo, et al. Informational [Page 18]
RFC 4117 3pcc Transcoding in SIP June 2005 Full Copyright Statement Copyright (C) The Internet Society (2005). This document is subject to the rights, licenses and restrictions contained in BCP 78, and except as set forth therein, the authors retain all their rights. This document and the information contained herein are provided on an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. Intellectual Property The IETF takes no position regarding the validity or scope of any Intellectual Property Rights 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; nor does it represent that it has made any independent effort to identify any such rights. Information on the procedures with respect to rights in RFC documents can be found in BCP 78 and BCP 79. Copies of IPR disclosures made to the IETF Secretariat and any assurances of licenses to be made available, or the result of an attempt made to obtain a general license or permission for the use of such proprietary rights by implementers or users of this specification can be obtained from the IETF on-line IPR repository at http://www.ietf.org/ipr. The IETF invites any interested party to bring to its attention any copyrights, patents or patent applications, or other proprietary rights that may cover technology that may be required to implement this standard. Please address the information to the IETF at ietf- ipr@ietf.org. Acknowledgement Funding for the RFC Editor function is currently provided by the Internet Society. Camarillo, et al. Informational [Page 19]
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