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Copyright Notice

Copyright © The IETF Trust (2008).

Abstract

The "music on hold" feature is one of the most desired features of telephone systems in the business environment. "Music on hold" is when one party to a call has the call "on hold", the other party receives a media stream (often either music or advertising). Architectural features of SIP make it difficult to implement music-on-hold in a way that is fully compliant with the standards. The implementation of music-on-hold described in this document is fully effective and standards-compliant, but is simpler than the methods previously documented.

Table of Contents

1.  Introduction
2.  Technique
    2.1.  Placing a Call on Hold and Providing an External Media Stream
    2.2.  Taking a Call off Hold and Terminating the External Media Stream
    2.3.  Example Message Flow
    2.4.  Managing o= Lines
3.  Advantages
4.  Security Considerations
5.  Acknowledgments
6.  Revision History
    6.1.  Changes from draft-worley-service-example-00 to draft-worley-service-example-01
7.  References
    7.1.  Normative References
    7.2.  Informative References
§  Author's Address
§  Intellectual Property and Copyright Statements

1. Introduction

Within SIP[1] (Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” June 2002.)-based systems, it is desirable to be able to provide features that are similar to those provided by traditional telephony systems. A frequently requested feature is "music on hold": The music-on-hold feature is when one party to a call has the call "on hold", that party's telephone provides a media stream to be rendered to the other party.

Architectural features of SIP make it difficult to implement music-on-hold in a way that is fully compliant with the standards. The purpose of this document is to describe a method that is reasonably simple yet fully effective and standards-compliant.

2. Technique

The essence of the technique is that when the user's UA (referred to hereafter as the "executing UA") establishes the hold state, it uses third party call control mechanisms to direct RTP from an audio source service to the held party. The executing UA establishes a dialog with the service to negotiate the audio stream, but does not act as a media relay; media flows from the service to the held party.

In order to accomplish this within the offer/answer model, the executing UA sends a re-INVITE to the remote UA to establish the hold state, but in that INVITE it provides no SDP offer, thus compelling the remote UA to provide an SDP offer. The executing UA then uses that offer (modified as described in Section 2.4 (Managing o= Lines)) in a new INVITE to the external media source. The external media source is thus directed to provide media directly to the remote UA. The media source's answer SDP is returned to the remote UA in the ACK to the re-INVITE.

2.1. Placing a Call on Hold and Providing an External Media Stream

1. The executing user instructs the executing UA to put the dialog on-hold.
2. The executing UA sends a re-INVITE without SDP to the remote UA, which forces the remote UA to provide an SDP offer in its 2xx response. The Contact header of the re-INVITE SHOULD include the '+sip.rendering="no"' field parameter so that the intent is clear and so that any dialog events indicate that it is putting the call on hold.[6] (Rosenberg, J., Schulzrinne, H., and R. Mahy, “An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP),” November 2005.) (The sip.rendering feature parameter is not required by the MOH mechanics, but rather to facilitate reporting the on-hold status in dialog events.)
3. The remote UA sends a 2xx to the re-INVITE, and includes an SDP offer giving its own listening address/port. If the remote UA understands the sip.rendering feature parameter, the offer may indicate that it will not send media by specifying the media directionalities as "recvonly" (the reverse of "on-hold") or perhaps "inactive". But the remote UA may offer to send media.
4. The executing UA uses this offer to construct the offer SDP of an initial INVITE that it sends to the configured music-on-hold (MOH) source. The SDP in this request is, with the exceptions below, copied from the SDP returned by the remote UA in the previous step, particularly regarding the provided listening address/port and payload type numbers. The changes to the SDP offer are:
   • The o= line of the offer is modified according to Section 2.4 (Managing o= Lines).
   • Any explicit or implicit media directionality of "sendrecv" is replaced with "recvonly", and any media directionality of "sendonly" is replaced with "inactive".
5. The MOH sends a 2xx response to the INVITE, which contains an SDP answer that should include its media source address as its listening address/port. This SDP must necessarily[2] (Rosenberg, J. and H. Schulzrinne, “An Offer/Answer Model with the Session Description Protocol (SDP),” June 2002.) specify "sendonly" or "inactive" as the directionality for all media streams. (Although this address/port should receive no RTP, the answer SDP will reach the remote UA, informing it of the address/port from which the MOH media will come, and presumably preventing it from ignoring the MOH media as SPIT. This requires the SDP answer to provide a real address/port in the c= line, even though the MOH server does not receive RTP.)
6. The executing UA copies the MOH service SDP answer, modifying it as listed below, and sends it as its SDP answer in the ACK for the re-INVITE to the remote UA.
   • The o= line of the answer is modified according to Section 2.4 (Managing o= Lines).
7. Due to the sip.rendering feature parameter in the Contact of the re-INVITE and the media directionality in the SDP answer contained in the ACK, the on-hold state (at the executing end) of the dialog is established. (The sip.rendering feature parameter will be reported in the dialog events generated by both UAs, allowing any monitoring application to determine that the call is on hold.)
8. After this point, the MOH source generates RTP containing the music-on-hold media, and sends it directly to the listening address/port of the remote UA. The executing UA maintains two dialogs (one to the remote UA, one to the MOH source), but does not see or handle the MOH RTP.

2.2. Taking a Call off Hold and Terminating the External Media Stream

1. The executing user instructs the executing UA to take the dialog off-hold.
2. The executing UA sends a re-INVITE to the remote UA with SDP that requests to receive media. The Contact header of the re-INVITE does not include the '+sip.rendering="no"' field parameter. (It may contain a sip.rendering field parameter with value "yes" or "unknown", or it may omit the field parameter.) Thus this INVITE removes the on-hold state (at the executing end) of the dialog.
3. When the remote UA sends a 2xx response to the re-INVITE, the executing UA sends a BYE request in the dialog to the MOH source to terminate the MOH media stream.
4. After this point, the MOH source does not generate RTP and ordinary RTP flow is re-established in the original dialog.

2.3. Example Message Flow

This section shows a message flow which is an example of this technique. The scenario is: Alice establishes a call with Bob. Bob then places the call on hold, with music-on-hold provided from an external server. Bob then takes the call off hold.

Note that this is just one possible message flow that illustrates this technique; numerous variations on these operations are allowed by the applicable standards.

Alice             Bob       Music Server

Alice establishes the call:

  |                |              |
  |    INVITE F1   |              |
  |--------------->|              |
  | 180 Ringing F2 |              |
  |<---------------|              |
  |    200 OK F3   |              |
  |<---------------|              |
  |     ACK F4     |              |
  |--------------->|              |
  |       RTP      |              |
  |<==============>|              |
  |                |              |

Bob places Alice on hold, compelling Alice's UA to provide SDP:

  |                |              |
  |   INVITE F5    |              |
  |   (no SDP)     |              |
  |<---------------|              |
  |   200 OK F6    |              |
  |   (SDP offer)  |              |
  |--------------->|              |
  |                |              |

Bob's UA initiates music-on-hold:

  |                |              |
  |                |  INVITE F7   |
  |                |  (SDP offer, |
  |                |   rev. hold) |
  |                |------------->|
  |                | 200 OK F8    |
  |                | (SDP answer, |
  |                |  hold)       |
  |                |<-------------|
  |                |    ACK F9    |
  |                |------------->|
  |                |              |

Bob's UA provides SDP answer containing the address/port
of the Music Server:

  |                |              |
  | ACK (hold) F10 |              |
  | (SDP answer)   |              |
  |<---------------|              |
  |    no RTP      |              |
  |                |              |
  |     Music-on-hold RTP         |
  |<==============================|
  |                |              |

The music on hold is active.

Bob takes Alice off hold:

  |                |              |
  |  INVITE F11    |              |
  |  (SDP offer)   |              |
  |<---------------|              |
  |   200 OK F12   |              |
  |   (SDP answer) |              |
  |--------------->|              |
  |     ACK F13    |              |
  |<---------------|              |
  |                |    BYE F14   |
  |                |------------->|
  |                |    200 F15   |
  |                |<-------------|
  |       RTP      |              |
  |<==============>|              |
  |                |              |

The normal media session between Alice and Bob is resumed.

Message Details

 /* Alice calls Bob. */

 F1 INVITE Alice -> Bob

 INVITE sips:bob@biloxi.example.com SIP/2.0
 Via: SIP/2.0/TLS atlanta.example.com:5061
  ;branch=z9hG4bK74bf9
 Max-Forwards: 70
 From: Alice <sips:alice@atlanta.example.com>;tag=1234567
 To: Bob <sips:bob@biloxi.example.com>
 Call-ID: 12345600@atlanta.example.com
 CSeq: 1 INVITE
 Contact: <sips:a8342043f@atlanta.example.com;gr>
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces, gruu
 Content-Type: application/sdp
 Content-Length: [omitted]

 v=0
 o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
 s=
 c=IN IP4 atlanta.example.com
 t=0 0
 m=audio 49170 RTP/AVP 0
 a=rtpmap:0 PCMU/8000


 F2 180 Ringing Bob -> Alice

 SIP/2.0 180 Ringing
 Via: SIP/2.0/TLS atlanta.example.com:5061
  ;branch=z9hG4bK74bf9
  ;received=192.0.2.103
 From: Alice <sips:alice@atlanta.example.com>;tag=1234567
 To: Bob <sips:bob@biloxi.example.com>;tag=23431
 Call-ID: 12345600@atlanta.example.com
 CSeq: 1 INVITE
 Contact: <sips:bob@biloxi.example.com>
 Content-Length: 0


 F3 200 OK Bob -> Alice

 SIP/2.0 200 OK
 Via: SIP/2.0/TLS atlanta.example.com:5061
  ;branch=z9hG4bK74bf9
  ;received=192.0.2.103
 From: Alice <sips:alice@atlanta.example.com>;tag=1234567
 To: Bob <sips:bob@biloxi.example.com>;tag=23431
 Call-ID: 12345600@atlanta.example.com
 CSeq: 1 INVITE
 Contact: <sips:bob@biloxi.example.com>
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces
 Content-Type: application/sdp
 Content-Length: [omitted]

 v=0
 o=bob 2890844527 2890844527 IN IP4 biloxi.example.com
 s=
 c=IN IP4 biloxi.example.com
 t=0 0
 m=audio 3456 RTP/AVP 0
 a=rtpmap:0 PCMU/8000


 F4 ACK Alice -> Bob

 ACK sips:bob@biloxi.example.com SIP/2.0
 Via: SIP/2.0/TLS atlanta.example.com:5061
  ;branch=z9hG4bK74bfd
 Max-Forwards: 70
 From: Alice <sips:alice@atlanta.example.com>;tag=1234567
 To: Bob <sips:bob@biloxi.example.com>;tag=23431
 Call-ID: 12345600@atlanta.example.com
 CSeq: 1 ACK
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces
 Content-Length: 0


 /* Bob places Alice on hold. */

 /* The re-INVITE contains no SDP, thus compelling Alice's UA
    to provide an offer. */

 F5 INVITE Bob -> Alice

 INVITE sips:a8342043f@atlanta.example.com;gr SIP/2.0
 Via: SIP/2.0/TLS biloxi.example.com:5061
  ;branch=z9hG4bK874bk
 To: Alice <sips:alice@atlanta.example.com>;tag=1234567
 From: Bob <sips:bob@biloxi.example.com>;tag=23431
 Call-ID: 12345600@atlanta.example.com
 CSeq: 712 INVITE
 Contact: <sips:bob@biloxi.example.com>;+sip.rendering="no"
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces
 Content-Length: 0

 /* Alice's UA provides an SDP offer.
    Since it does not know that it is being put on hold,
    the offer is the same as the original offer and describes
    bidirectional media. */

 F6 200 OK Alice -> Bob

 SIP/2.0 200 OK
 Via: SIP/2.0/TLS biloxi.example.com:5061
  ;branch=z9hG4bK874bk
  ;received=192.0.2.105
 To: Alice <sips:alice@atlanta.example.com>;tag=1234567
 From: Bob <sips:bob@biloxi.example.com>;tag=23431
 Call-ID: 12345600@atlanta.example.com
 CSeq: 712 INVITE
 Contact: <sips:a8342043f@atlanta.example.com;gr>
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces, gruu
 Content-Type: application/sdp
 Content-Length: [omitted]

 v=0
 o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
 s=
 c=IN IP4 atlanta.example.com
 t=0 0
 m=audio 49170 RTP/AVP 0
 a=rtpmap:0 PCMU/8000
 a=active


 /* Bob's UA initiates music-on-hold. */

 /* This INVITE contains Alice's offer, but with the media
    direction set to "reverse hold", receive-only. */

 F7 INVITE Bob -> Music Server

 INVITE sips:music@server.example.com SIP/2.0
 Via: SIP/2.0/TLS biloxi.example.com:5061
  ;branch=z9hG4bKnashds9
 Max-Forwards: 70
 From: Bob <sips:bob@biloxi.example.com>;tag=02134
 To: Music Server <sips:music@server.example.com>
 Call-ID: 4802029847@biloxi.example.com
 CSeq: 1 INVITE
 Contact: <sips:bob@biloxi.example.com>
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces, gruu
 Content-Type: application/sdp
 Content-Length: [omitted]

 v=0
 o=bob 2890844534 2890844534 IN IP4 atlanta.example.com
 s=
 c=IN IP4 atlanta.example.com
 t=0 0
 m=audio 49170 RTP/AVP 0
 a=rtpmap:0 PCMU/8000
 a=recvonly


 F8 200 OK Music Server -> Bob

 SIP/2.0 200 OK
 Via: SIP/2.0/TLS biloxi.example.com:5061
  ;branch=z9hG4bKnashds9
  ;received=192.0.2.105
 From: Bob <sips:bob@biloxi.example.com>;tag=02134
 To: Music Server <sips:music@server.example.com>;tag=56323
 Call-ID: 4802029847@biloxi.example.com
 Contact: <sips:music@server.example.com>
 CSeq: 1 INVITE
 Content-Length: [omitted]

 v=0
 o=MusicServer 2890844576 2890844576 IN IP4 server.example.com
 s=
 c=IN IP4 server.example.com
 t=0 0
 m=audio 49170 RTP/AVP 0
 a=rtpmap:0 PCMU/8000
 a=sendonly


 F9 ACK Bob -> Music Server

 ACK sips:music@server.example.com SIP/2.0
 Via: SIP/2.0/TLS server.example.com:5061
  ;branch=z9hG4bK74bT6
 From: Bob <sips:bob@biloxi.example.com>;tag=02134
 To: Music Server <sips:music@server.example.com>;tag=56323
 Max-Forwards: 70
 Call-ID: 4802029847@biloxi.example.com
 CSeq: 1 ACK
 Content-Length: 0


 /* Bob's UA now sends the ACK that completes the re-INVITE
    to Alice and completes the SDP offer/answer.
    The ACK contains the SDP received from the Music Server,
    and thus contains the address/port from which the Music Server
    will send media. */

 F10 ACK Bob -> Alice

 ACK sips:a8342043f@atlanta.example.com;gr SIP/2.0
 Via: SIP/2.0/TLS biloxi.example.com:5061
  ;branch=z9hG4bKq874b
 To: Alice <sips:alice@atlanta.example.com>;tag=1234567
 From: Bob <sips:bob@biloxi.example.com>;tag=23431
 Call-ID: 12345600@atlanta.example.com
 CSeq: 712 ACK
 Contact: <sips:bob@biloxi.example.com>;+sip.rendering="no"
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces
 Content-Length: [omitted]

 v=0
 o=bob 2890844527 2890844528 IN IP4 biloxi.example.com
 s=
 c=IN IP4 server.example.com
 t=0 0
 m=audio 49170 RTP/AVP 0
 a=rtpmap:0 PCMU/8000
 a=sendonly

 /* Bob picks up the call by sending a re-INVITE to Alice. */

 F11 INVITE Bob -> Alice

 INVITE sips:a8342043f@atlanta.example.com;gr SIP/2.0
 Via: SIP/2.0/TLS biloxi.example.com:5061
  ;branch=z9hG4bK874bk
 To: Alice <sips:alice@atlanta.example.com>;tag=1234567
 From: Bob <sips:bob@biloxi.example.com>;tag=23431
 Call-ID: 12345600@atlanta.example.com
 CSeq: 713 INVITE
 Contact: <sips:bob@biloxi.example.com>
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces
 Content-Type: application/sdp
 Content-Length: [omitted]

 v=0
 o=bob 2890844527 2890844529 IN IP4 biloxi.example.com
 s=
 c=IN IP4 biloxi.example.com
 t=0 0
 m=audio 3456 RTP/AVP 0
 a=rtpmap:0 PCMU/8000


 F12 200 OK Alice -> Bob

 SIP/2.0 200 OK
 Via: SIP/2.0/TLS biloxi.example.com:5061
  ;branch=z9hG4bK874bk
  ;received=192.0.2.105
 To: Alice <sips:alice@atlanta.example.com>;tag=1234567
 From: Bob <sips:bob@biloxi.example.com>;tag=23431
 Call-ID: 12345600@atlanta.example.com
 CSeq: 713 INVITE
 Contact: <sips:a8342043f@atlanta.example.com;gr>
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces, gruu
 Content-Type: application/sdp
 Content-Length: [omitted]

 v=0
 o=alice 2890844526 2890844527 IN IP4 atlanta.example.com
 s=
 c=IN IP4 atlanta.example.com
 t=0 0
 m=audio 49170 RTP/AVP 0
 a=rtpmap:0 PCMU/8000


 F13 ACK Bob -> Alice

 ACK sips:a8342043f@atlanta.example.com;gr SIP/2.0
 Via: SIP/2.0/TLS biloxi.example.com:5061
  ;branch=z9hG4bKq874b
 To: Alice <sips:alice@atlanta.example.com>;tag=1234567
 From: Bob <sips:bob@biloxi.example.com>;tag=23431
 Call-ID: 12345600@atlanta.example.com
 CSeq: 713 ACK
 Contact: <sips:bob@biloxi.example.com>
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces
 Content-Length: 0


 F14 BYE Bob -> Music Server

 BYE sips:music@server.example.com SIP/2.0
 Via: SIP/2.0/TLS biloxi.example.com:5061
  ;branch=z9hG4bK74rf
 Max-Forwards: 70
 From: Bob <sips:bob@biloxi.example.com>;tag=02134
 To: Music Server <sips:music@server.example.com>;tag=56323
 Call-ID: 4802029847@biloxi.example.com
 CSeq: 2 BYE
 Contact: <sips:bob@biloxi.example.com>
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Supported: replaces, gruu
 Content-Length: [omitted]


 F15 200 OK Music Server -> Alice

 SIP/2.0 200 OK
 Via: SIP/2.0/TLS atlanta.example.com:5061
  ;branch=z9hG4bK74rf
  ;received=192.0.2.103
 From: Bob <sips:bob@biloxi.example.com>;tag=02134
 To: Music Server <sips:music@server.example.com>;tag=56323
 Call-ID: 4802029847@biloxi.example.com
 CSeq: 2 BYE
 Content-Length: 0


 /* Normal media session between Alice and Bob is resumed */

2.4. Managing o= Lines

The rulesSection 2.4 (Managing o= Lines) regarding the use of o= lines in successive SDP offers/answers during a dialog are quit strict. In particular, successive SDPs sent by one UA in the dialog must have identical o= lines, other than that the version number field must be incremented by 1. The single exception is that if two or more successive SDPs are exactly the same, they may have the same version number.

In order for the executing UA to conform to these rules, it must modify the o= lines in any SDP that it passes from one dialog to the other. In particular:

1. The SDP in the initial INVITE to to the MOH server must have its o= line replaced with a newly constructed o= line that is valid within the dialog with the MOH server -- See [3] (Handley, M., Jacobson, V., and C. Perkins, “SDP: Session Description Protocol,” July 2006.) for the details.
2. Within any later SDP copied from the original dialog to the MOH dialog, the o= line must be replaced with a copy of the o= line that the executing UA used in the previous SDP that it sent in the MOH dialog, with the version number incremented by 1.
3. Within any SDP copied from the MOH dialog to the original dialog, the o= line must be replaced with a copy of the o= line that the executing UA used in the previous SDP that it sent in the original dialog, with the version number incremented by 1.
4. When the executing UA takes the call off-hold, the o= line of the SDP that it sends in the re-INVITE must have a version number one greater than in the o= line of the last SDP the executing UA copied from the MOH dialog.

3. Advantages

This technique for providing music-on-hold has advantages over other methods now in use:

1. The original dialog is not transferred to another UA, so the "remote endpoint URI" displayed by the remote endpoint's user interface and dialog event package[6] (Rosenberg, J., Schulzrinne, H., and R. Mahy, “An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP),” November 2005.) does not change during the call.[4] (Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and K. Summers, “Session Initiation Protocol Service Examples,” July 2007.)
2. The music-on-hold media is sent directly from the music-on-hold source to the remote UA, rather than being relayed through the holding UA.
3. The remote UA sees, in the incoming SDP, the address/port that the MOH server will send MOH media from, thus allowing it to render the media, even if it is filtering incoming media based on originating address as a SPIT preventative.
4. The technique requires relatively simple manipulation of SDP, and in particular: (1) does not require a SIP agent to modify unrelated SDP to be acceptable to be sent within an already established sequence of SDP (a problem with [5] (Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and K. Summers, “Session Initiation Protocol Service Examples,” October 2006.)), and (2) does not require converting an SDP answer into an SDP offer (which was a problem with the -00 version of this document, as well as with [5] (Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and K. Summers, “Session Initiation Protocol Service Examples,” October 2006.)).

4. Security Considerations

Some UAs filter incoming media based on the address of origin in order to avoid SPIT. This technique ensures that any UA that should render MOH media will be informed of the source address via the SDP that it receives. This should allow such UAs to filter without interfering with MOH operation.

5. Acknowledgments

The original version of this proposal was derived from [5] (Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and K. Summers, “Session Initiation Protocol Service Examples,” October 2006.) and the similar implementation of MOH in the Snom UA. Significant improvements to sequence of operations, allowing improvements to the SDP handling, were suggested by Venkatesh[7] (Venkatesh, “Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-service-examples-11.txt,” October 2006.).

John Elwell[8] (Elwell, J., “Subject: [Sipping] RE: I-D Action:draft-worley-service-example-00.txt,” November 2007.) pointed out the need for the executing UA to pass through re-INVITEs/UPDATEs in order to allow ICE negotiation, which suggested to me the need for pass-through to handle the remote UA placing its end of the call on-hold.

This version benefitted from Scott Lawrence's careful reading and comments.

6. Revision History

6.1. Changes from draft-worley-service-example-00 to draft-worley-service-example-01

Removed the original "Example Message Flow" and promoted the "Alternative Example Message Flow" to replace it because of a number of flaws that were described in the discussion of -00 on the SIPPING mailing list.

Described the use of the sip.rendering feature parameter to indicate on-hold status.

Added Acknowledgments section.

Added separate section on the management of o= lines.

7. References

7.1. Normative References

7.2. Informative References

Author's Address

Intellectual Property Statement

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