Session Description Protocol (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)
draft-ietf-mmusic-sdp-cs-21
The information below is for an old version of the document.
| Document | Type | This is an older version of an Internet-Draft that was ultimately published as an RFC. | |
|---|---|---|---|
| Authors | Miguel Angel García , Simo Veikkolainen | ||
| Last updated | 2013-08-08 (Latest revision 2013-06-26) | ||
| Replaces | draft-garcia-mmusic-sdp-cs | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews |
SECDIR Last Call review
(of
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Has Issues
GENART Last Call review
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Ready with Issues
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||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Flemming Andreasen | ||
| IESG | IESG state | Approved-announcement sent | |
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | Gonzalo Camarillo | ||
| IESG note | Flemming Andreasen (fandreas@cisco.com) is the document shepherd. | ||
| Send notices to | mmusic-chairs@tools.ietf.org, draft-ietf-mmusic-sdp-cs@tools.ietf.org | ||
| IANA | IANA review state | Version Changed - Review Needed | |
| IANA action state | RFC-Ed-Ack | ||
| RFC Editor | RFC Editor state | IESG | |
| Details |
draft-ietf-mmusic-sdp-cs-21
MMUSIC WG M. Garcia-Martin
Internet-Draft Ericsson
Intended status: Standards Track S. Veikkolainen
Expires: December 28, 2013 Nokia
June 26, 2013
Session Description Protocol (SDP) Extension For Setting Audio and Video
Media Streams Over Circuit-Switched Bearers In The Public Switched
Telephone Network (PSTN)
draft-ietf-mmusic-sdp-cs-21
Abstract
This memo describes use cases, requirements, and protocol extensions
for using the Session Description Protocol (SDP) Offer/Answer model
for establishing audio and video media streams over circuit-switched
bearers in the Public Switched Telephone Network (PSTN).
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
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."
This Internet-Draft will expire on December 28, 2013.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Conventions Used in This Document . . . . . . . . . . . . . . 4
3. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 4
4. Overview of Operation . . . . . . . . . . . . . . . . . . . . 5
4.1. Example Call Flow . . . . . . . . . . . . . . . . . . . . 5
5. Protocol Description . . . . . . . . . . . . . . . . . . . . 7
5.1. Level of Compliance . . . . . . . . . . . . . . . . . . . 7
5.2. Extensions to SDP . . . . . . . . . . . . . . . . . . . . 7
5.2.1. Connection Data . . . . . . . . . . . . . . . . . . . 7
5.2.2. Media Descriptions . . . . . . . . . . . . . . . . . 9
5.2.3. Correlating the PSTN Circuit-Switched Bearer with SDP 10
5.2.3.1. The "cs-correlation" attribute . . . . . . . . . 11
5.2.3.2. Caller-ID Correlation Mechanism . . . . . . . . . 11
5.2.3.3. User-User Information Element Correlation
Mechanism . . . . . . . . . . . . . . . . . . . . 12
5.2.3.4. DTMF Correlation Mechanism . . . . . . . . . . . 14
5.2.3.5. Extensions to correlation mechanisms . . . . . . 15
5.3. Negotiating the correlation mechanisms . . . . . . . . . 15
5.3.1. Determining the Direction of the Circuit-Switched
Bearer Setup . . . . . . . . . . . . . . . . . . . . 15
5.3.2. Populating the cs-correlation attribute . . . . . . . 16
5.3.3. Considerations on correlations . . . . . . . . . . . 17
5.4. Considerations for Usage of Existing SDP . . . . . . . . 18
5.4.1. Originator of the Session . . . . . . . . . . . . . . 18
5.4.2. Contact information . . . . . . . . . . . . . . . . . 18
5.5. Considerations for Usage of Third Party Call Control
(3PCC) . . . . . . . . . . . . . . . . . . . . . . . . . 19
5.6. Offer/Answer mode extensions . . . . . . . . . . . . . . 19
5.6.1. Generating the Initial Offer . . . . . . . . . . . . 19
5.6.2. Generating the Answer . . . . . . . . . . . . . . . . 21
5.6.3. Offerer processing the Answer . . . . . . . . . . . . 24
5.6.4. Modifying the session . . . . . . . . . . . . . . . . 26
5.7. Formal Syntax . . . . . . . . . . . . . . . . . . . . . . 26
6. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 27
6.1. Single PSTN audio stream . . . . . . . . . . . . . . . . 27
6.2. Advanced SDP example: Circuit-Switched Audio and
Video Streams . . . . . . . . . . . . . . . . . . . . . . 29
7. Security Considerations . . . . . . . . . . . . . . . . . . . 31
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 32
8.1. Registration of new cs-correlation SDP attribute . . . . 32
8.2. Registration of a new "nettype" value . . . . . . . . . . 33
8.3. Registration of new "addrtype" value . . . . . . . . . . 33
8.4. Registration of a new "proto" value . . . . . . . . . . . 33
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9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 34
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 34
10.1. Normative References . . . . . . . . . . . . . . . . . . 34
10.2. Informative References . . . . . . . . . . . . . . . . . 34
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 36
1. Introduction
The Session Description Protocol (SDP) [RFC4566] is intended for
describing multimedia sessions for the purposes of session
announcement, session invitation, and other forms of multimedia
session initiation. SDP is most commonly used for describing media
streams that are transported over the Real-Time Transport Protocol
(RTP) [RFC3550], using the profiles for audio and video media defined
in RTP Profile for Audio and Video Conferences with Minimal Control
[RFC3551].
However, SDP can be used to describe other media transport protocols
than RTP. Previous work includes SDP conventions for describing ATM
bearer connections [RFC3108] and the Message Session Relay Protocol
[RFC4975].
SDP is commonly carried in Session Initiation Protocol (SIP)
[RFC3261] messages in order to agree on a common media description
among the endpoints. An Offer/Answer Model with Session Description
Protocol (SDP) [RFC3264] defines a framework by which two endpoints
can exchange SDP media descriptions and come to an agreement as to
which media streams should be used, along with the media related
parameters.
In some scenarios it might be desirable to establish the media stream
over a circuit-switched bearer connection even if the signaling for
the session is carried over an IP bearer. An example of such a
scenario is illustrated with two mobile devices capable of both
circuit-switched and packet-switched communication over a low-
bandwidth radio bearer. The radio bearer may not be suitable for
carrying real-time audio or video media, and using a circuit-switched
bearer would offer a better perceived quality of service. So,
according to this scenario, SDP and its higher layer session control
protocol (e.g., the Session Initiation Protocol (SIP) [RFC3261]) are
used over regular IP connectivity, while the audio or video is
received through the classical circuit-switched bearer.
This document addresses only the use of circuit-switched bearers in
the PSTN, not a generic circuit-switched network. The mechanisms
presented below require a call signaling protocol of the PSTN to be
used (such as ITU-T Q.931 [ITU.Q931.1998] or 3GPP TS 24.008
[TS.24.008]).
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Setting up a signaling relationship in the IP domain instead of just
setting up a circuit-switched call offers also the possibility of
negotiating in the same session other IP based media that is not
sensitive to jitter and delay, for example, text messaging or
presence information.
At a later point in time the mobile device might move to an area
where a high-bandwidth packet-switched bearer, for example a Wireless
Local Area Network (WLAN) connection, is available. At this point
the mobile device may perform a handover and move the audio or video
media streams over to the high-speed bearer. This implies a new
exchange of SDP Offer/Answer that leads to a re-negotiation of the
media streams.
Other use cases exist. For example, an endpoint might have at its
disposal circuit-switched and packet-switched connectivity, but the
same audio or video codecs are not feasible for both access networks.
For example, the circuit-switched audio or video stream supports
narrow-bandwidth codecs, while the packet-switched access allows any
other audio or video codec implemented in the endpoint. In this
case, it might be beneficial for the endpoint to describe different
codecs for each access type and get an agreement on the bearer
together with the remote endpoint.
There are additional use cases related to third party call control
where the session setup time is improved when the circuit-switched
bearer in the PSTN is described together with one or more codecs.
The rest of the document is structured as follows: Section 2 provides
the document conventions, Section 3 introduces the requirements,
Section 4 presents an overview of the proposed solutions, and
Section 5 contains the protocol description. Section 6 provides an
example of descriptions of circuit-switched audio or video streams in
SDP. Section 7 and Section 8 contain the Security and IANA
considerations, respectively.
2. Conventions Used in This Document
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP
14, RFC 2119 [RFC2119] and indicate requirement levels for compliant
implementations.
3. Requirements
This section presents the general requirements that are specific for
the audio or video media streams over circuit-switched bearers.
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REQ-1: A mechanism for endpoints to negotiate and agree on an audio
or video media stream established over a circuit-switched bearer
MUST be available.
REQ-2: The mechanism MUST allow the endpoints to combine circuit-
switched audio or video media streams with other complementary
media streams, for example, text messaging.
REQ-3: The mechanism MUST allow the endpoint to negotiate the
direction of the circuit-switched bearer, i.e., which endpoint is
active when initiating the circuit-switched bearer.
REQ-4: The mechanism MUST be independent of the type of the circuit-
switched access (e.g., Integrated Services Digital Network (ISDN),
Global System for Mobile Communication (GSM), etc.)
REQ-5: There MUST be a mechanism that helps an endpoint to correlate
an incoming circuit-switched bearer with the one negotiated in
SDP, as opposed to another incoming call that is not related to
that.
REQ-6: It MUST be possible for endpoints to advertise different
lists of audio or video codecs in the circuit-switched audio or
video stream from those used in a packet-switched audio or video
stream.
REQ-7: It MUST be possible for endpoints to not advertise the list
of available codecs for circuit-switched audio or video streams.
4. Overview of Operation
The mechanism defined in this memo extends SDP and allows describing
an audio or video media stream established over a circuit-switched
bearer. A new network type ("PSTN") and a new protocol type ("PSTN")
are defined for the "c=" and "m=" lines to be able to describe a
media stream over a circuit-switched bearer. These SDP extensions
are described in Section 5.2. Since circuit-switched bearers are
connection-oriented media streams, the mechanism re-uses the
connection-oriented extensions defined in RFC 4145 [RFC4145] to
negotiate the active and passive sides of a connection setup. This
is further described in Section 5.3.1.
4.1. Example Call Flow
Consider the example presented in Figure 1. In this example,
Endpoint A is located in an environment where it has access to both
IP and circuit-switched bearers for communicating with other
endpoints. Endpoint A decides that the circuit-switched bearer
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offers a better perceived quality of service for voice, and issues an
SDP Offer containing the description of an audio media stream over
circuit-switched bearer.
Endpoint A Endpoint B
| (1) SDP Offer (PSTN audio) |
|----------------------------------->|
| |
| (2) SDP Answer (PSTN audio) |
|<-----------------------------------|
| |
| PSTN call setup |
|<-----------------------------------|
| |
| |
|<===== media over PSTN bearer =====>|
| |
Figure 1: Example Flow
Endpoint B receives the SDP offer and determines that it is located
in an environment where the IP based bearer is not suitable for real-
time audio media. However, Endpoint B also has PSTN circuit-switched
bearer available for audio. Endpoint B generates an SDP answer
containing a description of the audio media stream over a circuit-
switched bearer.
During the offer-answer exchange Endpoints A and B also agree the
direction in which the circuit-switched bearer should be established.
In this example, Endpoint B becomes the active party, in other words,
it establishes the circuit-switched call to the other endpoint. The
Offer/Answer exchange contains identifiers or references that can be
used on the circuit-switched network for addressing the other
endpoint, as well as information that is used to determine that the
incoming circuit-switched bearer establishment is related to the
ongoing session between the two endpoints.
Endpoint B establishes a circuit-switched bearer towards Endpoint A
using whatever mechanisms are defined for the network type in
question. When receiving the incoming circuit-switched connection
attempt, Endpoint A is able to determine that the attempt is related
to the session it is just establishing with B.
Endpoint A accepts the circuit-switched connection; the circuit-
switched bearer setup is completed. The two endpoints can now use
the circuit-switched connection for two-way audio media.
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If, for some reason, Endpoint B would like to reject the offered
stream, it would set the port number of the specific stream to zero,
as specified in RFC3264 [RFC3264]. Also, if B does not understand
some of the SDP attributes specified in this document, it would
ignore them, as specified in RFC4566 [RFC4566].
5. Protocol Description
5.1. Level of Compliance
Implementations according to this specification MUST implement the
SDP extensions described in Section 5.2, and MUST implement the
considerations discussed in Section 5.3, Section 5.4 and Section 5.6.
5.2. Extensions to SDP
This section provides the syntax and semantics of the extensions
required for providing a description of audio or video media streams
over circuit-switched bearers in SDP.
5.2.1. Connection Data
According to SDP [RFC4566], the connection data line in SDP has the
following syntax:
c=<nettype> <addrtype> <connection-address>
where <nettype> indicates the network type, <addrtype> indicates the
address type, and the <connection-address> is the connection address,
which is dependent on the address type.
At the moment, the only network type defined is "IN", which indicates
Internet network type. The address types "IP4" and "IP6" indicate
the type of IP addresses.
This memo defines a new network type for describing a circuit-
switched bearer network type in the PSTN. The mnemonic "PSTN" is
used for this network type.
For the address type, we initially consider the possibility of
describing E.164 telephone numbers. We define a new "E164" address
type to be used within the context of a "PSTN" network type. The
"E164" address type indicates that the connection address contains an
E.164 number represented according to the ITU-T E.164 [ITU.E164.1991]
recommendation.
It is a common convention that an international E.164 number contains
a leading '+' sign. For consistency's sake, we also require the
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E.164 telephone is prepended with a '+', even if that is not
necessary for routing of the call in the PSTN network.
There are cases, though, when the endpoint is merely aware of a
circuit-switched bearer, without having further information about the
E.164 number allocated to it. In these cases a dash ("-") is used to
indicate an unknown connection address. This makes the connection
data line be according to the SDP syntax.
Please note that the "E164" address type defined in this memo is
exclusively defined to be used in conjunction with the "PSTN" network
type in accordance with [RFC4566]. Usage of "E164" address type in
conjunction with other network types may be defined elsewhere.
This memo exclusively uses the international representation of E.164
numbers, i.e., those including a country code and, as described above
prepended with a '+' sign. Implementations conforming to this
specification and using the "E164" address type together with the
"PSTN" network type MUST use the 'global-number-digits' construction
specified in RFC 3966 [RFC3966] for representing international E.164
numbers. This representation requires the presence of the '+' sign,
and additionally allows for the presence of one or more 'visual-
separator' constructions for easier human readability (see
Section 5.7).
Note that <connection-address> MUST NOT be omitted when unknown since
this would violate basic syntax of SDP [RFC4566]. In such cases, it
MUST be set to a "-".
The following are examples of the extension to the connection data
line:
c=PSTN E164 +441134960123
c=PSTN E164 -
When the <addrtype> is E164, the connection address is defined as
follows:
o an international E.164 number (prepended with a '+' sign)
o the value "-", signifying that the address is unknown
o any other value resulting from the production rule of connection-
address in RFC4566 [RFC4566], but in all cases any value
encountered will be ignored.
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5.2.2. Media Descriptions
According to SDP [RFC4566], the media description line in SDP has the
following syntax:
m=<media> <port> <proto> <fmt> ...
The <media> subfield carries the media type. For establishing an
audio bearer, the existing "audio" media type is used. For
establishing a video bearer, the existing "video" media type is used.
The <port> subfield is the transport port to which the media stream
is sent. Circuit-switched access lacks the concept of a port number,
and therefore the <port> subfield does not carry any meaningful
value. In order to be compliant with SDP syntax, implementations
SHOULD set the <port> subfield to the discard port value "9" and MUST
ignore it on reception.
According to RFC 3264 [RFC3264], a port number of zero in the offer
of a unicast stream indicates that the stream is offered but must not
be used. If a port number of zero is present in the answer of a
unicast stream, it indicates that the stream is rejected. These
rules are still valid when the media line in SDP represents a
circuit-switched bearer.
The <proto> subfield is the transport protocol. The circuit-switched
bearer uses whatever transport protocol it has available. This
subfield SHOULD be set to the mnemonic "PSTN" to be syntactically
correct with SDP [RFC4566] and to indicate the usage of circuit-
switched protocols in the PSTN.
The <fmt> subfield is the media format description. In the classical
usage of SDP to describe RTP-based media streams, when the <proto>
subfield is set to "RTP/AVP" or "RTP/SAVP", the <fmt> subfield
contains the payload types as defined in the RTP audio profile
[RFC3551].
When "RTP/AVP" is used in the <proto> field, the <fmt> subfield
contains the RTP payload type numbers. We use the <fmt> subfield to
indicate the list of available codecs over the circuit-switched
bearer, by re-using the conventions and payload type numbers defined
for RTP/AVP. The RTP audio and video media types, when applied to
PSTN circuit-switched bearers, represent merely an audio or video
codec. If the endpoint is able to determine the list of available
codecs for circuit-switched media streams, it MUST use the
corresponding payload type numbers in the <fmt> subfield.
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In some cases, the endpoint is not able to determine the list of
available codecs for circuit-switched media streams. In this case,
in order to be syntactically compliant with SDP [RFC4566], the
endpoint MUST include a single dash ("-") in the <fmt> subfield.
As per RFC 4566 [RFC4566], the media format descriptions are listed
in priority order.
Examples of media descriptions for circuit-switched audio streams
are:
m=audio 9 PSTN 3 0 8
m=audio 9 PSTN -
Similarly, an example of a media description for circuit-switched
video stream is:
m=video 9 PSTN 34
m=video 9 PSTN -
5.2.3. Correlating the PSTN Circuit-Switched Bearer with SDP
The endpoints should be able to correlate the circuit-switched bearer
with the session negotiated with SDP in order to avoid ringing for an
incoming circuit-switched bearer that is related to the session
controlled with SDP (and SIP).
Several alternatives exist for performing this correlation. This
memo provides three mutually non-exclusive correlation mechanisms.
Other correlation mechanisms may exist, and their usage will be
specified when need arises. All mechanisms share the same principle:
some unique information is sent in the SDP and in the circuit-
switched signaling protocol. If these pieces of information match,
then the circuit-switched bearer is part of the session described in
the SDP exchange. Otherwise, there is no guarantee that the circuit-
switched bearer is related to such session.
The first mechanism is based on the exchange of PSTN caller-ID
between the endpoints. The caller-ID is also available as the
Calling Party ID in the circuit-switched signaling.
The second mechanism is based on the inclusion in SDP of a value that
is also sent in the User-to-User Information Element that is part of
the bearer setup signaling in the PSTN.
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The third mechanism is based on sending in SDP a string that
represents Dual Tone MultiFrequency (DTMF) digits that will be later
sent right after the circuit-switched bearer is established.
Implementations MUST support the three correlation mechanisms
specified in Section 5.2.3.2, Section 5.2.3.3 and Section 5.2.3.4,
and SHOULD include all supported mechanisms the initial Offer.
5.2.3.1. The "cs-correlation" attribute
In order to provide support for the correlation mechanisms, we define
a new media-level SDP attribute called "cs-correlation". This "cs-
correlation" attribute can include any of the "callerid", "uuie", or
"dtmf" subfields, which specify additional information required by
the Caller-ID, User to User Information, or DTMF correlation
mechanisms, respectively. The list of correlation mechanisms may be
extended by other specifications, see Section 5.2.3.5 for more
details. There MUST be at most one "cs-correlation" attribute per
media description.
The following sections provide more detailed information of these
subfields.
The values "callerid", "uuie" and "dtmf" refer to the correlation
mechanisms defined in Section 5.2.3.2, Section 5.2.3.3, and
Section 5.2.3.4, respectively. The formal Augmented Backus-Naur
Format (ABNF) syntax of the "cs-correlation" attribute is presented
in Section 5.7.
5.2.3.2. Caller-ID Correlation Mechanism
The Caller-ID correlation mechanisms consists of an exchange of the
calling party number as an international E.164 number in SDP,
followed by the availability of the Calling Party Number information
element in the call setup signaling of the circuit switched
connection. If both pieces of information match, the circuit-
switched bearer is correlated to the session described in SDP.
Example of inclusion of an international E.164 number in the "cs-
correlation" attribute is:
a=cs-correlation:callerid:+441134960123
The presence of the "callerid" subfield indicates that the endpoint
supports use of the calling party number as a means of correlating a
PSTN call with the session being negotiated. The "callerid" subfield
MAY be accompanied by the international E.164 number of the party
inserting the parameter.
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Note that there are no guarantees that this correlation mechanism
works or is even available, due a number of problems:
o The endpoint might not be aware of its own E.164 number, in which
case it cannot populate the SDP appropriately.
o The Calling Party Number information element in the circuit-
switched signaling might not be available, e.g., due to policy
restrictions of the network operator or caller restriction due to
privacy.
o The Calling Party Number information element in the circuit-
switched signaling might be available, but the digit
representation of the E.164 number might differ from the one
expressed in the SDP, due to, e.g., lack of of country code. To
mitigate this problem implementations should consider only some of
the rightmost digits from the E.164 number for correlation. For
example, the numbers +44-113-496-0123 and 0113-496-0123 could be
considered as the same number. This is also the behavior of some
cellular phones, which correlate the incoming calling party with a
number stored in the phone book, for the purpose of displaying the
caller's name. Please refer to ITU-T E.164 reccommendation
[ITU.E164.1991] for consideration of the relevant number of digits
to consider.
5.2.3.3. User-User Information Element Correlation Mechanism
A second correlation mechanism is based on including in SDP a string
that represents the User-User Information Element that is part of the
call setup signaling of the circuit-switched bearer. The User-User
Information Element is specified in ITU-T Q.931 [ITU.Q931.1998] and
3GPP TS 24.008 [TS.24.008], among others. The User-User Information
Element has a maximum size of 35 or 131 octets, depending on the
actual message of the PSTN protocol where it is included and the
network settings.
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The mechanism works as follows: An endpoint creates a User-User
Information Element, according to the requirements of the call setup
signaling protocol. The same value is included in the SDP offer or
SDP answer, in the "uuie" subfield of the "cs-correlation" attribute.
When the SDP Offer/Answer exchange is completed, each endpoint has
become aware of the value that will be used in the User-User
Information Element of the call setup message of the PSTN protocol.
The endpoint that initiates the call setup attempt includes this
value in the User-User Information Element. The recipient of the
call setup attempt can extract the User-User Information Element and
correlate it with the value previously received in the SDP. If both
values match, then the call setup attempt corresponds to that
indicated in the SDP.
According to ITU-T Q.931 [ITU.Q931.1998], the User-User Information
Element (UUIE) identifier is composed of a first octet identifying
this as a User-User Information Element, a second octet containing
the Length of the user-user contents, a third octet containing a
Protocol Discriminator, and a value of up to 32 or 128 octets
(depending on network settings) containing the actual User
Information (see Figure 4-36 in ITU-T Q.931). The first two octets
of the UUIE MUST NOT be used for correlation, only the octets
carrying the Protocol Discriminator and the User Information value
are input to the creation of the value of the "uuie" subfield in the
"cs-correlation" attribute. Therefore, the value of the "uuie"
subfield in the "cs-correlation" attribute MUST start with the
Protocol Discriminator octet, followed by the User Information
octets. The value of the Protocol Discriminator octet is not
specified in this document; it is expected that organizations using
this technology will allocate a suitable value for the Protocol
Discriminator.
Once the binary value of the "uuie" subfield in the "cs-correlation"
attribute is created, it MUST be base 16 (also known as "hex")
encoded before it is inserted in SDP. Please refer to RFC 4648
[RFC4648] for a detailed description of base 16 encoding. The
resulting encoded value needs to have an even number of hexadecimal
digits, and MUST be considered invalid if it has an odd number.
Note that the encoding of the "uuie" subfield of the "cs-
correlation" attribute is largely inspired by the encoding of the
same value in the User-to-User header field in SIP, according to
the document "A Mechanism for Transporting User to User Call
Control Information in SIP" [I-D.ietf-cuss-sip-uui].
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As an example, an endpoint willing to send a UUIE containing a
protocol discriminator with the hexadecimal value of %x56 and an
hexadecimal User Information value of %xA390F3D2B7310023 would
include a "cs-correlation" attribute line as follows:
a=cs-correlation:uuie:56A390F3D2B7310023
Note that the value of the User-User Information Element is
considered as an opaque string and only used for correlation
purposes. Typically call signaling protocols impose requirements on
the creation of User-User Information Element for end-user protocol
exchange. The details regarding the generation of the User-User
Information Element are outside the scope of this specification.
Please note that there are no guarantees that this correlation
mechanism works. On one side, policy restrictions might not make the
User-User information available end to end in the PSTN. On the other
hand, the generation of the User-User Information Element is
controlled by the PSTN circuit-switched call protocol, which might
not offer enough freedom for generating different values from one
endpoint to another one, or from one call to another in the same
endpoint. This might result in the same value of the User-User
Information Element for all calls.
5.2.3.4. DTMF Correlation Mechanism
We introduce a third mechanism for correlating the circuit-switched
bearer with the session described with SDP. This is based on
agreeing on a sequence of digits that are negotiated in the SDP Offer
/Answer exchange and sent as Dual Tone Multifrequency (DTMF) ITU-T
Recommendation Q.23 [ITU.Q23.1988] tones over the circuit-switched
bearer once this bearer is established. If the DTMF digit sequence
received through the circuit-switched bearer matches the digit string
negotiated in the SDP, the circuit-switched bearer is correlated with
the session described in the SDP. The mechanism is similar to many
voice conferencing systems which require the user to enter a PIN code
using DTMF tones in order to be accepted in a voice conference.
The mechanism works as follows: An endpoint selects a DTMF digit
sequence. The same sequence is included in the SDP offer or SDP
answer, in a "dtmf" subfield of the "cs-correlation" attribute. When
the SDP Offer/Answer exchange is completed, each endpoint has become
aware of the DTMF sequence that will be sent right after the circuit-
switched bearer is set up. The endpoint that initiates the call
setup attempt sends the DTMF digits according to the procedures
defined for the circuit-switched bearer technology used. The
recipient (passive side of the bearer setup) of the call setup
attempt collects the digits and compares them with the value
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previously received in the SDP. If the digits match, then the call
setup attempt corresponds to that indicated in the SDP.
Implementations are advised to select a number of DTMF digits that
provide enough assurance that the call is related, but on the
other hand do not prolong the bearer setup time unnecessarily. A
number of 5 to 10 digits is a good compromise.
As an example, an endpoint willing to send DTMF tone sequence "14D*3"
would include a "cs-correlation" attribute line as follows:
a=cs-correlation:dtmf:14D*3
If the endpoints successfully agree on the usage of the DTMF digit
correlation mechanism, but the passive side does not receive any DTMF
digits after successful circuit-switched bearer setup, or receives a
set of DTMF digits that do not match the value of the "dtmf"
attribute (including receiving too many digits), the passive side
SHOULD consider that this DTMF mechanism has failed to correlate the
incoming call.
5.2.3.5. Extensions to correlation mechanisms
New values for the "cs-correlation" attribute may be specified. The
registration policy for new values is "Specification Required", see
Section 8. Any such specification MUST include a description of how
SDP Offer/Answer mechanism is used to negotiate the use of the new
values, taking into account how endpoints determine which side will
become active or passive (see Section 5.3 for more details).
If, during the Offer/Answer negotiation, either endpoint encounters
an unknown value in the "cs-correlation" attribute, it MUST consider
that mechanism as unsupported, and MUST NOT include that value in
subsequent Offer/Answer negotiation.
5.3. Negotiating the correlation mechanisms
The three correlation mechanisms presented above (based on called
party number, User-User Information Element and DTMF digit sending)
are non-exclusive, and can be used independently of each other. In
order to know how to populate the "cs-correlation" attribute, the
endpoints need to agree which endpoint will become the active party,
i.e., the one that will set up the circuit-switched bearer.
5.3.1. Determining the Direction of the Circuit-Switched Bearer Setup
In order to avoid a situation where both endpoints attempt to
initiate a connection simultaneously, the direction in which the
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circuit-switched bearer is set up MUST be negotiated during the Offer
/Answer exchange.
The framework defined in RFC 4145 [RFC4145] allows the endpoints to
agree which endpoint acts as the active endpoint when initiating a
TCP connection. While RFC 4145 [RFC4145] was originally designed for
establishing TCP connections, it can be easily extrapolated to the
connection establishment of circuit-switched bearers. This
specification uses the concepts specified in RFC 4145 [RFC4145] for
agreeing on the direction of establishment of a circuit-switched
bearer.
RFC 4145 [RFC4145] defines two new attributes in SDP: "setup" and
"connection". The "setup" attribute indicates which of the endpoints
should initiate the connection establishment of the PSTN circuit-
switched bearer. Four values are defined in Section 4 of RFC 4145
[RFC4145]: "active", "passive", "actpass", "holdconn". Please refer
to Section 4 of RFC 4145 [RFC4145] for a detailed description of this
attribute.
The "connection" attribute indicates whether a new connection is
needed or an existing connection is reused. The attribute can take
the values "new" or "existing". Please refer to Section 5 of RFC
4145 [RFC4145] for a detailed description of this attribute.
Implementations according to this specification MUST support the
"setup" and "connection" attributes specified in RFC 4145 [RFC4145],
but applied to circuit-switched bearers in the PSTN.
We define the active party as the one that initiates the circuit-
switched bearer after the Offer/Answer exchange. The passive party
is the one receiving the circuit-switched bearer. Either party may
indicate its desire to become the active or passive party during the
Offer/Answer exchange using the procedures described in Section 5.6.
5.3.2. Populating the cs-correlation attribute
By defining values for the subfields in the "a=cs-correlation"
attribute, the endpoint indicates that it is willing to become the
active party, and that it can use those values in the Calling party
number, User-User Information Element, or as DTMF tones during the
circuit-switched bearer setup.
Thus, the following rules apply:
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An endpoint that can only become the active party in the circuit-
switched bearer setup MUST include all correlation mechanisms it
supports in the "a=cs-correlation" attribute, and MUST also
specify values for the subfields.
An endpoint that can only become the passive party in the circuit-
switched bearer setup MUST include all correlation mechanisms it
supports in the "a=cs-correlation" attribute, but MUST NOT specify
values for the subfields.
An endpoint that is willing to become either the active or passive
party (by including the "a=setup:actpass" attribute in the Offer),
MUST include all correlation mechanisms it supports in the "a=cs-
correlation" attribute, and MUST also specify values for the
subfields.
5.3.3. Considerations on correlations
Passive endpoints should expect an incoming CS call for setting up
the audio bearer. Passive endpoints MAY suppress the incoming CS
alert during a certain time periods. Additional restrictions can be
applied, such as the passive endpoint not alerting incoming calls
originated from the number that was observed during the offer/answer
negotiation.
There may be cases when an endpoint is not willing to include all
three correlation mechanisms in the "a=cs-correlation" attribute line
even if it supports all three mechansism. For example, some
correlation mechanism can be omitted if the endpoint is certain that
the PSTN network does not support carrying the correlation
identifier. Also, since using the DTMF based correlation mechanism
requires the call to be accepted before DTMF tones cane be sent, some
endpoints may enforce a policy restricting this due to for example
cost associated with received calls, making the DTMF based mechanism
unusable.
Note that it cannot be guaranteed that any given correlation
mechanism will succeed even if the usage of those was agreed
beforehand. This is due to the fact that the correlation mechanisms
require support from the circuit-switched bearer technology used.
Therefore, even a single positive indication using any of these
mechanisms SHOULD be interpreted by the passive endpoint so that the
circuit-switched bearer establishment is related to the ongoing
session, even if the other correlation mechanisms fail.
If, after negotiating one or more correlation mechanisms in the SDP
offer/answer exchange, an endpoint receives a circuit-switched bearer
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with no correlation information present, the endpoint has two
choices: it can either treat the call as unrelated, or treat the call
as related to the ongoing session in the IP domain.
An endpoint may for example specify a time window after SDP offer/
answer exchange during which received calls are treated as correlated
even if the signaling in the circuit-switched domain does not carry
any correlation information. In this case, there is a chance that
the call is erroneously treated as related to the ongoing session.
An endpoint may also choose to always treat an incoming call as
unrelated if the signaling in the circuit-switched domain does not
carry any correlation information. In this case, there is a chance
that the call is erroneously treated as unrelated.
Since, in these cases, no correlation information can be deduced from
the signaling, it is up to the implementation to decide how to
behave. One option is also to let the user decide whether to accept
the call as related, or to treat the call as unrelated.
5.4. Considerations for Usage of Existing SDP
5.4.1. Originator of the Session
According to SDP [RFC4566], the origin line in SDP has the following
syntax:
o=<username> <sess-id> <sess-version> <nettype> <addrtype>
<unicast-address>
Of interest here are the <nettype> and <addrtype> fields, which
indicate the type of network and type of address, respectively.
Typically, this field carries the IP address of the originator of the
session. Even if the SDP was used to negotiate an audio or video
media stream transported over a circuit-switched bearer, the
originator is using SDP over an IP bearer. Therefore, <nettype> and
<addrtype> fields in the "o=" line should be populated with the IP
address identifying the source of the signaling.
5.4.2. Contact information
SDP [RFC4566] defines the "p=" line which may include the phone
number of the person responsible for the conference. Even though
this line can carry a phone number, it is not suited for the purpose
of defining a connection address for the media. Therefore, we have
selected to define the PSTN specific connection addresses in the "c="
line.
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5.5. Considerations for Usage of Third Party Call Control (3PCC)
Best Current Practices for Third Party Call Control (3pcc) in the
Session Initiation Protocol (SIP) [RFC3725] outlines several flows
which are possible in third party call control scenarios and
recommends some flows for specific situations.
One of the assumptions in [RFC3725] is that an SDP Offer may include
a "black hole" connection address, which has the property that
packets sent to it will never leave the host which sent them. For
IPv4, this "black hole" connection address is 0.0.0.0, or a domain
name within the .invalid DNS top level domain.
When using an E.164 address scheme in the context of third-party call
control, when the User Agent needs to indicate an unknown phone
number, it MUST populate the <addrtype> of the SDP "c=" line with a
"-" string.
Note that this may result in the recipient of the initial offer
rejecting such offer if the recipient of the offer was not aware
of its own E.164 number. Consequently it will not be possible to
establish a circuit-switched bearer, since neither party is aware
of their E.164 number.
5.6. Offer/Answer mode extensions
In this section, we define extensions to the Offer/Answer model
defined in The Offer/Answer Model in SDP [RFC3264] to allow for PSTN
addresses to be used with the Offer/Answer model.
5.6.1. Generating the Initial Offer
The Offerer, wishing to use PSTN audio or video stream, MUST populate
the "c=" and "m=" lines as follows.
The endpoint MUST set the <nettype> in the "c=" line to "PSTN", and
the <addrtype> to "E164". Furthermore, the endpoint SHOULD set the
<connection-address> field to its own international E.164 number
(with a leading "+"). If the endpoint is not aware of its own E.164
number, it MUST set the <connection-address> to "-".
In the "m=" line, the endpoint MUST set the <media> subfield to
"audio" or "video", depending on the media type, and the <proto>
subfield to "PSTN". The <port> subfield SHOULD be set to "9" (the
discard port). The values "audio" or "video" in the <media> subfield
MUST NOT be set by the endpoint unless it has knowledge that these
bearer types are available on the circuit-switched network.
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The <fmt> subfield carries the payload type number(s) the endpoint is
wishing to use. Payload type numbers in this case refer to the
codecs that the endpoint wishes to use on the PSTN media stream. For
example, if the endpoint wishes to use the GSM codec, it would add
payload type number 3 in the list of codecs. The list of payload
types MUST only contain those codecs the endpoint is able to use on
the PSTN bearer. In case the endpoint is not aware of the codecs
available for the circuit-switched media streams, it MUST include a
dash ("-") in the <fmt> subfield.
The mapping table of static payload types numbers to payload types is
initally specified in [RFC3551] and maintained by IANA. For dynamic
payload types, the endpoint MUST define the set of valid encoding
names and related parameters using the "a=rtpmap" attribute line.
See Section 6 of RFC 4566 [RFC4566] for details.
When generating the Offer, the Offerer MUST include an attribute line
"a=cs-correlation" in the SDP offer. The Offerer MUST NOT include
more than one "cs-correlation" attribute per media decription. The
"a=cs-correlation" line SHOULD contain an enumeration of all the
correlation mechanisms supported by the Offerer, in the format of
subfields. See Section 5.3.3 for more information on usage of the
correlation mechanisms.
The current list of subfields include "callerid", "uuie" and "dtmf"
and they refer to the correlation mechanisms defined in
Section 5.2.3.2, Section 5.2.3.3, and Section 5.2.3.4, respectively.
If the Offerer supports any of the correlation mechanisms defined in
this memo, and is willing to become the active party, the Offerer
MUST add the "callerid", "uuie", and/or "dtmf" subfields and MUST
specify values for those subfields.
o the international E.164 number as the value in the "callerid"
subfield,
o the contents of the User-User information element as the value of
the "uuie" subfield, and/or
o the DTMF tone string as the value of the "dtmf" subfield
If the Offerer is only able to become the passive party in the
circuit-switched bearer setup, it MUST add at least one of the
possible correlation mechanisms, but MUST NOT specify values for
those subfields.
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For example, if the Offerer is willing to use the User-User
Information element and DTMF digit sending mechanisms, but can only
become the passive party, it includes the following lines in the SDP:
a=cs-correlation:uuie dtmf
a=setup:passive
If, on the other hand, the Offerer is willing to use the User-User
Information element and the DTMF correlation mechanisms, and is able
to become the active or passive side, it includes the following lines
in the SDP:
a=cs-correlation:uuie:56A390F3D2B7310023 dtmf:14D*3
a=setup:actpass
The negotiation of the value of the 'setup' attribute takes place as
defined in Section 4.1 of RFC4145 [RFC4145].
The Offerer states which role or roles it is willing to perform; and
the Answerer, taking the Offerer's willingness into consideration,
chooses which roles both endpoints will actually perform during the
circuit-switched bearer setup.
By 'active' endpoint, we refer to an endpoint that will establish the
circuit-switched bearer; and by 'passive' endpoint, we refer to an
endpoint that will receive a circuit-switched bearer.
If an Offerer does not know its international E.164 number, it MUST
set the 'a=setup' attribute to the value 'active'. If the Offerer
knows its international E.164 number, it SHOULD set the value to
either 'actpass' or 'passive'.
Also 'holdconn' is a permissible value in the 'a=setup' attribute.
It indicates that the connection should not be established for the
time being.
The Offerer uses the "a=connection" attribute to decide whether a new
circuit-switched bearer is to be established or not. For the initial
Offer, the Offerer MUST use value 'new'.
5.6.2. Generating the Answer
If the Offer contained a circuit-switched audio or video stream, the
Answerer first determines whether it is able to accept and use such
streams on the circuit-switched network. If the Answerer does not
support or is not willing to use circuit-switched media for the
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session, it MUST construct an Answer where the port number for such
media stream(s) is set to zero, according to Section 6 of [RFC3264].
If the Answerer is willing to use circuit-switched media for the
session, it MUST ignore the received port number (unless the port
number is set to zero).
If the Offer included a "-" as the payload type number, it indicates
that the Offerer is not willing or able to define any specific
payload type. Most often, a "-" is expected to be used instead of
the payload type when the endpoint is not aware of or not willing to
define the codecs which will eventually be used on the circuit-
switched bearer. The circuit-switched signaling protocols have their
own means of negotiating or indicating the codecs, therefore an
Answerer SHOULD accept such Offers, and SHOULD set the payload type
to "-" also in the Answer.
If the Answerer explicitly wants to specify a codec for the circuit-
switched media, it MAY set the respective payload numbers in the
<fmt> subfield in the answer. This behavior, however, is NOT
RECOMMENDED.
When receiving the Offer, the Answerer MUST determine whether it
becomes the active or passive party.
If the SDP in the Offer indicates that the Offerer is only able to
become the active party, the Answerer needs to determine whether it
is able to become the passive party. If this is not possible e.g.
due to the Answerer not knowing its international E.164 number, the
Answerer MUST reject the circuit-switched media by setting the port
number to zero on the Answer. If the Answerer is aware of its
international E.164 number, it MUST include the "a=setup" attribute
in the Answer and set it to value "passive" or "holdconn". The
Answerer MUST also include its E.164 number on the "c=" line.
If the SDP in the Offer indicates that the Offerer is only able to
become the passive party, the Answerer MUST verify that the Offerer's
E.164 number is included in the "c=" line of the Offer. If the
number is included, the Answerer MUST include the "a=setup" attribute
in the Answer and set it to value "active" or "holdconn". If the
number is not included, or the recipient of the Offer is not willing
to establish a connection the E.164 based on a priori knowledge of
cost, or other reasons, call establishment is not possible, and the
Answerer MUST reject the circuit-switched media by setting the port
number to zero in the Answer.
If the SDP in the Offer indicates that the Offerer is able to become
either the active or passive party, the Answerer determines which
role it will take. If the Offer includes an international E.164
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number in the "c=" line, the Answerer SHOULD become the active party.
If the Answerer does not become the active party, and if the Answerer
is aware of its E.164 number, it MUST become the passive party. If
the Answerer does not become the active or the passive party, it MUST
reject the circuit-switched media by setting the port number to zero
in the Answer.
For each media description where the Offer includes a "a=cs-
correlation" attribute, the Answerer MUST select from the Offer those
correlation mechanisms it supports, and include in the Answer one "a
=cs-correlation" attribute line containing those mechanisms it is
willing to use. The Answerer MUST only add one "a=cs-correlation"
attribute in those media descriptions where also the Offer included a
"a=cs-correlation" attribute. The Answerer MUST NOT add any
mechanisms which were not included in the offer. If there are more
than one "cs-correlation" attributes per media description in the
Offer, the Answerer MUST discard all but the first for any media
description. Also, the Answerer MUST discard all unknown "cs-
correlation" attribute values.
If the Answerer becomes the active party, it MUST add a value to any
of the possible subfields.
If the Answerer becomes the passive party, it MUST NOT add any values
to the subfields in the "cs-correlation" attribute.
After generating and sending the Answer, if the Answerer became the
active party, it
o MUST extract the E.164 number from the "c=" line of the Offer and
MUST establish a circuit-switched bearer to that address.
o if the SDP Answer contained a value for the "callerid" subfield,
MUST set the Calling Party Number Information Element to that
number,
o if the SDP Answer contained a value for the "uuie" subfield, MUST
send the User-User Information element according to the rules
defined for the circuit-switched technology used, and set the
value of the Information Element to that received in the SDP
Offer,
o if the SDP Answer contained a value for the "dtmf" subfield, MUST
send those DTMF digits according to the circuit-switched
technology used.
If, on the other hand, the Answerer became the passive party, it
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o MUST be prepared to receive a circuit-switched bearer,
o if the Offer contained a value for the "callerid" subfield, MUST
compare that value to the Calling Party Number Information Element
of the circuit-switched bearer. If the received Calling Party
Number Information Element matches the value of the "callerid"
subfield, the call SHOULD be treated as correlated to the ongoing
session.
o if the Offer contained a value for the "dtmf" subfield, MUST be
prepared to receive and collect DTMF digits once the circuit-
switched bearer is set up. The Answerer MUST compare the received
DTMF digits to the value of the "dtmf" subfield. If the received
DTMF digits match the value of the "dtmf" subfield in the "cs-
correlation" attribute, the call SHOULD be treated as correlated
to the ongoing session.
o if the Offer contained a value for the "uuie" subfield, MUST be
prepared to receive a User-User Information element once the
circuit-switched bearer is set up. The Answerer MUST compare the
received UUI to the value of the "uuie" subfield. If the value of
the received UUI matches the value of the "uuie" subfield, the
call SHOULD be treated as correlated to the ongoing session.
If the Answerer becomes the active party, generates an SDP answer,
and then it finds out that the circuit-switched call cannot be
established, then the Answerer MUST create a new SDP offer where
circuit-switched stream is removed from the session (actually, by
setting the corresponding port in the m= line to zero) and send it to
its counterpart. This is to synchronize both parties (and potential
intermediaries) on the state of the session.
5.6.3. Offerer processing the Answer
When receiving the Answer, if the SDP does not contain "a=cs-
correlation" attribute line, the Offerer should take that as an
indication that the other party does not support or is not willing to
use the procedures defined in the document for this session, and MUST
revert to normal processing of SDP.
When receiving the Answer, the Offerer MUST first determine whether
it becomes the active or passive party, as described in
Section 5.3.1.
If the Offerer becomes the active party, it
o MUST extract the E.164 number from the "c=" line and MUST
establish a circuit-switched bearer to that address.
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o if the SDP Answer contained a value for the "uuie" subfield, MUST
send the User-User Information element according to the rules
defined for the circuit-switched technology used, and set the
value of the Information Element to that received in the SDP
Answer,
o if the SDP Answer contained a value for the "dtmf" subfield, MUST
send those DTMF digits according to the circuit-switched
technology used.
If the Offerer becomes the passive party, it
o MUST be prepared to receive a circuit-switched bearer,
o Note that if delivery of the Answer is delayed for some reason,
the circuit-switched call attempt may arrive at the Offerer before
the Answer has been processed. In this case, since the
correlation mechanisms are negotiated as part of the Offer/Answer
exchange, the Answerer cannot know whether or not the incoming
circuit-switched call attempt is correlated with the session being
negotiated, the Offerer SHOULD answer the circuit-switched call
attempt only after it has received and processed the Answer.
o If the Answer contained a value for the "dtmf" subfield, the
Offerer MUST be prepared to receive and collect DTMF digits once
the circuit-switched bearer is set up. The Offerer SHOULD compare
the received DTMF digits to the value of the "dtmf" subfield. If
the received DTMF digits match the value of the "dtmf" subfield in
the "cs-correlation" attribute, the call SHOULD be treated as
correlated to the ongoing session.
o If the Answer contained a value for the "uuie" subfield, the
Offerer MUST be prepared to receive a User-User Information
element once the circuit-switched bearer is set up. The Offerer
SHOULD compare the received UUI to the value of the "uuie"
subfield. If the value of the received UUI matches the value of
the "uuie" subfield, the call SHOULD be treated as correlated to
the ongoing session.
According the Offer/Answer Model with SDP [RFC3264], the Offerer
needs to be ready to receive media as soon as the Offer has been
sent. It may happen that the Answerer, if it became the active
party, will initiate a circuit-switched bearer setup which will
arrive at the Offerer before the Answer has arrived. However, the
Offerer needs to receive the Answer and examine the information about
the correlation mechanisms in order to successfully perform
correlation of the circuit-switched call to the session. Therefore,
if the Offerer receives an incoming circuit-switched call, it MUST
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NOT accept the call before the Answer has been received. If no
Answer is received during an implementation specific time, the
Offerer MUST either modify the session according to [RFC3264] or
terminate it according to the session signaling procedures in
question (for terminating a SIP session, see Section 15 of
[RFC3261]).
5.6.4. Modifying the session
If, at a later time, one of the parties wishes to modify the session,
e.g., by adding new media stream, or by changing properties used on
an existing stream, it may do so via the mechanisms defined for An
Offer/Answer Model with SDP [RFC3264].
If there is an existing circuit-switched bearer between the
endpoints, and the Offerer wants to reuse that, the Offerer MUST set
the value of the "a=connection" attribute to 'existing'.
If either party removes the circuit-switched media from the session
(by setting the port number to zero), it MUST terminate the circuit-
switched bearer using whatever mechanism is appropriate for the
technology in question.
If either party wishes to drop and reestablish an existing call, that
party MUST first remove the circuit-switched media from the session
by setting the port number to zero, and then use another Offer/Answer
exchange where it MUST set the "a=connection" attribute to 'new'".
If the media types are different (for example, a different codec will
be used for the circuit-switched bearer), the media descriptions for
terminating the existing bearer and the new bearer can be in the same
Offer.
5.7. Formal Syntax
The following is the formal Augmented Backus-Naur Form (ABNF)
[RFC5234] syntax that supports the extensions defined in this
specification. The syntax is built above the SDP [RFC4566] and the
tel URI [RFC3966] grammars. Implementations according to this
specification MUST be compliant with this syntax.
Figure 2 shows the formal syntax of the extensions defined in this
memo.
; extension to the connection field originally specified
; in RFC4566
connection-field = [%x63 "=" nettype SP addrtype SP
connection-address CRLF]
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; CRLF defined in RFC5234
;nettype and addrtype are defined in RFC 4566
connection-address /= global-number-digits / "-"
; global-number-digits specified in RFC3966
;subrules for correlation attribute
attribute /= cs-correlation-attr
; attribute defined in RFC4566
cs-correlation-attr = "cs-correlation:" corr-mechanisms
corr-mechanisms = corr-mech *(SP corr-mech)
corr-mech = caller-id-mech / uuie-mech /
dtmf-mech / ext-mech
caller-id-mech = "callerid" [":" caller-id-value]
caller-id-value = "+" 1*15DIGIT
; DIGIT defined in RFC5234
uuie-mech = "uuie" [":" uuie-value]
uuie-value = 1*65(HEXDIG HEXDIG)
;This represents up to 130 HEXDIG
; (65 octets)
;HEXDIG defined in RFC5234
;HEXDIG defined as 0-9, A-F
dtmf-mech = "dtmf" [":" dtmf-value]
dtmf-value = 1*32(DIGIT / %x41-44 / %x23 / %x2A )
;0-9, A-D, '#' and '*'
ext-mech = ext-mech-name [":" ext-mech-value]
ext-mech-name = token
ext-mech-value = token
; token is specified in RFC4566
Figure 2: Syntax of the SDP extensions
6. Examples
In the examples below, where an SDP line is too long to be displayed
as a single line, a breaking character "\" indicates continuation in
the following line. Note that this character is included for display
purposes only. Implementations MUST write a single line without
breaks.
6.1. Single PSTN audio stream
Endpoint A Endpoint B
| |
| (1) SDP Offer (PSTN audio) |
|--------------------------------->|
| |
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| (2) SDP Answer (PSTN audio) |
|<---------------------------------|
| |
| PSTN call setup |
|<---------------------------------|
| |
|<==== media over PSTN bearer ====>|
| |
Figure 3: Basic flow
Figure 3 shows a basic example that describes a single audio media
stream over a circuit-switched bearer. Endpoint A generates a SDP
Offer which is shown in Figure 4. The Offer describes a PSTN
circuit-switched bearer in the "m=" and "c=" line where it also
indicates its international E.164 number format. Additionally,
Endpoint A expresses that it can initiate the circuit-switched bearer
or be the recipient of it in the "a=setup" attribute line. The SDP
Offer also includes correlation identifiers that this endpoint will
insert in the Calling Party Number and/or User-User Information
Element of the PSTN call setup if eventually this endpoint initiates
the PSTN call.
v=0
o=alice 2890844526 2890842807 IN IP4 192.0.2.5
s=
t=0 0
m=audio 9 PSTN -
c=PSTN E164 +441134960123
a=setup:actpass
a=connection:new
a=cs-correlation:callerid:+441134960123 \
uuie:56A390F3D2B7310023
Figure 4: SDP offer (1)
Endpoint B generates a SDP Answer (Figure 5), describing a PSTN audio
media on port 9 without information on the media sub-type on the "m="
line. The "c=" line contains B's international E.164 number. In the
"a=setup" line Endpoint B indicates that it is willing to become the
active endpoint when establishing the PSTN call, and it also includes
the "a=cs-correlation" attribute line containing the values it is
going to include in the Calling Party Number and User-User IE of the
PSTN call establishment.
v=0
o=- 2890973824 2890987289 IN IP4 192.0.2.7
s=
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t=0 0
m=audio 9 PSTN -
c=PSTN E164 +441134960124
a=setup:active
a=connection:new
a=cs-correlation:callerid:+441134960124 \
uuie:74B9027A869D7966A2
Figure 5: SDP Answer with circuit-switched media
When Endoint A receives the Answer, it examines that B is willing to
become the active endpoint when setting up the PSTN call. Endoint A
temporarily stores B's E.164 number and the User-User IE value of the
"cs-correlation" attribute, and waits for a circuit-switched bearer
establishment.
Endpoint B initiates a circuit-switched bearer using whatever
circuit-switched technology is available for it. The called party
number is set to A's number, and calling party number is set to B's
own number. Endpoint B also sets the User-User Information Element
value to the one contained in the SDP Answer.
When Endpoint A receives the circuit-switched bearer establishment,
it examines the UUIE and the calling party number, and by comparing
those received during O/A exchange determines that the call is
related to the SDP session.
It may also be that neither the UUIE nor the calling party number is
received by the called party, or the format of the calling party
number is changed by the PSTN. Implementations may still accept such
call establishment attempts as being related to the session that was
established in the IP network. As it cannot be guaranteed that the
values used for correlation are always passed intact through the
network, they should be treated as additional hints that the circuit-
switched bearer is actually related to the session.
6.2. Advanced SDP example: Circuit-Switched Audio and Video Streams
Endpoint A Endpoint B
| |
| (1) SDP Offer (PSTN audio and video) |
|------------------------------------------->|
| |
| (2) SDP Answer (PSTN audio) |
|<-------------------------------------------|
| |
| PSTN call setup |
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|<-------------------------------------------|
| |
|<======== media over PSTN bearer ==========>|
| |
Figure 6: Circuit-Switched Audio and Video streams
Figure 6 shows an example of negotiating audio and video media
streams over circuit-switched bearers.
v=0
o=alice 2890844526 2890842807 IN IP4 192.0.2.5
s=
t=0 0
a=setup:actpass
a=connection:new
c=PSTN E164 +441134960123
m=audio 9 PSTN -
a=cs-correlation:dtmf:1234536
m=video 9 PSTN 34
a=rtpmap:34 H263/90000
a=cs-correlation:callerid:+441134960123
Figure 7: SDP offer with circuit-switched audio and video (1)
Upon receiving the SDP offer described in Figure 7, Endpoint B
rejects the video stream as the device does not currently support
video, but accepts the circuit-switched audio stream. As Endoint A
indicated that it is able to become either the active or passive
party, Endpoint B gets to select which role it would like to take.
Since the Offer contained the international E.164 number of Endpoint
A, Endpoint B decides that it becomes the active party in setting up
the circuit-switched bearer. B includes a new value in the "dtmf"
subfield of the "cs-correlation" attribute, which it is going to send
as DTMF tones once the bearer setup is complete. The Answer is
described in Figure 8
v=0
o=- 2890973824 2890987289 IN IP4 192.0.2.7
s=
t=0 0
a=setup:active
a=connection:new
c=PSTN E164 +441134960124
m=audio 9 PSTN -
a=cs-correlation:dtmf:654321
m=video 0 PSTN 34
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a=cs-correlation:callerid:+441134960124
Figure 8: SDP answer with circuit-switched audio and video (2)
7. Security Considerations
This document provides an extension on top of RFC 4566 [RFC4566], and
RFC 3264 [RFC3264]. As such, the security considerations of those
documents apply.
This memo provides mechanisms to agree on a correlation identifier or
identifiers that are used to evaluate whether an incoming circuit-
switched bearer is related to an ongoing session in the IP domain.
If an attacker replicates the correlation identifier and establishes
a call within the time window the receiving endpoint is expecting a
call, the attacker may be able to hijack the circuit-switched bearer.
These types of attacks are not specific to the mechanisms presented
in this memo. For example, caller ID spoofing is a well known attack
in the PSTN. Users are advised to use the same caution before
revealing sensitive information as they would on any other phone
call. Furthermore, users are advised that mechanisms that may be in
use in the IP domain for securing the media, like Secure RTP (SRTP)
[RFC3711], are not available in the CS domain.
For the purposes of establishing a circuit-switched bearer, the
active endpoint needs to know the passive endpoint's phone number.
Phone numbers are sensitive information, and some people may choose
not to reveal their phone numbers when calling using supplementary
services like Calling Line Identification Restriction (CLIR) in GSM.
Implementations should take the caller's preferences regarding
calling line identification into account if possible, by restricting
the inclusion of the phone number in SDP "c=" line if the caller has
chosen to use CLIR. If this is not possible, implementations may
present a prompt informing the user that their phone number may be
transmitted to the other party.
Similarly as with IP addresses, if there is a desire to protect the
SDP containing phone numbers carried in SIP, implementers are advised
to follow the security mechanisms defined in [RFC3261].
It is possible that an attacker creates a circuit-switched session
whereby the attacked endpoint should dial a circuit-switched number,
perhaps even a premium-rate telephone number. To mitigate the
consequences of this attack, endpoints MUST authenticate and trust
remote endpoints users who try to remain passive in the circuit-
switched connection establishment. It is RECOMMENDED that endpoints
have local policies precluding the active establishment of circuit
switched connections to certain numbers (e.g., international,
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premium, long distance). Additionally, it is strongly RECOMMENDED
that the end user is asked for consent prior to the endpoint
initiating a circuit-switched connection.
8. IANA Considerations
This document instructs IANA to register a number of SDP tokens
according to the following data.
8.1. Registration of new cs-correlation SDP attribute
Contact: Miguel Garcia <miguel.a.garcia@ericsson.com>
Attribute name: cs-correlation
Long-form attribute name: PSTN Correlation Identifier
Type of attribute: media level only
Subject to charset: No
Description: This attribute provides the Correlation Identifier
used in PSTN signaling
Appropriate values:see Section 5.2.3.1
Specification: RFC XXXX
The IANA is requested to create a subregistry for 'cs-correlation'
attribute under the Session Description Protocol (SDP) Parameters
registry. The initial values for the subregistry are presented in
the following, and IANA is requested to add them into its database:
Value of 'cs-correlation' attribute Reference Description
----------------------------------- --------- -----------
callerid RFC XXXX Caller ID
uuie RFC XXXX User-User
Information Element
dtmf RFC XXXX Dual-tone
Multifrequency
Note for the RFC Editor: 'RFC XXXX' above should be replaced by a
reference to the RFC number of this draft.
As per the terminology in [RFC5226], the registration policy for new
values of 'cs-correlation' parameter is 'Specification Required'.
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8.2. Registration of a new "nettype" value
This memo provides instructions to IANA to register a new "nettype"
in the Session Description Protocol Parameters registry [1]. The
registration data, according to RFC 4566 [RFC4566] follows.
Type SDP Name Reference
---- ------------------ ---------
nettype PSTN [RFCxxxx]
8.3. Registration of new "addrtype" value
This memo provides instructions to IANA to register a new "addrtype"
in the Session Description Protocol Parameters registry [2]. The
registration data, according to RFC 4566 [RFC4566] follows.
Type SDP Name Reference
---- ------------------ ---------
addrtype E164 [RFCxxxx]
Note: RFC XXXX defines the "E164" addrtype in the context of the
"PSTN" nettype only. Please refer to the relevant RFC for a
description of that representation.
8.4. Registration of a new "proto" value
This memo provides instructions to IANA to register a new "proto" in
the Session Description Protocol Parameters registry [3]. The
registration data, according to RFC 4566 [RFC4566] follows.
Type SDP Name Reference
-------------- --------------------------- ---------
proto PSTN [RFCxxxx]
The related "fmt" namespace re-uses the conventions and payload type
number defined for RTP/AVP. In RFC XXXX, the RTP audio and video
media types, when applied to PSTN circuit-switched bearers, represent
merely an audio or video codec in its native format directly on top
of a single PSTN bearer.
In come cases, the endpoint is not able to determine the list of
available codecs for circuit-switched media streams. In this case,
in order to be syntactically compliant with SDP [RFC4566], the
endpoint MUST include a single dash ("-") in the <fmt> subfield.
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9. Acknowledgments
The authors want to thank Paul Kyzivat, Flemming Andreasen, Thomas
Belling, John Elwell, Jari Mutikainen, Miikka Poikselka, Jonathan
Rosenberg, Ingemar Johansson, Christer Holmberg, Alf Heidermark, Tom
Taylor, Thomas Belling, Keith Drage, and Andrew Allen for providing
their insight and comments on this document.
10. References
10.1. Normative References
[ITU.Q931.1998]
, "Digital Subscriber Signalling System No. 1 (DSS 1) -
ISDN User - Network Interface Layer 3 Specification for
Basic Call Control", ISO Standard 9594-1, May 1998.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, June
2002.
[RFC3966] Schulzrinne, H., "The tel URI for Telephone Numbers", RFC
3966, December 2004.
[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
the Session Description Protocol (SDP)", RFC 4145,
September 2005.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4648] Josefsson, S., "The Base16, Base32, and Base64 Data
Encodings", RFC 4648, October 2006.
[RFC5226] Narten, T. and H. Alvestrand, "Guidelines for Writing an
IANA Considerations Section in RFCs", BCP 26, RFC 5226,
May 2008.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234, January 2008.
10.2. Informative References
[I-D.ietf-cuss-sip-uui]
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Johnston, A. and J. Rafferty, "A Mechanism for
Transporting User to User Call Control Information in
SIP", draft-ietf-cuss-sip-uui-10 (work in progress), April
2013.
[ITU.E164.1991]
International Telecommunications Union, "The International
Public Telecommunication Numbering Plan", ITU-T
Recommendation E.164, 1991.
[ITU.Q23.1988]
International Telecommunications Union, "Technical
features of push-button telephone sets ", ITU-T Technical
Recommendation Q.23, 1988.
[RFC3108] Kumar, R. and M. Mostafa, "Conventions for the use of the
Session Description Protocol (SDP) for ATM Bearer
Connections", RFC 3108, May 2001.
[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.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC3725] 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.
[RFC4975] Campbell, B., Mahy, R., and C. Jennings, "The Message
Session Relay Protocol (MSRP)", RFC 4975, September 2007.
[TS.24.008]
3GPP , "Mobile radio interface Layer 3 specification; Core
network protocols; Stage 3 ", 3GPP TS 24.008 3.20.0,
December 2005.
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Authors' Addresses
Miguel A. Garcia-Martin
Ericsson
Calle Via de los Poblados 13
Madrid, ES 28033
Spain
Email: miguel.a.garcia@ericsson.com
Simo Veikkolainen
Nokia
P.O. Box 226
NOKIA GROUP, FI 00045
Finland
Phone: +358 50 486 4463
Email: simo.veikkolainen@nokia.com
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