| MMUSIC WG | M. Garcia-Martin |
| Internet-Draft | Ericsson |
| Intended status: Standards Track | S. Veikkolainen |
| Expires: July 18, 2013 | Nokia |
| January 14, 2013 |
Session Description Protocol (SDP) Extension For Setting Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)
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).
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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 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.
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.
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.
This section presents the general requirements that are specific for the audio or video media streams over circuit-switched bearers.
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.
Consider the example presented in Figure 1. In this example, Alice is located in an environment where she has access to both IP and circuit-switched bearers for communicating with other endpoints. Alice decides that the circuit-switched bearer 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.
Alice Bob | (1) SDP Offer (PSTN audio) | |----------------------------------->| | | | (2) SDP Answer (PSTN audio) | |<-----------------------------------| | | | PSTN call setup | |<-----------------------------------| | | | | |<===== media over PSTN bearer =====>| | |
Figure 1: Example Flow
Bob receives the SDP offer and determines that he is located in an environment where the IP based bearer is not suitable for real-time audio media. However he also has PSTN circuit-switched bearer available for audio. Bob generates an SDP answer containing a description of the audio media stream over a circuit-switched bearer.
During the offer-answer exchange Alice and Bob also agree the direction in which the circuit-switched bearer should be established. In this example, Bob becomes the active party, in other words, he 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 Alice and Bob.
Bob establishes a circuit-switched bearer towards Alice using whatever mechanisms are defined for the network type in question. When receiving the incoming circuit-switched connection attempt, Alice is able to determine that the attempt is related to the session she is just establishing with Bob.
Alice accepts the circuit-switched connection; the circuit-switched bearer setup is completed. Bob and Alice can now use the circuit-switched connection for two-way audio media.
If, for some reason, Bob would like to reject the offered stream, he would set the port number of the specific stream to zero, as specified in RFC3264 [RFC3264]. Also, if Bob does not understand some of the SDP attributes specified in this document, he would ignore them, as specified in RFC4566 [RFC4566].
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.
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.
According to SDP [RFC4566], the connection data line in SDP has the following syntax:
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 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 address type or the E.164 number allocated to it. In these cases a dash ("-") is used to indicate an unknown address type or connection address. This makes the connection data line be according to the SDP syntax.
Please note that these "E164" and "-" address types defined in this memo are exclusively defined to be used in conjunction with the "PSTN" network type in accordance with [RFC4566]. Usage of "E164" or "-" address types 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 <addrtype> and/or <connection-address> MUST NOT be omitted when unknown since this would violate basic syntax of SDP [RFC4566]. In such cases, they MUST be set to a "-".
The following are examples of the extension to the connection data line:
When the <addrtype> is PSTN, the connection address is defined as follows:
When the <addrtype> is "-", the connection address is defined as follows:
According to SDP [RFC4566], the media description line in SDP has the following syntax:
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, which, when applied to PSTN circuit-switched bearers, represent merely an audio or video codec. The endpoint SHOULD only use those payload type whose corresponding codecs is available for PSTN media streams.
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:
Similarly, an example of a media description for circuit-switched video stream is:
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.
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 MAY use any of these mechanisms and MAY use two or more mechanisms simultaneously.
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. Section 5.7> defined the formal syntax.
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.
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:
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.
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.
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.
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:
Note that, for correlation purposes, 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.
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) 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 previously received in the SDP. If the digits match, then the call setup attempt corresponds to that indicated in the SDP.
As an example, an endpoint willing to send DTMF tone sequence "14D*3" would include a "cs-correlation" attribute line as follows:
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.
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.
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.
In order to avoid a situation where both endpoints attempt to initiate a connection simultaneously, the direction in which the 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.
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:
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 uduring the offer/answer negotiation.
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 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.
According to SDP [RFC4566], the origin line in SDP has the following syntax:
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.
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.
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.
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.
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 <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 SHOULD 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.
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 SDP [RFC4566] for details.
When generating the Offer, if the Offerer supports any of the correlation mechanisms defined in this memo, it 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 contains an enumeration of the correlation mechanisms supported by the Offerer, in the format of subfields.
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.
If the Offerer is only able to become the passive party in the circuit-switched bearer setup, it MUST add the "callerid", "uuie" and/or "dtmf" subfields, but MUST NOT specify values for those subfields.
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:
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:
The negotiation of the value of the 'setup' attribute takes place as defined in Section 4.1 of TCP-Based Media Transport in the SDP [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 is not 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'.
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. If the Answerer is not willing to use circuit-switched media for the session, it MUST construct an Answer where the port number for such media stream(s) is set to zero, according to Section 6 of An Offer/Answer Model with the Session Description Protocol (SDP) [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, 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 needs to determine which role it would like to take. If the Offer includes an international E.164 number in the "c=" line, the Answerer SHOULD become the active party. If the Offer does not include an E.164 number, and if the Answerer is aware of its international E.164 number, it MUST become the passive party. If the Offer does not include an E.164 number in the "c=" line and the Answerer is not aware of its E.164 number, 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 any of the "callerid", "uuie" or "dtmf" subfield values.
If the Answerer becomes the passive party, it MUST NOT add values to the "callerid", "uuie" and/or "dtmf" subfields.
After generating and sending the Answer, if the Answerer became the active party, it
If, on the other hand, the Answerer became the passive party, it
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.
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
If the Offerer becomes the passive party, it
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.
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 RFC 4566
connection-field = [%x63 "=" nettype SP addrtype SP
connection-address CRLF]
;nettype and addrtype are defined in RFC 4566
connection-address /= global-number-digits / "-"
; global-number-digits specified in RFC 3966
;subrules for correlation attribute
attribute /= cs-correlation-attr
; attribute defined in RFC 4566
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
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
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.
Alice Bob
| |
| (1) SDP Offer (PSTN audio) |
|--------------------------------->|
| |
| (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. Alice 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, Alice expresses that she 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=jdoe 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)
Bob 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 Bob's international E.164 number. In the "a=setup" line Bob indicates that he is willing to become the active endpoint when establishing the PSTN call, and he also includes the "a=cs-correlation" attribute line containing the values he 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=
t=0 0
m=audio 9 PSTN -
c=PSTN E164 +441134960124
a=setup:active
a=connection:new
a=cs-correlation:callerid:+441134960124 \
uuie:56A390F3D2B7310023
Figure 5: SDP Answer with circuit-switched media
When Alice receives the Answer, she examines that Bob is willing to become the active endpoint when setting up the PSTN call. Alice temporarily stores Bob's E.164 number and the User-User IE value of the "cs-correlation" attribute, and waits for a circuit-switched bearer establishment.
Bob initiates a circuit-switched bearer using whatever circuit-switched technology is available for him. The called party number is set to Alice's number, and calling party number is set to Bob's own number. Bob also sets the User-User Information Element value to the one contained in the SDP Answer.
When Alice receives the circuit-switched bearer establishment, she 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.
Alice Bob
| |
| (1) SDP Offer (PSTN audio and video) |
|------------------------------------------->|
| |
| (2) SDP Answer (PSTN audio) |
|<-------------------------------------------|
| |
| PSTN call setup |
|<-------------------------------------------|
| |
|<======== 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=jdoe 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, Bob rejects the video stream as his device does not currently support video, but accepts the circuit-switched audio stream. As Alice indicated that she is able to become either the active, or passive party, Bob gets to select which role he would like to take. Since the Offer contained the international E.164 number of Alice, Bob decides that he becomes the active party in setting up the circuit-switched bearer. Bob includes a new value in the "dtmf" subfield of the "cs-correlation" attribute, which he 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
a=cs-correlation:callerid:+441134960124
Figure 8: SDP answer with circuit-switched audio and video (2)
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 wellknown 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, 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.
This document instructs IANA to register a number of SDP tokens according to the following data.
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'.
This memo provides instructions to IANA to register a new "nettype" in the Session Description Protocol Parameters registry . The registration data, according to RFC 4566 [RFC4566] follows.
Type SDP Name Reference
---- ------------------ ---------
nettype PSTN [RFCxxxx]
This memo provides instructions to IANA to register two new "addrtype" in the Session Description Protocol Parameters registry . The registration data, according to RFC 4566 [RFC4566] follows.
Type SDP Name Reference
---- ------------------ ---------
addrtype E164 [RFCxxxx]
- [RFCxxxx]
Note: RFC XXXX defines the "E164" and "-" addrtypes in conjunction with the "PSTN" nettype only. Please refer to the relevant RFC for a description of that representation.
Note to IANA: The current "addrtype" sub-registry structure does not capture the fact that a given addrtype is defined in the context of a particular "nettype". The sub-registry structure should be to correct that as part of this registration.
This memo provides instructions to IANA to register a new "proto" in the Session Description Protocol Parameters registry . 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 this document, 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.
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.
| [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. |