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Versions: (draft-camarillo-sipping-early-media) 00 01 02 RFC 3960

Internet Engineering Task Force                               SIPPING WG
Internet Draft                                              G. Camarillo
                                                                Ericsson
                                                          H. Schulzrinne
                                                     Columbia University
draft-ietf-sipping-early-media-02.txt
June 1, 2004
Expires: December, 2004


                Early Media and Ringing Tone Generation
                in the Session Initiation Protocol (SIP)

STATUS OF THIS MEMO

   By submitting this Internet-Draft, I certify that any applicable
   patent or other IPR claims of which I am aware have been disclosed,
   and any of which I become aware will be disclosed, in accordance with
   RFC 3668.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that other
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   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
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   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt.

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.


Abstract

   This document describes how to manage early media in SIP using two
   models; the gateway model and the application server model. It also
   describes the inputs one needs to consider to define local policies
   for ringing tone generation.









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                           Table of Contents



   1          Introduction ........................................    3
   2          Session Establishment in SIP ........................    3
   3          The Gateway Model ...................................    4
   3.1        Forking .............................................    5
   3.2        Ringing Tone Generation .............................    6
   3.3        Absence of an Early Media Indicator .................    8
   3.4        Applicability of the Gateway Model ..................    8
   4          The Application Server Model ........................    9
   4.1        In-Band Versus Out-of-Band Session Progress
              Information .........................................   10
   5          Alert-Info Header Field .............................   10
   6          Security Considerations .............................   10
   7          Acknowledgments .....................................   11
   8          Authors' Addresses ..................................   11
   9          Normative References ................................   12
   10         Informative References ..............................   12




























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1 Introduction

   Early media refers to media (e.g., audio and video) that is exchanged
   before a particular session is accepted by the called user. Within a
   dialog, early media occurs from the moment the initial INVITE is sent
   until the UAS generates a final response. It may be unidirectional or
   bidirectional, and can be generated by the caller, the callee, or
   both. Typical examples of early media generated by the callee are
   ringing tone and announcements (e.g., queuing status). Early media
   generated by the caller typically consists of voice commands or DTMF
   tones to drive IVRs.

   The basic SIP specification (RFC 3261 [1]) only supports very simple
   early media mechanisms. These simple mechanisms have a number of
   problems which relate to forking and security, and do not satisfy the
   requirements of most applications. This document goes beyond the
   mechanisms defined in RFC 3261 [1] and describes two models to
   implement early media using SIP: the gateway model and the
   application server model.

   Although both early media models described in this document are
   superior to the one specified in RFC 3261 [1], the gateway model
   still presents a set of issues. In particular, the gateway model does
   not work well with forking. Nevertheless, the gateway model is needed
   because some SIP entities (in particular, some gateways) cannot
   implement the application server model.

   The application server model addresses some of the issues present in
   the gateway model. This model uses the early-session disposition
   type, which is specified in [2].

   The remainder of this document is organized as follows. Section 2
   describes the offer/answer model in absence of early media, and
   Section 3 introduces the gateway model. In this model, the early
   media session is established using the early dialog established by
   the original INVITE. Section 3.1, Section 3.2 and Section 3.4
   describe the limitations of the gateway model and the scenarios where
   it is appropriate to use this model. Section 4 introduces the
   application server model, which, as stated previously, resolves some
   of the issues present in the gateway model. Section 5 discusses the
   interactions between the Alter-Info header field in both early media
   models.

2 Session Establishment in SIP

   Before presenting both early media models, we will briefly summarize
   how session establishment works in SIP. This will let us keep
   separate features that are intrinsic to SIP (e.g., media being played



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   before the 200 (OK) to avoid media clipping) from early media
   operations.

   SIP [1] uses the offer/answer model [3] to negotiate session
   parameters. One of the user agents - the offerer - prepares a session
   description that is called the offer. The other user agent - the
   answerer - responds with another session description called the
   answer. This two-way handshake allows both user agents to agree upon
   the session parameters to be used to exchange media.

   The idea behind the offer/answer model is to decouple the
   offer/answer exchange from the messages used to transport the session
   descriptions. For example, the offer can be sent in an INVITE request
   and the answer can arrive in the 200 (OK) response for that INVITE,
   or, alternatively, the offer can be sent in the 200 (OK) for an empty
   INVITE and the answer be sent in the ACK. When reliable provisional
   responses [4] and UPDATE requests [5] are used, there are many more
   possible ways to exchange offers and answers.

   Media clipping occurs when the user (or the machine generating media)
   believes that the media session is already established but the
   establishment process has not finished yet. The user starts speaking
   (i.e., generating media) and the first few syllables or even the
   first few words are lost.

   When the offer/answer exchange takes place in the 200 (OK) response
   and in the ACK, media clipping is unavoidable. The called user starts
   speaking at the same time as the 200 (OK) is sent, but the UAS cannot
   send any media until the answer from the UAC arrives in the ACK.

   On the other hand, media clipping does not appear in the most common
   offer/answer exchange (an INVITE with an offer and a 200 (OK) with an
   answer). UACs are ready to play incoming media packets as soon as
   they send an offer. They do this because they cannot count on the
   reception of the 200 (OK) to start playing out media for the caller;
   SIP signalling and media packets typically traverse different paths,
   and so, media packets may arrive before the 200 (OK) response.

   Another form of media clipping (not related to early media either)
   occurs in the caller->callee direction. When the callee picks up and
   starts speaking, the UAS sends a 200 (OK) response with an answer and
   the first media packets in parallel. If the first media packets
   arrive to the UAC before the answer, and the caller starts speaking
   as well, the UAC cannot send media until the 2xx response from the
   UAS arrives.

3 The Gateway Model




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   SIP uses the offer/answer model to negotiate session parameters (as
   described in Section 2). An offer/answer exchange that takes place
   before a final response for the INVITE is sent establishes an "early"
   media session. Early media sessions terminate when a final response
   for the INVITE is sent. If the final response is a 2xx, the early
   media session transitions to a regular media session. If the final
   response is a non-2xx final response, the early media session is
   simply terminated.

   Media exchanged within an early media session is, not surprisingly,
   referred to as early media. The gateway model consists of managing
   early media sessions using offer/answer exchanges in reliable
   provisional responses, PRACKs, and UPDATEs.

   The gateway model presents serious limitations in presence of
   forking, as described in Section 3.1. Therefore, its use in only
   acceptable when the UA cannot distinguish between early and regular
   media, as described in Section 3.4. In any other situation (the
   majority of UAs), it is strongly recommended that the application
   server model described in Section 4 is used instead.

3.1 Forking

   In the absence of forking, assuming that the initial INVITE contains
   an offer, the gateway model does not introduce media clipping.
   Following normal SIP procedures, the UAC is ready to play any
   incoming media as soon as it sends the initial offer in the INVITE.
   The UAS sends the answer in a reliable provisional response and can
   send media as soon as there is media to send. Even if the first media
   packets arrive to the UAC before the 1xx response, the UAC will play
   them.

        Note that, in some situations, the UAC does need to receive
        the answer before being able to play any media. UAs in such
        a situation (e.g., QoS, media authorization or media
        encryption is required) use preconditions to avoid media
        clipping.

   On the other hand, if the INVITE forks, the gateway model may
   introduce media clipping. This happens when the UAC receives
   different answers to its offer in several provisional responses from
   different UASs. The UAC has to deal with bandwidth limitations and
   early media session selection.

   If the UAC receives early media from different UASs, it needs to
   present it to the user. If the early media consists of audio, playing
   several audio streams to the user at the same time may be confusing.
   Other media types (e.g., video), on the other hand, can be presented



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   to the user at the same time. The UAC can, for example, build a
   mosaic with the different inputs.

   However, even with media types that can be played at the same time to
   the user, if the UAC has limited bandwidth, it will not be able to
   receive early media from all the different UASs at the same time.
   Therefore, many times, the UAC needs to choose a single early media
   session and "mute" the rest of them sending UPDATE requests.

        It is difficult to decide which early media session carry
        more important information from the caller's perspective.
        In fact, in some scenarios, the UA cannot even correlate
        media packets with their particular SIP early dialog.
        Therefore, UACs typically pick up one early dialog randomly
        and mute the rest.

   If one of the early media sessions that was muted transitions to a
   regular media session (i.e., the UAS sends a 2xx response), media
   clipping is likely to appear. The UAC typically sends an UPDATE with
   a new offer (upon reception of the 200 OK for the INVITE) to unmute
   the media session. The UAS cannot send any media until it receives
   the offer from the UAC. Therefore, if the caller starts speaking
   before the offer from the UAC is received, his words will get lost.

        Having the UAS send the UPDATE to unmute the media session
        (instead of the UAC) does not avoid media clipping in the
        backward direction and it causes possible race conditions.

3.2 Ringing Tone Generation

   In the PSTN, telephone switches typically play ringing tones to the
   caller to indicate that the callee is being alerted. When, where and
   how these ringing tones are generated has been standardized (i.e.,
   the local exchange of the callee generates a standardized ringing
   tone while the callee is being alterted). A standardized approach to
   provide this type of feedback for the user makes sense in a
   homogeneous environment such as the PSTN, where all the terminals
   have a similar user interface.

   This homogeneity is not found among SIP user agents. SIP user agents
   have different capabilities, different user interfaces and may be
   used to establish sessions that do not involve audio at all. Because
   of this, the way a SIP UA provides the user with information about
   the progress of session establishment is a matter of local policy.
   For example, a UA with a GUI may choose to display a message on the
   screen when the callee is being alerted while another UA may choose
   to show a picture of a phone ringing instead. Many SIP UAs choose to
   imitate the user interface of the PSTN phones. They provide a ringing



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   tone to the caller when the callee is being alerted. Such a UAC is
   supposed to generate ringing tones locally for its user as long as no
   early media is received from the UAS. If the UAS generates early
   media (e.g., an announcement or a special ringing tone), the UAC is
   supposed to play it rather than generating the ringing tone locally.

   The problem is that, sometimes, it is not an easy task for a UAC to
   know whether it should generate local ringing or it will be receiving
   early media. A UAS can send early media without using reliable
   provisional responses (very simple UASs do that) or it can send an
   answer in a reliable provisional response without any intention of
   sending early media (this is the case when preconditions are used).
   Therefore, by only looking at the SIP signalling, a UAC cannot be
   sure whether or not there will be early media for a particular
   session. The UAC needs to check if media packets are arriving at a
   given moment.

        An implementation could even choose to look at the contents
        of the media packets, since they could carry only silence
        or comfort noise.

   With this in mind, a UAC should develop its local policy regarding
   local ringing generation. For example, a POTS-like SIP UA could
   implement the following local policy:

        1.   Unless a 180 (Ringing) response is received, never generate
             local ringing.

        2.   If a 180 (Ringing) has been received but there are no
             incoming media packets, generate local ringing.

        3.   If a 180 (Ringing) has been received and there are incoming
             media packets, play them and do not generate local ringing.

        Note that a 180 (Ringing) response means that the callee is
        being alerted, and a UAS should send such a response if the
        callee is being alerted, regardless of the status of the
        early media session.

   At first sight, such a policy may look difficult to implement in
   decomposed UAs (i.e., media gateway controller and media gateway),
   but this policy is the same as the one described in Section 2, which
   must be implemented by any UA. That is, any UA should play incoming
   media packets (and stop local ringing tone generation if it was being
   performed) in order to avoid media clipping, even if the 200 (OK)
   response has not arrived. So, the tools to implement this early media
   policy are available already to any UA that uses SIP.




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   Note that, while it is not desirable to standardize a common local
   policy to be followed by every SIP UA, a particular subset of more or
   less homogeneous SIP UAs could use the same local policy by
   convention. Examples of such subsets of SIP UAs may be "all the
   PSTN/SIP gateways" or "every 3G IMS terminal". However, defining the
   particular common policy that such groups of SIP devices may use is
   outside the scope of this document.

3.3 Absence of an Early Media Indicator

   SIP, as opposed to other signalling protocols, does not provide an
   early media indicator. That is, there is no information about the
   presence or absence of early media in SIP. Such an indicator could be
   potentially used to avoid generation of local ringing tone by the UAC
   when UAS intends to provide in-band ringing tone or some type of
   announcement. However, due to the way SIP works, such an indicator
   would, in the majority of the cases, be of little use.

   One important reason that would limit the benefit of a potential
   early media indicator is the loose coupling between SIP signalling
   and the media path. SIP signalling traverses a different path than
   the media. The media path is typically optimized to reduce the end-
   to-end delay (e.g., minimum number of intermediaries) while the SIP
   signalling path typically traverses a number of proxies providing
   different services for the session. Due to that reason, it is very
   likely that the media packets with early media reach the UAC before
   any SIP message which could contain an early media indicator.

   Nevertheless, sometimes, SIP responses arrive at the UAC before any
   media packet. There are situations when the UAS intends to send early
   media but cannot do it straight away. For example, UAs using ICE [6]
   may need to exchange several STUN messages before being able to
   exchange media. In this situations, an early media indicator would
   keep the UAC from generating local ringing tone during this time.
   However, while the early media is not arriving to the UAC, the user
   would not be aware of the fact that the remote user is being alerted,
   even though a 180 (Ringing) had been received. Therefore, a better
   solution would be to apply local ringing tone until the early media
   packets could be sent from the UAS to the UAC. This solution does not
   require any early media indicator.

        Note that migrations from local ringing tone to early media
        at the UAC happen in the presence of forking as well; one
        UAS sends a 180 (Ringing) response, and later, another UAS
        starts sending early media.

3.4 Applicability of the Gateway Model




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   Section 3 described some of the limitations of the gateway model. It
   produces media clipping in forking scenarios and requires media
   detection to generate local ringing properly. These issues are
   addressed by the application server model, described in Section 4,
   which is the recommended way of generating early media that is not
   continuous with the regular media generated during the session.

   The gateway model is, therefore, acceptable in situations where the
   UA cannot distinguish between early media and regular media. A PSTN
   gateway is an example of this type of situation. The PSTN gateway
   receives media from the PSTN over a circuit, and sends it to the IP
   network. The gateway is not aware of the contents of the media, and
   it does not exactly know when the transition from early to regular
   media takes place. From the PSTN perspective, the circuit is a
   continuous source of media.

4 The Application Server Model

   The application server model consists of having UAS behave as an
   application server to establish early media sessions with the UAC.
   The UAC indicates support for the early-session disposition type
   (defined in [2]) using the early-session option tag. This way, UASs
   know that they can keep offer/answer exchanges for early media
   (early-session disposition type) and for regular media (session
   disposition type) separate.

   Sending early media using a different offer/answer exchange than the
   one used for sending regular media helps avoid media clipping in case
   of forking. The UAC can reject or mute new offers for early media
   without muting the sessions that will carry media when the original
   INVITE is accepted. The UAC can give priority to media received over
   the latter sessions. This way, the application server model
   transitions from early to regular media at the right moment.

   Having a separate offer/answer exchange for early media also helps
   UACs decide whether or not local ringing should be generated. If a
   new early session is established and that early session contains at
   least an audio stream, the UAC can assume that there will be incoming
   early media and it can then avoid generating local ringing.


        An alternative model would consist of adding a new stream
        labeled as "early media" to the original session between
        the UAC and the UAS using an UPDATE, instead of
        establishing a new early session. We have chosen to
        establish a new early session to be coherent with the
        mechanism used by application servers that are NOT co-
        located with the UAS. This way, the UAS uses the same



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        mechanism as any application server in the network to
        interact with the UAC.

4.1 In-Band Versus Out-of-Band Session Progress Information

   Note that, even when the application server model is used, a UA will
   have to choose which early media sessions are muted and which ones
   are rendered to the user. In order to make this choice easier to UAs,
   it is strongly recommended that information that is not essential for
   the session is not transmitted using early media. For instance, UAs
   should not use early media to send special ringing tones. SIP already
   provides a means to inform the remote user about session
   establishment progress which does not cause any of the problems
   associated with early media; the status code and the reason phrase in
   provisional responses.

5 Alert-Info Header Field

   The Alert-Info header field allows specifying an alternative ringing
   content, such as ringing tone, to the UAC. This header field tells
   the UAC which tone should be played in case local ringing is
   generated, but it does not tell the UAC when to generate local
   ringing. A UAC should follow the rules described above for ringing
   tone generation in both models. If, after following those rules, the
   UAC decides to play local ringing, it can then use the Alert-Info
   header field to generate it.

6 Security Considerations

   SIP uses the offer/answer model [3] to establish early sessions in
   both the gateway and the application server models. User Agents (UAs)
   generate a session description, which contains the transport address
   (i.e., IP address plus port) where they want to receive media, and
   send it to their peer in a SIP message. When media packets arrive at
   this transport address, the UA assumes that they come from the
   receiver of the SIP message carrying the session description.
   Nevertheless, attackers may attempt to gain access to the contents of
   the SIP message and send packets to the transport address contained
   in the session description. To prevent this situation, UAs SHOULD
   encrypt their session descriptions (e.g., using S/MIME).

   Still, even if a UA encrypts its session descriptions, an attacker
   may try to guess the transport address used by the UA and send media
   packets to that address. Guessing such a transport address is
   sometimes easier than it may seem because many UAs always pick up the
   same initial media port. To prevent this situation, UAs SHOULD use
   media-level authentication mechanisms (e.g., SRTP [7]). In addition,
   UAs that wish to keep their communications confidential SHOULD use



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   media-level encryption mechanisms (e.g, SRTP [7]).

   Attackers may attempt to make a UA send media to a victim as part of
   a DoS attack. This can be done by sending a session description with
   the victim's transport address to the UA. To prevent this attack, the
   UA SHOULD engage in a handshake with the owner of the transport
   address received in a session descriptions (just verifying
   willingness to receive media) before sending a large amount of data
   to the transport address. This check can be performed by using a
   connection oriented transport protocol, by using STUN [8] in an end-
   to-end fashion, or by the key exchange in SRTP [7].

   In any event, note that the previous security considerations are not
   early media specific, but apply to the usage of the offer/answer
   model in SIP to establish sessions in general.

   Additionally, an early media-specific risk (roughly speaking, an
   equivalent to forms of "toll fraud" in the PSTN) attempts to exploit
   the different charging policies some operators apply to early and to
   regular media. When UAs are allowed to exchange early media for free,
   but are required to pay for regular media sessions, rogue UAs may try
   to establish a bidirectional early media session and never send a 2xx
   response for the INVITE.

   On the other hand, some application servers (e.g., Interactive Voice
   Response systems) use bidirectional early media to obtain information
   from the callers (e.g., the PIN code of a calling card). So, we do
   not recommend that operators disallow bidirectional early media.
   Instead, operators should consider a remedy of charging early media
   exchanges that last too long, or stopping them at the media level
   (according to the operator's policy).

7 Acknowledgments

   Jon Peterson provided useful ideas on the separation between the
   gateway model and the application server model.

   Paul Kyzivat, Christer Holmberg, Bill Marshall, Francois Audet, John
   Hearty, Adam Roach, Eric Burger, Rohan Mahy, and Allison Mankin
   provided useful comments and suggestions.

8 Authors' Addresses

   Gonzalo Camarillo
   Ericsson
   Advanced Signalling Research Lab.
   FIN-02420 Jorvas
   Finland



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   electronic mail:  Gonzalo.Camarillo@ericsson.com

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University 1214 Amsterdam Avenue, MC 0401
   New York, NY 10027
   USA
   electronic mail:  schulzrinne@cs.columbia.edu

9 Normative References

   [1] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J.
   Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: session
   initiation protocol," RFC 3261, Internet Engineering Task Force, June
   2002.

   [2] G. Camarillo, "The early session disposition type for the session
   initiation protocol (SIP)," Internet Draft draft-ietf-sipping-early-
   disposition-01, Internet Engineering Task Force, Jan. 2004.  Work in
   progress.

   [3] J. Rosenberg and H. Schulzrinne, "An offer/answer model with
   session description protocol (SDP)," RFC 3264, Internet Engineering
   Task Force, June 2002.

10 Informative References

   [4] J. Rosenberg and H. Schulzrinne, "Reliability of provisional
   responses in session initiation protocol (SIP)," RFC 3262, Internet
   Engineering Task Force, June 2002.

   [5] J. Rosenberg, "The session initiation protocol (SIP) UPDATE
   method," RFC 3311, Internet Engineering Task Force, Oct. 2002.

   [6] J. Rosenberg, "Interactive connectivity establishment (ICE): a
   methodology for nettwork address translator (NAT) traversal for the
   session initiation protocol (SIP)," internet draft, Internet
   Engineering Task Force, July 2003.  Work in progress.

   [7] M. Baugher, D. McGrew, M. Naslund, E. Carrara, and K. Norrman,
   "The secure real-time transport protocol (SRTP)," RFC 3711, Internet
   Engineering Task Force, Mar 2004.

   [8] J. Rosenberg, J. Weinberger, C. Huitema, and R. Mahy, "STUN -
   simple traversal of user datagram protocol (UDP) through network
   address translators (nats)," RFC 3489, Internet Engineering Task
   Force, Mar. 2003.




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