path: root/doc/rfc/rfc3960.txt

Network Working Group                                       G. Camarillo
Request for Comments: 3960                                      Ericsson
Category: Informational                                   H. Schulzrinne
                                                     Columbia University
                                                           December 2004


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

Status of This Memo

   This memo provides information for the Internet community.  It does
   not specify an Internet standard of any kind.  Distribution of this
   memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (2004).

Abstract

   This document describes how to manage early media in the Session
   Initiation Protocol (SIP) using two models: the gateway model and the
   application server model.  It also describes the inputs one needs to
   consider in defining local policies for ringing tone generation.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  2
   2.  Session Establishment in SIP . . . . . . . . . . . . . . . . .  3
   3.  The Gateway Model. . . . . . . . . . . . . . . . . . . . . . .  4
       3.1.  Forking. . . . . . . . . . . . . . . . . . . . . . . . .  4
       3.2.  Ringing Tone Generation. . . . . . . . . . . . . . . . .  5
       3.3.  Absence of an Early Media Indicator. . . . . . . . . . .  7
       3.4.  Applicability of the Gateway Model . . . . . . . . . . .  8
   4.  The Application Server Model . . . . . . . . . . . . . . . . .  8
       4.1.  In-Band Versus Out-of-Band Session Progress Information.  9
   5.  Alert-Info Header Field. . . . . . . . . . . . . . . . . . . .  9
   6.  Security Considerations. . . . . . . . . . . . . . . . . . . .  9
   7.  Acknowledgments. . . . . . . . . . . . . . . . . . . . . . . . 10
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
       8.1.  Normative References . . . . . . . . . . . . . . . . . . 11
       8.2.  Informative References . . . . . . . . . . . . . . . . . 11
       Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 12
       Full Copyright Statement . . . . . . . . . . . . . . . . . . . 13





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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 User Agent Server (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 dual tone multi-frequency (DTMF) tones
   to drive interactive voice response (IVR) systems.

   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 of early
   media implementations 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 the 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.  Sections 3.1, 3.2, and 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 Alert-Info header field in both early media
   models.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", " NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [9].




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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
   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 offer/answer model decouples 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
   can 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 the 200 (OK) is sent, but the UAS
   cannot send any media until the answer from the User Agent Client
   (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, 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-to-callee direction.  When the callee picks up
   and starts speaking, the UAS sends a 200 (OK) response with an
   answer, in parallel with the first media packets.  If the first media



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   packets arrive at the UAC before the answer and the caller starts
   speaking, the UAC cannot send media until the 200 (OK) response from
   the UAS arrives.

3.  The Gateway Model

   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 200 (OK), the
   early media session transitions to a regular media session.  If the
   final response is a non-200 class final response, the early media
   session is simply terminated.

   Not surprisingly, media exchanged within an early media session is
   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 is seriously limited in the presence of forking, as
   described in Section 3.1.  Therefore, its use is only acceptable when
   the User Agent (UA) cannot distinguish between early and regular
   media, as described in Section 3.4.  In any other situation (the
   majority of UAs), use of the application server model described in
   Section 4 is strongly recommended 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 at the UAC before the 1xx response, the UAC will
   play them.

      Note that, in some situations, the UAC needs 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.



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   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.  On the other hand, other media types (e.g., video) can be
   presented to the user at the same time.  For example, the UAC can
   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" those sending UPDATE requests.

      It is difficult to decide which early media sessions 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
      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.  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 for the
   caller, indicating 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 alerted).  It makes sense for
   a standardized approach to provide this type of feedback for the user
   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 Graphical User Interface (GUI) may choose to



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   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 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
   generate the ringing tone locally.

   The problem is that, sometimes, it is not an easy task for a UAC to
   know whether it will be receiving early media or it should generate
   local ringing.  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 ("Plain Old Telephone
   Service")-like SIP User Agent (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



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   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 already available to any UA that uses SIP.

   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 3GPP IMS (Third Generation Partnership
   Project Internet Multimedia System) 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 the generation of local ringing tone by
   the UAC when UAS intends to provide an in-band ringing tone or some
   type of announcement.  However, in the majority of the cases, such an
   indicator would be of little use due to the way SIP works.

   One important reason limiting 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.  Hence, it is very likely that the media
   packets with early media reach the UAC before any SIP message that
   could contain an early media indicator.

   Nevertheless, sometimes SIP responses arrive at the UAC before any
   media packet.  There are situations in which the UAS intends to send
   early media but cannot do it straight away.  For example, UAs using
   Interactive Connectivity Establishment (ICE) [6] may need to exchange
   several Simple Traversals of the UDP Protocol through NAT (STUN)
   messages before being able to exchange media.  In this situation, an
   early media indicator would keep the UAC from generating a local
   ringing tone during this time.  However, while the early media is not
   arriving at the UAC, the user would not be aware that the remote user
   is being alerted, even though a 180 (Ringing) had been received.
   Therefore, a better solution would be to apply a 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.




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      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

   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 the 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) separate from regular media (session
   disposition type).

   Sending early media using a different offer/answer exchange than the
   one used for sending regular media helps avoid media clipping in
   cases 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.





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      An alternative model would include the addition of a new stream,
      with an "early media" label, 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
      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 for
   UAs, it is strongly recommended that information that is not
   essential for the session not be transmitted using early media.  For
   instance, UAs should not use early media to send special ringing
   tones.  The status code and the reason phrase in SIP can already
   inform the remote user about the progress of session establishment,
   without incurring the problems associated with early media.

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).




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   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 such as the Secure Realtime
   Transport Protocol (SRTP)[7].  In addition, UAs that wish to keep
   their communications confidential SHOULD use 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 description (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,
   equivalent to forms of "toll fraud" in the PSTN) attempts to exploit
   the different charging policies some operators apply to early and
   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 200 (OK) 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.



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8.  References

8.1.  Normative References

   [1] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
       Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
       Session Initiation Protocol", RFC 3261, June 2002.

   [2] Camarillo, G., "The Early Session Disposition Type for the
       Session Initiation Protocol (SIP)", RFC 3959, December 2004.

   [3] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
       Session Description Protocol (SDP)", RFC 3264, June 2002.

8.2.  Informative References

   [4] Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional
       Responses in Session Initiation Protocol (SIP)", RFC 3262, June
       2002.

   [5] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
       Method", RFC 3311, October 2002.

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

   [7] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
       Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
       3711, March 2004.

   [8] Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
       "STUN - Simple Traversal of User Datagram Protocol (UDP) Through
       Network Address Translators (NATs)", RFC 3489, March 2003.

   [9] Bradner, S., "Key words for use in RFCs to Indicate Requirement
       Levels", BCP 14, RFC 2119, March 1997.













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Authors' Addresses

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

   EMail:  Gonzalo.Camarillo@ericsson.com


   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University 1214 Amsterdam Avenue, MC 0401
   New York, NY 10027
   USA

   EMail:  schulzrinne@cs.columbia.edu

































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Full Copyright Statement

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Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.







Camarillo & Schulzrinne      Informational                     [Page 13]
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