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)
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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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