Internet Engineering Task Force SIP WG
Internet Draft G. Camarillo
Ericsson
H. Schulzrinne
Columbia University
draft-camarillo-sipping-early-media-01.txt
February 10, 2003
Expires: August, 2003
Early Media and Ringback Tone Generation
in the Session Initiation Protocol
STATUS OF THIS MEMO
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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 which inputs need to be taken into consideration to define
local policies for ringback tone generation.
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Table of Contents
1 Introduction ........................................ 3
2 The Gateway Model ................................... 3
2.1 Media Clipping ...................................... 4
2.1.1 Forking ............................................. 5
2.2 Ringback Tone Generation ............................ 6
2.3 Applicability of the Gateway Model .................. 7
3 The Application Server Model ........................ 8
4 Alert-Info Header Field ............................. 8
5 Acknowledgments ..................................... 9
6 Authors' Addresses .................................. 9
7 Bibliography ........................................ 9
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1 Introduction
Early media refers to media (e.g., audio and/or video) that is
exchanged before a particular session is accepted by the called user.
Early media within a particular SIP dialog takes place from the
moment the initial INVITE is sent until the UAS generates a final
response. Early media can be unidirectional or bi-directional and can
be generated by the caller or/and the callee. Typical examples of
early media generated by the callee are ringback tone and
announcements (e.g., queuing status). Early media generated by the
caller typically consist of voice commands or DTMF tones to drive
IVRs.
The basic SIP spec [1] only supports very simple early media. In
order to support fully-featured early media, UAs need to implement
some extensions in addition to the basic SIP spec. This document
describes two models to implement early media and the extensions
needed in each model.
Section 2 introduces the gateway model. In this model, the early
media session is established using the early dialog established by
the original INVITE. Section 2.1, Section 2.2 and Section 2.3
describe the limitation of the gateway model and the scenarios where
it is appropriate to use this model. Section 3 introduces the
application server model, which resolves some of the issues present
in the gateway model. Section 4 discusses the interactions between
the Alter-Info header field in both early media models.
2 The Gateway Model
SIP [1] uses the offer/answer model [2] 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 mechanism 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 [3] and UPDATE requests [4] are used, there are
many more possible ways to exchange offers and answers.
An offer/answer exchange that takes place before a final response for
the INVITE is sent establishes an "early" media session. Early media
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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 reliable provisional responses, PRACKs and
UPDATEs.
2.1 Media Clipping
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.
Media clipping is closely related to the user's expectations. For
example, in the PSTN, there usually isn't media clipping in the
forward direction because callers are used to wait until the callee
answers in order to start speaking. People do not typically start
saying "Hello" while they are hearing a ringback tone.
However, callees in the PSTN are used to pick up the phone and start
speaking right away. That is why, in the PSTN, when the callee picks
up the phone, the media path is already established. It avoids media
clipping in the backward direction.
Unlike in the PSTN, there are some situations involving SIP where the
callee accepts a session invitation (e.g., picks up a SIP phone) but
the media session has not been established yet. This happens, for
instance, when the callee's receives an empty INVITE request and
sends an offer in a 200 (OK) response. The UAS will not be able to
send any media until it receives the answer in the ACK. Everything
the callee says during that round trip time will get lost.
However, the situation described above is a general SIP issue, not
specifically related to early media. Therefore, it falls outside of
the scope of this document. Section 2.1.1 focuses on the scenarios
where the gateway model introduces media clipping.
Another form of media clipping (not related to early media either)
occurs in the caller->callee direction. If the callee picks up and
starts speaking, the UAS will send a 2xx 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 will not be able to send media until the 2xx
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response from the UAS arrives. Section 2.1.1 does not deal with this
situation either, since it is not early media related.
2.1.1 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
starts sending media right away. Even if the first media packets
arrive to the UAS before the 1xx response, the UAS 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.
However, 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 can be confusing.
Other media types (e.g., video), on the other hand, can be presented
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.
Therefore, UACs typically pick up one early media session
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
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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.
2.2 Ringback Tone Generation
In the PSTN, telephone switches typically play ringback tones to the
caller to indicate that the callee is being alerted. When, where and
how these ringback tones are generated has been standardized (i.e.,
the local exchange of the callee generates a standardized ringback
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
ringback tone to the caller when the callee is being alerted. Such a
UAC is supposed to generate ringback 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 ringback tone), the
UAC is supposed to play it rather than generating the ringback tone
locally.
The problem is that, sometimes, it is not an easy task for a UAC to
know whether it should generate local ringback 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.
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With this in mind, a UAC should develop its local policy regarding
local ringback 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 ringback.
2. If a 180 (Ringing) has been received but there are no
incoming media packets, generate local ringback.
3. If a 180 (Ringing) has been received and there are incoming
media packets, play them and do not generate local
ringback.
Note, however, that implementing such a policy in a decomposed
gateway (media gateway controller and media gateway) can be complex.
The media gateway needs to inform the media gateway controller about
the presence of incoming media, and based on that information, the
media gateway controller needs to control the generation of local
ringback in the media gateway. This type of gateway could choose to
generate local ringback upon reception of a 180 (Ringing) response,
and mix it with any incoming media that happens to arrive (if it does
at all).
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.
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.
2.3 Applicability of the Gateway Model
Section 2.1 and Section 2.2 described some of the limitations of the
gateway model. It produces media clipping in forking scenarios and
requires media detection to generate local ringback properly. These
issues are addressed by the application server model, described in
Section 3, which is the recommended way of generating early media
that is not continuous with the regular media that will be generated
during the session.
The gateway model is, therefore, acceptable in situations where the
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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.
3 The Application Server Model
The application server model consists of having the UAS behave as any
other application server in the session [5]. The UAC includes a Join
header field in the initial INVITE. In order to send early media, the
UAS establishes a new dialog by sending a new INVITE to the URI in
the Join header field.
Sending early media using a different dialog than the one used for
sending regular media helps avoid media clipping in case of forking.
The UAC can reject or mute new invitations 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 achieves a smooth
transition from early to regular media.
Having a separate dialog for early media also helps UAs decide
whether or not local ringback should be generated. If a new dialog to
send early media is established, and that dialog contains at least an
audio stream, the UAC can assume that there will be incoming early
media and it can then avoid generating local ringback.
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 session. We have chosen to establish a
new 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 other
application server in the network to interact with the UAC.
4 Alert-Info Header Field
The Alert-Info header field allows specifying an alternative ringback
tone to the UAC. This header field tells the UAC which tone should be
played in case local ringback is generated, but it does not tell the
UAC when to generate local ringback. A UAC should follow the rules
described above for ringback tone generation in both models. If,
after following those rules, the UAC decides to play local ringback,
it can then use the Alert-Info header field to generate it.
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5 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 and Rohan Mahy provided useful comments and
suggestions.
6 Authors' Addresses
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
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
7 Bibliography
[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] J. Rosenberg and H. Schulzrinne, "An offer/answer model with
session description protocol (SDP)," RFC 3264, Internet Engineering
Task Force, June 2002.
[3] J. Rosenberg and H. Schulzrinne, "Reliability of provisional
responses in session initiation protocol (SIP)," RFC 3262, Internet
Engineering Task Force, June 2002.
[4] J. Rosenberg, "The session initiation protocol (SIP) UPDATE
method," RFC 3311, Internet Engineering Task Force, Oct. 2002.
[5] J. Rosenberg, "A framework and requirements for application
interaction in SIP," internet draft, Internet Engineering Task Force,
Nov. 2002. Work in progress.
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