Session Initiation Protocol (SIP) Session Mobility
RFC 5631
| Document | Type |
RFC - Informational
(October 2009; No errata)
Was draft-shacham-sipping-session-mobility (individual in rai area)
|
|
|---|---|---|---|
| Authors | Ron Shacham , Wolfgang Kellerer , Srisakul Thakolsri , Henning Schulzrinne | ||
| Last updated | 2015-10-14 | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html pdf htmlized bibtex | ||
| Reviews | |||
| Stream | WG state | (None) | |
| Document shepherd | No shepherd assigned | ||
| IESG | IESG state | RFC 5631 (Informational) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Unknown | ||
| Telechat date | |||
| Responsible AD | Jon Peterson | ||
| Send notices to | dean.willis@softarmor.com | ||
Network Working Group R. Shacham
Request for Comments: 5631 H. Schulzrinne
Category: Informational Columbia University
S. Thakolsri
W. Kellerer
DoCoMo Euro-Labs
October 2009
Session Initiation Protocol (SIP) Session Mobility
Abstract
Session mobility is the transfer of media of an ongoing communication
session from one device to another. This document describes the
basic approaches and shows the signaling and media flow examples for
providing this service using the Session Initiation Protocol (SIP).
Service discovery is essential to locate targets for session transfer
and is discussed using the Service Location Protocol (SLP) as an
example. This document is an informational document.
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 and License Notice
Copyright (c) 2009 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the BSD License.
This document may contain material from IETF Documents or IETF
Contributions published or made publicly available before November
10, 2008. The person(s) controlling the copyright in some of this
material may not have granted the IETF Trust the right to allow
modifications of such material outside the IETF Standards Process.
Without obtaining an adequate license from the person(s) controlling
Shacham, et al. Informational [Page 1]
RFC 5631 SIP Session Mobility October 2009
the copyright in such materials, this document may not be modified
outside the IETF Standards Process, and derivative works of it may
not be created outside the IETF Standards Process, except to format
it for publication as an RFC or to translate it into languages other
than English.
Table of Contents
1. Overview ........................................................3
2. Requirements ....................................................4
3. Roles of Entities ...............................................4
4. Device Discovery ................................................5
5. Session Mobility ................................................7
5.1. Options for Session Mobility ...............................7
5.1.1. Transfer and Retrieval ..............................7
5.1.2. Whole and Split Transfer ............................7
5.1.3. Transfer Modes ......................................8
5.1.3.1. Mobile Node Control (MNC) Mode .............8
5.1.3.2. Session Handoff (SH) Mode ..................8
5.1.4. Types of Transferred Media ..........................8
5.2. Addressing of Local Devices ................................9
5.3. Mobile Node Control Mode ..................................10
5.3.1. Transferring a Session to a Single Local Device ....10
5.3.1.1. RTP Media .................................10
5.3.1.2. MSRP Sessions .............................11
5.3.2. Transfer to Multiple Devices .......................13
5.3.3. Retrieval of a Session .............................16
5.4. Session Handoff (SH) mode .................................16
5.4.1. Transferring a Session to a Single Local Device ....16
5.4.2. Retrieval of a Session .............................19
5.4.3. Transfer to Multiple Devices .......................21
5.5. Distributing Sessions for Incoming Call ...................23
5.6. Use of ICE in Session Mobility ............................24
6. Reconciling Device Capability Differences ......................25
6.1. Codec Differences .........................................25
6.2. Display Resolution and Bandwidth Differences ..............27
7. Simultaneous Session Transfer ..................................27
8. Session Termination ............................................28
9. Security Considerations ........................................29
9.1. Authorization for Using Local Devices .....................29
9.2. Maintaining Media Security During Session Mobility ........29
9.2.1. Establishing Secure RTP Using SDP ..................29
9.2.2. Securing Media Using the Transport Layer ...........31
9.3. Flooding Attacks in MNC Mode ..............................31
10. Acknowledgments ...............................................32
11. References ....................................................32
11.1. Normative References .....................................32
11.2. Informative References ...................................33
Shacham, et al. Informational [Page 2]
RFC 5631 SIP Session Mobility October 2009
1. Overview
As mobile devices improve and include more enhanced capabilities for
IP-based multimedia communications, they will remain limited in terms
of bandwidth, display size, and computational power. Stationary IP
multimedia endpoints (including hardware IP phones, videoconferencing
units, embedded devices and software phones) allow more convenience
of use, but are not mobile. Moving active multimedia sessions
between these devices allows mobile and stationary devices to be used
concurrently or interchangeably in mid-session, combining their
advantages into a single "virtual device". An approach to session
mobility based on the Session Initiation Protocol (SIP) [1] was
described first in [20], where two different methods are proposed:
third-party call control (3pcc) [2] and the REFER method [3].
This document expands on this concept, defining a framework for
session mobility that allows a Mobile Node to discover available
devices and to include them in an active session. In particular, the
framework for session mobility presented in this document describes
basic approaches for using existing protocols and shows the signaling
and media flow examples for providing session mobility using SIP. It
is intended as an informational document.
The devices selected as session transfer targets may be either
personal or public. Personal devices are ones used by a single
individual, such as one's PC or phone. Public devices are ones
available for use by a large group of people and include large
conference-room displays. Two capabilities are required to transfer
sessions:
Device Discovery - At all times, a user is aware of the devices
that are available in his local area, along with their
capabilities.
Session Mobility - While in a session with a remote participant,
the user may transfer any subset of the active media sessions to
one or more devices.
This document describes session mobility examples for SIP. It does
not mandate any particular protocol for device discovery. Many
different protocols exist and we discuss the tradeoffs involved in
choosing between them. For our examples, we use the Service Location
Protocol (SLP) [17], primarily because it is the only such protocol
standardized by the IETF.
Shacham, et al. Informational [Page 3]
RFC 5631 SIP Session Mobility October 2009
2. Requirements
This session mobility framework seeks to fulfill the following
requirements:
o REQ 1: Backward Compatibility - We distinguish two kinds of
devices. Enhanced devices support the call flows described in
Section 5 and can perform discovery, while basic devices can do
neither and only have basic SIP capabilities. Devices initiating
session mobility must have enhanced functionality, while all
others can be either basic or enhanced devices. This includes the
transfer destinations, such as the local video camera, as well as
the device being used by the remote participant.
o REQ 2: Flexibility - Differences in device capabilities should be
reconciled. Transfer should be possible to devices that do not
support the codec being used in the session, and even to devices
that do not have a codec in common with the remote participant. A
transfer should also take into account device differences in
display resolution and bandwidth.
o REQ 3: Minimal Disruption - Session transfer should involve
minimal disruption of the media flow and should not appear to the
remote participant as a new call.
3. Roles of Entities
Session mobility involves five types of components: A Correspondent
Node (CN), a Mobile Node (MN), one or more local devices used as
targets for session transfer, an SLP [17] Directory Agent (DA), and,
optionally, a transcoder. The Correspondent Node (CN) is a basic
multimedia endpoint being used by a remote participant and may be
located anywhere. It may be a SIP User Agent (UA), or a Public
Switched Telephone Network (PSTN) phone reachable through a gateway.
The Mobile Node (MN) is a mobile device, containing a SIP UA for
standard SIP call setup, as well as specialized SIP-handling
capabilities for session mobility, and an SLP [17] User Agent (UA)
for discovering local devices. The local devices are located in the
user's local environment for discovery and use in his current
session. They may be mobility-enhanced or basic. Basic devices,
such as IP phones, are SIP-enabled, but have no other special
capabilities. Mobility-enhanced devices have SLP Service Agent
capabilities for advertising their services and session mobility
handling. They also contain an SLP UA, whose purpose will be
explained in the discussion of multi-device systems in Section 5.4.3.
The SLP Directory Agent (DA) keeps track of devices, including their
locations and capabilities. The use of SLP will be described in more
detail in Section 4. SIP-based transcoding services [18] are used,
Shacham, et al. Informational [Page 4]
RFC 5631 SIP Session Mobility October 2009
when necessary, to translate between media streams, as described in
Section 6.
4. Device Discovery
A Mobile Node must be able to discover suitable devices in its
vicinity. This is outside the scope of SIP, and a separate service
location protocol is needed. It is outside the scope of this
document to define any service location protocol. This section
discusses various options, and describes one of them in more detail.
While having a global infrastructure for discovering devices or other
services in any location would be desirable, nothing of this sort is
currently deployed or standardized. However, this document assumes
that such an infrastructure is unnecessary for discovering devices
that are in close proximity, such as in the same room. It is
possible for such devices to be discovered through direct
communication over a short-range wireless protocol such as the
Bluetooth Session Description Protocol (SDP). Two other categories
of service discovery protocols may be used, assuming that devices
that are physically close to each other are also within the same
network and/or part of the same DNS domain. Multicast-based
protocols, such as SLP [17] (in its serverless mode) or Bonjour
(mDNS-SD [30]), may be used as long as the Mobile Node is within the
same subnet as the local devices. When devices are part of the same
DNS domain, server-mode SLP or non-multicast DNS Service Discovery
(DNS-SD) [29] are possible solutions. Such protocols can discover
devices within a larger geographical area, and have the advantage
over the first category in that they allow for the discovery of
devices at different location granularities, such as at the room or
building level, and in locations other than the current one. In
order to discover devices in a specific location, location
attributes, such as room number, must be used in the search, e.g., as
service attributes in SLP or as a domain name in DNS-SD. The mobile
device must ascertain its location in order to perform this search.
We note that many of these techniques could be difficult to implement
in practice. For example, different wireless networks may be
deployed by different organizations, which could make it unrealistic
to have a common DNS setup.
We describe here how SLP is used in server mode in general, then how
it may be used to discover devices based on their location. As
mentioned before, this is only one of many ways to perform service
discovery. SLP identifies services by a "service type", a "service
URL", which can be any URL, and a set of attributes, defined as
name-value pairs. The attributes may be information such as vendor,
supported media codecs, and display resolution. SLP defines three
roles: Service Agents (SAs), which send descriptions of services;
Shacham, et al. Informational [Page 5]
RFC 5631 SIP Session Mobility October 2009
User Agents (UAs), which query for services; and Directory Agents
(DAs), which receive the registrations and queries. An SA registers
a service description to a DA with a service registration (SrvReg)
message that includes its service type, service URL, and attribute-
value set. A UA queries for services by sending a service request
(SrvRqst) message, narrowing the query based on service type and
attribute values. It receives a reply (SrvRply) that contains a list
of URLs of services that match the query. It may then ascertain the
specific attributes of each service using an attribute request
(AttrRqst) message.
This document assumes the following use of SLP for discovering local
devices. Devices have a service type of "sip-device" and a SIP URI
as the service URI. Section 5.2 describes the form of this URI.
Attributes specify device characteristics such as vendor, supported
media codec, display resolution, as well as location coordinates,
such as street address and room number. SAs are co-located with SIP
UAs on session-mobility enhanced devices, while a separate SA is
available to send SrvReg messages on behalf of basic devices, which
do not have integrated SLP SAs.
The Mobile Node includes an SLP UA that discovers available local
devices and displays them to the user, showing, for example, a map of
all devices in a building or a list of devices in a current room.
Once the MN receives its current location in some manner, its SLP UA
issues a SrvRqst message to the DA requesting all SIP devices, using
the location attributes to filter out those that are not in the
current room. A SrvRply message is sent to the mobile device with a
list of SIP URIs for all devices in the room. A separate attribute
request (AttrRqst) is then sent for each URL to get the attributes of
the service. The MN displays for the user the available devices in
the room and their attributes. Figure 1 shows this protocol flow.
Shacham, et al. Informational [Page 6]
RFC 5631 SIP Session Mobility October 2009
Device DA MN
|(1) SrvReg | |
|------------------------->| |
|(2) SrvRply | |
|<-------------------------| |
| | |
| | |
| |(3) SrvRqst |
| |<----------------------|
| |(4) SrvRply URL list |
| |---------------------->|
| |(5) AttrRqst URL1 |
| |<----------------------|
| |(6) AttrRply |
| |---------------------->|
| | ... |
| | |
Figure 1. SLP message flow for the device to register its service
and the MN to discover it, along with its attributes.
5. Session Mobility
5.1. Options for Session Mobility
5.1.1. Transfer and Retrieval
Session mobility involves both transfer and retrieval of an active
session. A transfer means to move the session on the current device
to one or more other devices. A retrieval causes a session currently
on another device to be transferred to the local device. This may
mean returning a session to the device on which it had originally
been before it was transferred to another device. For example, after
discovering a large video monitor, a user transfers the video output
stream to that device; when he walks away, he returns the stream to
his mobile device for continued communication. One may also move a
session to a device that had not previously carried it. For example,
a participant in an audio call on his stationary phone may leave his
office in the middle of the call and transfer the call to the mobile
device as he is running out the door.
5.1.2. Whole and Split Transfer
The set of session media may either be transferred completely to a
single device or split across multiple devices. For instance, a user
may only wish to transfer the video component of his session while
maintaining the audio component on his PDA. Alternatively, he may
find separate video and audio devices and wish to transfer one media
Shacham, et al. Informational [Page 7]
RFC 5631 SIP Session Mobility October 2009
type to each. Furthermore, even the two directions of a full-duplex
session may be split across devices. For example, a PDA's display
may be too small for a good view of the other call participant, so
the user may transfer video output to a projector and continue to use
the PDA camera.
5.1.3. Transfer Modes
Two different modes are possible for session transfer, Mobile Node
Control (MNC) mode and Session Handoff (SH) mode. We describe them
below in turn.
5.1.3.1. Mobile Node Control (MNC) Mode
In Mobile Node Control mode, the Mobile Node uses third-party call
control [2]. It establishes a SIP session with each device used in
the transfer and updates its session with the CN, using the SDP
parameters to establish media sessions between the CN and each
device, which take the place of the current media sessions with the
CN. The shortcoming of this approach is that it requires the MN to
remain active to maintain the sessions.
5.1.3.2. Session Handoff (SH) Mode
A user may need to transfer a session completely because, for
example, the battery on his mobile device is running out or he is
losing radio connectivity. Alternatively, the user of a stationary
device who leaves the area and wishes to transfer the session to his
mobile device will not want the session to remain on the stationary
device when he is away, since it will allow others to easily tamper
with his call. In such a case, Session Handoff mode, which
completely transfers the session signaling and media to another
device, is useful.
Based on our experiments, we have found MNC mode to be more
interoperable with existing devices used on the CN's side. The
remainder of this section describes the transfer, retrieval, and
splitting of sessions in each of the two session transfer modes.
5.1.4. Types of Transferred Media
A communication session may consist of a number of media types, and a
user should be able to transfer any of them to his device of choice.
This document considers audio, video, and messaging. Audio and video
are carried by RTP and negotiated in the SDP body of the SIP requests
and responses. Three different methods exist for carrying text
messages, and possibly other MIME types, that are suitable for SIP
endpoints. RTP may be used to transport text payloads in real time,
Shacham, et al. Informational [Page 8]
RFC 5631 SIP Session Mobility October 2009
based on [9]. Any examples given for audio and video will work
identically for text, as only the payloads differ. For the transfer
of entire messages (as opposed to a small number of characters in
RTP), either the SIP MESSAGE method [21] or the Message Session Relay
Protocol (MSRP) [7] may be used. MESSAGE is used to send individual
page-mode messages. The messages are not associated with a session,
and no negotiation is done to establish a session. Typically, a SIP
UA will allow the user to send MESSAGE requests during an established
dialog, and they are sent to the same contact address as all
signaling messages are sent in mid-session. We discuss later how
these messages are affected by session mobility. MSRP, on the other
hand, is based on sessions that are established like the real-time
media sessions previously described. As such, transferring them is
similar to transferring other media sessions. However, this document
will point out where special handling is necessary for these types of
sessions.
5.2. Addressing of Local Devices
As stated before, this document assumes both personal and public
devices. We assume that public devices use a dedicated Address of
Record (AOR), such as sip:device11@example.com. A personal device
already uses the owner's AOR, so that he should be reachable there;
that AOR could also be used for transferring sessions. However, it
is preferable to distinguish the role of a device as a transfer
target from its existing role. Therefore, all devices are assumed to
have dedicated AORs.
Since every transfer device has its own AOR, there is a one-to-one
mapping between AOR and device. Therefore, a transfer request could
be addressed to the AOR, which would resolve to the device. However,
in Section 5.4.3, we present a model where devices create multi-
device systems to pool their capabilities. Therefore, a single
device must be reachable by multiple URIs representing different
combinations of devices. The appropriate solution is to define each
combination of devices with a Globally Routable UA URI (GRUU) [12].
Therefore, we assume the following addressing for the remainder of
the document. As mentioned earlier, a device has a unique AOR. It
registers a separate contact URI for itself and for each system of
devices that it controls. Each contact has an associated GRUU, which
is registered with SLP as the Service URI, and may be directly
addressed by another device in a request sent through the proxy.
When the proxy forwards the request to the device, it will replace
the GRUU with the contact URI, as described in [12]. Therefore, the
contact URI, not the associated GRUU, will be used by devices to
determine whether the request is for the device itself or for a
multi-device system. We assume that the public GRUU is used.
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RFC 5631 SIP Session Mobility October 2009
5.3. Mobile Node Control Mode
5.3.1. Transferring a Session to a Single Local Device
5.3.1.1. RTP Media
local device MN CN
|(1) INVITE no sdp | |
|<------------------------| |
|(2) 200 OK local params | |
|------------------------>| |
| |(3) INVITE local params |
| |------------------------>|
| RTP | |
|<..................................................|
| |(4) 200 OK CN params |
| |<------------------------|
| |(5) ACK |
| |------------------------>|
|(6) ACK CN params | |
|<------------------------| RTP |
|..................................................>|
| | |
| | |
Figure 2. Mobile Node Control mode flow for transfer to a single
device.
Figure 2 shows the message flow for transferring a session to a
single local device. It follows Third Party Call Control Flow I
(specified in [2]), which is recommended as long as the endpoints
will immediately answer. The MN sends a SIP INVITE request to the
local device used for the transfer, requesting that a new session be
established, but does not include an SDP body. The local device's
response contains an SDP body that includes the address and port it
will use for any media, as well as a list of codecs it supports for
each. The MN updates the session with the CN by sending an INVITE
request (re-INVITE) containing the local device's media parameters in
the SDP body, as follows:
v=0
c=IN IP4 av_device.example.com
m=audio 4400 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
m=video 5400 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
Shacham, et al. Informational [Page 10]
RFC 5631 SIP Session Mobility October 2009
Sending both audio and video media lines will transfer both media
sessions of an existing audio/video call to the local device.
Alternatively, the MN may select a subset of the media available on
the local device, and use the local device's parameters for those
media in the request sent to the CN, while continuing to use its own
parameters for the rest of the media. For example, if it only wishes
to transfer an audio session to a local device that supports audio
and video, it will isolate the appropriate media line for audio from
the response received from the local device and put it in the request
sent to the CN, along with its own video parameters. The CN will
send a response and includes, in its body, the media parameters that
it will use, which may or may not be the same as the ones used in the
existing session. The MN will send an ACK message to the local
device, which includes these parameters in the body. The MN will
establish a session with the local device and maintain its session
with the CN, while the media flow will be established directly
between the CN and the local device. Only the MN, who will be in an
ongoing session with the CN, will later be allowed to retrieve the
media session. Parsing of unknown SDP attributes by the controller
is discussed in [2].
5.3.1.2. MSRP Sessions
In figure 2, the message sequence for transferring an MSRP message
session using MNC mode is identical to that used for audio or video,
although the contents of the messages differ. To simplify the
example, we assume that an MSRP session, with no other media, is
being transferred to a local messaging node, MSGN. In the following
flow, we refer to the corresponding messages in Figure 2. An empty
INVITE request (1) is sent to the local messaging node, MSGN, as
follows:
INVITE sip:msgn@example.com;gr=urn:uuid:jtr5623n SIP/2.0
To: <sip:msgn@example.com;gr=urn:uuid:jtr5623n>
From: <sip:bob@example.com>;tag=786
Call-ID: 893rty@mn.example.com
Content-Type: application/sdp
The messaging node responds with all of its media capabilities,
including MSRP, as follows (2):
SIP/2.0 200 OK
To: <sip:msgn@example.com;gr=urn:uuid:jtr5623n;tag=087js>;tag=087js
From: <sip:bob@example.com>;tag=786
Call-ID: 893rty@mn.example.com
Content-Type: application/sdp
Shacham, et al. Informational [Page 11]
RFC 5631 SIP Session Mobility October 2009
v=0
c=IN IP4 msgn.example.com
m=message 52000 msrp/tcp *
a=accept-types:text/plain
a=path:msrp://msgn.example.com:12000/kjhd37s2s2;tcp
m=audio 4400 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
m=video 5400 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
The same request is then sent by the MN to the CN (3), but containing
the MSRP media and attribute lines with the path given in the MSGN
response above. The CN responds (4) with its own path. The MN
includes this in the ACK that it sends to the MSGN (6).
MSRP sessions are carried over a reliable connection, using TCP or
TLS (Transport Layer Security). Therefore, unlike in the case of
real-time media, this connection must be established. According to
the MSRP specifications, the initiator of a message session, known as
the "offerer", must be the active endpoint, and open the TCP
connection between them. In this transfer scenario, the offerer of
both sessions is the MN, who is on neither end of the desired TCP
connection. As such, neither endpoint will establish the connection.
A negotiation mechanism could be used to assign the role of active
endpoint during session setup. However, while MSRP leaves open this
possibility, it is not currently included in this document due to
complexity. The only other way that such session transfer would be
possible is if both the CN and the local device ordinarily use an
MSRP relay [8], since no direct connection must be established
between them. When each new endpoint receives the INVITE request
from the MN, it will create a TLS connection with one of its
preconfigured relays if such a connection does not yet exist (the CN
will already have one because of its session with the MN) and receive
the path of the relay. In its response to the MN, it will include
the entire path that must be traversed, including its relay, in the
path attribute. For instance, the response from the MSGN will look
as follows:
SIP/2.0 200 OK
To: <sip:msgn@example.com;gr=urn:uuid:jtr5623n;tag=087js>;tag=087js
From: <sip:bob@example.com>;tag=786
Call-ID: 893rty@mn.example.com
Content-Type: application/sdp
Shacham, et al. Informational [Page 12]
RFC 5631 SIP Session Mobility October 2009
v=0
c=IN IP4 msgn.example.com
m=message 52000 msrp/tcp *
a=accept-types:text/plain
a=path:msrp://relayA.example.com:12000/kjhd37s2s2;tcp \
path:msrp://msgn.example.com:12000/kjhd37s2s2;tcp
Since the CN and the local device each establish a TLS connection
with their relay, as they would for any session, and the relays will
establish a connection between them when a subsequent MSRP message is
sent, neither party needs to establish any special connection. The
existing protocol may therefore be used for session transfer.
5.3.2. Transfer to Multiple Devices
In order to split the session across multiple devices, the MN
establishes a new session with each local device through a separate
INVITE request, and updates the existing session with the CN with an
SDP body that combines appropriate media parameters it receives in
their responses. For instance, in order to transfer an audio and
video call to two devices, the MN initiates separate sessions with
each of them, combines the audio media line from one response and the
video media line from the other, and sends them together as the
request to the CN, as follows:
v=0
m=audio 48400 RTP/AVP 0
c= IN IP4 audio_dev.example.com
a=rtpmap:0 PCMU/8000
m=video 58400 RTP/AVP 34
c= IN IP4 video_dev.example.com
a=rtpmap:34 H263/90000
The CN responds with its own parameters for audio and video. The MN
splits them and sends one to each local device in the ACK that
completes each session setup.
Shacham, et al. Informational [Page 13]
RFC 5631 SIP Session Mobility October 2009
video_dev audio_dev MN CN
| |(1) INVITE no sdp | |
| |<-------------------| RTP Audio |
| |(2) 200 params | |
| |------------------->| |
| |(3) INVITE no sdp | |
|<---------------------------------------| |
| |(4) 200 params | |
|--------------------------------------->| |
| | |(5) INVITE a/v params|
| | |---------------------->|
| | RTP Audio | |
| RTP Video |<...........................................|
|<...............................................................|
| | |(6) 200 OK |
| | |<----------------------|
| | |(7) ACK |
| | |---------------------->|
| |(8) ACK CN audio | |
| |<-------------------| RTP Audio |
| |...........................................>|
| |(9) ACK CN video | |
|<---------------------------------------| RTP Video |
|...............................................................>|
| | | |
| | | |
Figure 3. Mobile Node Control mode flow for transfer to multiple
devices.
Splitting a full-duplex media service, such as video, across an input
and an output device, such as a camera and a video display, is a
simple extension of this approach. The signaling is identical to
that of Figure 3, with the audio and video devices replaced by a
video output and a video input device. The SDP, however, is slightly
different. The MN invites the local devices into two different
sessions, but does not include any SDP body. They each respond with
all of their available media. If they only support unidirectional
media, as is the case for a camera or display-only device, they will
include the "sendonly" or "recvonly" attributes. Otherwise, the MN
will have to append the appropriate attribute to each one's media
line before sending the combined SDP body to the CN. That body will
look as follows:
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RFC 5631 SIP Session Mobility October 2009
m=video 50900 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendonly
c=IN IP4 camera.example.com
m=video 50800 RTP/AVP 34
a=rtpmap:34 H263/90000
a=recvonly
c=IN IP4 display.example.com
In updating an SDP session, according to Section 8 of [4], the i-th
media line in the new SDP corresponds to the i-th media line in the
previous SDP. In the above cases, if a media type is added during
the transfer, the media line(s) should follow the existing ones.
When an existing media is transferred to a different device, the
media line should appear in the same place that it did in the
previous SDP, as should the lines for all media that have not been
altered. When a duplex media stream is being split across an input
and output device, the stream corresponding to the input device
should appear in place of the duplex media stream. Since this new
stream is the one that will be received by the CN, including it in
place of the old one ensures that the CN views the new stream as a
replacement of the old one. The media line corresponding to the
output device must appear after all existing media lines. In the
last example, if the SDP had initially contained a video line
followed by an audio line, the updated SDP sent to the CN would look
as follows:
m=video 50900 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendonly
c=IN IP4 camera.example.com
m=audio 45000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 50800 RTP/AVP 34
a=rtpmap:34 H263/90000
a=recvonly
c=IN IP4 display.example.com
During the course of the session, the CN may send a MESSAGE request
to the MN containing text conversation from the remote user. If the
mobile user wishes to have such messages displayed on a device other
than the MN, the request is simply forwarded to that device. The
forwarded message should be composed as though it were any other
message from the MN to the local device, and include the body of the
received message. The local device will send any MESSAGE request to
the MN, who will forward it to the CN.
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5.3.3. Retrieval of a Session
The MN may later retrieve the session by sending an INVITE request to
the CN with its own media parameters, causing the media streams to
return. It then sends a BYE message to each local device to
terminate the session.
5.4. Session Handoff (SH) mode
5.4.1. Transferring a Session to a Single Local Device
Session Handoff mode uses the SIP REFER method [3]. This message is
a request sent by a "referrer" to a "referee", which "refers" it to
another URI, the "refer target", which may be a SIP URI to be
contacted with an INVITE or other request, or a non-SIP URI, such as
a web page. This URI is specified in the Refer-To header. The
Referred-By [5] header is used to give the referrer's identity, which
is sent to the refer target for authorization. Essential headers
from this message may also be encrypted and sent in the message body
as Secure/Multipurpose Internet Mail Extensions (S/MIME) to
authenticate the REFER request. Figure 4 shows the flow for
transferring a session.
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device15 MN CN
|(1) REFER | |
|<----------------------------| |
|(2) 202 Accepted | |
|---------------------------->| |
|(3) INVITE, Replaces | |
|-------------------------------------------------->|
| RTP |
|<..................................................|
|(4) 200 OK | |
|<--------------------------------------------------|
| RTP |
|..................................................>|
|(5) ACK | |
|-------------------------------------------------->|
| |(6) BYE |
| |<--------------------|
| |(7) 200 OK |
| |-------------------->|
|(8) NOTIFY | |
|---------------------------->| |
|(9) 200 OK | |
|<----------------------------| |
| | |
| | |
Figure 4. Session Handoff mode flow for transfer to a single
device.
The MN sends the following REFER request (1) to a local device:
REFER sip:device15@example.com;gr=urn:uuid:qfnb443ccui SIP/2.0
To: <sip:device15@example.com;gr=urn:uuid:qfnb443ccui>
From: <sip:bob@example.com>
Refer-To:<sip:corresp@example.com;gr=urn:uuid:bbb6981;audio;video?
Replaces="1@mn.example.com;
to-tag=bbb;from-tag=aaa">
Referred-By: <sip:bob@example.com>
[S/MIME authentication body]
This message refers the local device to invite the refer target, the
CN, into a session. The "audio" and "video" tokens in the Refer-To
URI are callee capabilities [10]. Here they are used to inform the
referee that it should initiate an audio and video session with the
CN. Also included in the URI is the Replaces header field,
specifying that a Replaces header field should be included with the
specified value in the subsequent INVITE request. The Replaces
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RFC 5631 SIP Session Mobility October 2009
header identifies an existing session that should be replaced by the
new session. Here, the local device will request that the CN
replaces its current session with the MN with the new session.
According to [6], the CN should only accept a request to replace a
session from certain authorized categories of users. One such type
of user is the current participant in the session. The MN may,
therefore, refer the local device to replace its current session with
the CN. However, it provides authentication by encrypting several
headers from the original REFER request in an S/MIME body that it
sends in the REFER. The local device sends this body to the CN.
This keeps a malicious user from indiscriminately replacing another
user's session. Once the local device receives the REFER request, it
sends an INVITE request to the CN, and a normal session setup ensues.
The CN then tears down its session with the MN.
Once the local device has established a session with the CN, it sends
a NOTIFY request to the MN, as specified in [3]. This NOTIFY
contains the To (including tag), From (including tag), and Call-ID
header fields from the established session to allow the MN to
subsequently retrieve the session, as described in Section 5.4.2.
Once a session is transferred, the destination for MESSAGE requests
moves automatically. Since a new session is established between the
CN and the local device, any subsequent MESSAGE requests will be sent
to that device.
The transfer flow described above for media sessions may also be used
to transfer an MSRP session. The local device will initiate an MSRP
session in message (4), along with the other sessions. The REFER
request (1) indicates that an MSRP session should be established
using callee capabilities in the Refer-To header field, as it does
for audio and video. Such a media feature tag, "message" has already
been defined [11]. Once the local device receives the REFER request,
it initiates an MSRP session with the CN. As the initiator, it will
establish a TCP connection in order to carry the session (as
specified in [7]), or will set up the session through its relay if
configured to do so.
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5.4.2. Retrieval of a Session
device15 MN CN
|(1) REFER | |
|<----------------------------| |
|(2) 202 Accepted | |
|---------------------------->| |
|(3) REFER | |
|---------------------------->| |
|(4) 202 Accepted | |
|<----------------------------| |
| |(5) INVITE, Replaces |
| |-------------------->|
| | RTP |
| |<....................|
| |(6) 200 OK |
| |<--------------------|
| | RTP |
| |....................>|
| |(7) ACK |
| |-------------------->|
| (8) BYE | |
|<--------------------------------------------------|
| (9) 200 OK | |
|-------------------------------------------------->|
| | |
| | |
Figure 5. Session Handoff mode flow for session retrieval.
Figure 5 shows the flow for retrieval by the MN of a session
currently on a local device. In order to better motivate the message
flow, we start by describing the final INVITE (5) and work backwards.
In order for a device to retrieve a session in Session Handoff mode,
it must initiate a session with the CN that replaces the CN's
existing session. The following message is sent by the MN to the CN
(5):
INVITE sip:corresp@example.com;gr=urn:uuid:bbb6981 SIP/2.0
To: <sip:corresp@example.com;gr=urn:uuid:bbb6981>
From: <sip:bob@example.com>
Replaces: 1@device15.example.com;to-tag=aaa;from-tag=bbb
Referred-By: <sip:device15@example.com>
[S/MIME authentication body]
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Since the users on the MN and the local device are different
identities, the MN needs to be referred by the local device and
include its URI in the Referred-By header, in addition to including
an S/MIME authentication body from the local device, in order to be
permitted to replace the session. Therefore, the MN must receive a
REFER request from the local device referring it to send this INVITE
request. The user could use the user interface of the local device
to send this REFER message. However, such an interface may not be
available. Also, the user may wish to execute the transfer while
running out of the office with mobile device in hand. In order for
the MN to prompt the REFER from the local device, it sends a "nested
REFER" [5], a REFER request for another REFER. In this case, the
second REFER is sent back to the Mobile Node. That REFER must
specify that the Replaces header be included in the subsequent INVITE
request. The REFER request from the local device to the MN (3) is
composed as follows:
REFER sip:bob@example.com;gr=urn:uuid:ytav223h67gb3 SIP/2.0
To: <sip:bob@example.com;gr=urn:uuid:ytav223h67gb3>
From: <sip:device15@example.com>
Refer-To: <sip:correspondent@example.com;gr=urn:uuid:bbb6981;audio;
video?Replaces="1@device15.example.com;to-tag=aaa;
from-tag=bbb">
Referred-By: <sip:device15@example.com>
[S/MIME authentication body]
A header field is included in the Refer-To URI to specify the value
of the Replaces header in the target INVITE request. In order to
have this message sent to it, the MN must send the following REFER
request (1):
REFER sip:device15@example.com;gr=urn:uuid:qfnb443ccui SIP/2.0
To: <sip:device15@example.com;gr=urn:uuid:qfnb443ccui>
From: <sip:bob@example.com>
Refer-To:<sip:bob@example.com;gr=urn:uuid:ytav223h67gb3;method=REFER
?Refer-To="<sip:correspondent@example.com;gr=urn:uuid:bbb6981;
audio;video?Replaces=%221@device15.example.com;
to-tag=aaa;from-tag=bbb%221>">
The Refer-To header specifies that the MN is the refer target and
that the referral be in the form of a REFER request. The header
field specifies that the REFER request contains a Refer-To header
containing the URI of the CN. That URI, itself, contains the "audio"
and "video" callee capabilities that will tell the MN to initiate an
audio and video call, and a header field specifying that the ultimate
INVITE request contains a Replaces header. If the MN had previously
transferred the session to the local device, it would have received
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RFC 5631 SIP Session Mobility October 2009
these in the NOTIFY sent by the local device following the
establishment of the session. If, on the other hand, the MN is
retrieving a session it had not previously held, as mentioned above
in Section 5.1.1, it gets these parameters by subscribing to the
Dialog Event Package [13] of the local device. Such a subscription
would only be granted, for instance, to the owner of the original
device that carried the session. Even when these parameters are
provided in the Replaces header, the local device does not accept the
REFER request from anybody except the original participant in the
session or the owner of the device. The MN receives the REFER
request from the local device, sends the INVITE request to the CN,
which accepts it, and, once the session is established, terminates
its session with the local device.
5.4.3. Transfer to Multiple Devices
Splitting a session in SH mode requires multiple media sessions to be
established between the CN and local devices, without the MN
controlling the signaling. This could be done by sending multiple
REFER requests to the local devices, referring each to the CN. The
disadvantage of this method is that there is currently no standard
way to associate multiple sessions as part of a single call in SIP.
Therefore, each session between the CN and a local device will be
treated as a separate call. They may occupy different parts of the
user interface, their media may not be available simultaneously, and
they may have to be terminated separately. This certainly does not
fulfill the requirement of seamlessness.
This document describes the use of multi-device systems to overcome
this problem. A local device's SLP UA queries for other devices and
joins with them to create a "virtual device", or a Multi-Device
System (MDS). We refer to the controlling device as the Multi-Device
System Manager (MDSM). In a system that includes at least one
mobility-enhanced device, one of them may act as the MDSM. In a
system consisting entirely of basic devices, either a dedicated host
or another local device from outside of the system acts as MDSM.
When the MDSM subsequently receives a REFER request, it uses third-
party call control to set up media sessions between the CN and each
device in the system. Specifically, it invites each local device
into a separate session, and uses their media parameters (and
possibly its own) in the INVITE request it sends to the CN.
A single device may act as an MDSM for several different groups of
devices, and also act as an ordinary device with only its native
capabilities. There must be a way to address a request to a device
and specify whether it is to the device itself or one of the multi-
device systems it controls. As mentioned above in Section 5.2, a
device registers a separate contact for itself and for each of its
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RFC 5631 SIP Session Mobility October 2009
multi-device systems. For example, the device with AOR
"sip:device11@example.com" and hostname "device11.example.com" will
register a contact "sip:device11@device11.example.com" that
represents its own capabilities. Once it discovers other devices and
creates an MDS, it will register a new contact,
"sip:av1@device11.example.com". It associates a GRUU with each of
these contacts. The device itself and any new system is registered
in SLP using the GRUU. When the proxy receives a request addressed
to a GRUU, it will rewrite it as the contact URI before forwarding
the request to the device. The device will use this unique contact
to determine whether to handle the request natively or with one of
its systems.
Figure 6 shows the transfer of a session to a multi-device system.
The audio device has previously discovered the video device and
created a multi-device system. The REFER request sent to
"sip:device11@example.com;gr=urn:uuid:893eeeyuinm981" prompts the
audio device to invite the video device into a session to ascertain
its SDP, and then to invite the CN into a session using its own SDP
and that of the video device.
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RFC 5631 SIP Session Mobility October 2009
video audio MN CN
| |(1) REFER | |
| |<--------------------| |
| |(2) 202 Trying | |
| (3) INVITE no sdp |-------------------->| |
|<---------------------| | |
| (4) 200 OK SDP | | |
|--------------------->| | |
| |(5) INVITE a/v SDP, Replaces |
| |--------------------------------->|
| | RTP Audio |
| |<.................................|
| | RTP Video |
|<........................................................|
| |(6) 200 OK CN SDP |
| |<---------------------------------|
| | RTP Audio |
| (7) ACK CN Video SDP |.................................>|
|<---------------------| | |
| RTP Video | | |
|........................................................>|
| |(8) ACK | |
| |--------------------------------->|
| | |(9) BYE |
| | |<-----------|
| | |(10) 200 OK |
| | |----------->|
| | | |
| | | |
Figure 6. Session Handoff to a multi-device system.
5.5. Distributing Sessions for Incoming Call
The examples presented above have involved an established session
that a user transfers to one or more devices. Another scenario would
be for an incoming call to be immediately distributed between
multiple devices when the user accepts the call. In such a case, the
initial session would not yet be established when the transfer takes
place.
The transfer could be carried out in either of the transfer modes.
However, complete handoff to a separate device, which is done in
Session Handoff mode, could be achieved through existing means, such
as proxying or redirection. MNC mode would be useful in a case where
the user wishes to automatically include an additional device in a
call. For instance, a user with a desk IP phone and a PC with a
video camera could join the two into a single logical device. The
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RFC 5631 SIP Session Mobility October 2009
SIP UA on the PC would, for any incoming call, send an INVITE request
to the desk phone, setting the display name in the From header field
to "Bob Jones (audio portion)", for instance, so that the user can
identify the caller on the phone. The user could then either accept
or reject, as he would with a call coming directly to the phone. If
he accepts, the PC UA, acting as the controller, would respond to the
caller with its video parameters and the phone's audio parameters in
the SDP body. The final ACK from the Correspondent Node would then
complete the session establishment.
If the desk phone is registered as a contact for the user, it would
also ring in response to the direct call being proxied there, in
addition to the INVITE request sent by the controller, causing
confusion to the user. The use of caller preferences can solve this
problem, as the caller would indicate that the call should
preferentially be proxied to devices with audio and video
capabilities. It is likely that the caller would use caller
preferences in any case, if they were available to him, to avoid the
callee unknowingly picking up the IP phone when he has a video-
capable device available. However, since caller preferences are not
yet widely supported on commercial devices, the callee must ensure
the proper routing of the call. One solution would be for the PC to
register its contact with a higher priority than the one given to the
phone. The Call Processing Language (CPL) [22] (the "proxy" node)
could then be used to specify that forking should be done to the set
of user devices in sequence, rather than in parallel. Since all
calls would first be sent to the PC as long as it were online, it
would redirect any request that included only audio in its SDP.
5.6. Use of ICE in Session Mobility
Interactive Connectivity Establishment (ICE) [27] is a protocol for
Network Address Translator (NAT) traversal that may be used with SIP.
Rather than negotiating addresses and ports used for media sessions
directly in SDP, a list of possible address/ports (candidates) is
exchanged, and the Session Traversal Utilities for NAT (STUN) [28]
protocol is used to check which pairs of candidates may be used. ICE
could be used in the call flows described in this section. In MNC
mode, the candidates would be sent by each local device to the MN,
who would exchange them with the CN. Afterward, each device would
perform checks with the CN to determine an appropriate candidate. In
SH mode, where the local device establishes a session with the CN,
ICE would work no differently than in the standard case.
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RFC 5631 SIP Session Mobility October 2009
6. Reconciling Device Capability Differences
Session mobility sometimes involves the transfer of a session between
devices with different capabilities. For example, the codec being
used in the current session may not be available on the new device.
Furthermore, that device may not support any codec that is supported
by the CN. In addition to codecs, devices may have different
resolutions or bandwidth limitations that must be taken into account
when carrying out a session transfer.
6.1. Codec Differences
Before executing a session transfer, the device checks the
capabilities of the CN and the new device. These may be found
through either the SIP OPTIONS method, used in SIP to query a
device's media capabilities, or may be included as SLP service
attributes. Since the OPTIONS method is standard, it is suggested to
be used to query the CN, while SLP is suggested to be used to get the
media capabilities of local devices, since it is already being used
for them.
If the CN and the local device are found to have a common codec, the
transfer flow will negotiate that this should become the codec used
in the media session. In MNC mode, the MN forwards the response from
the local device to the CN, who will choose a codec it supports from
those available. In Session Handoff mode, the MN sends a REFER
request to the local device and allows it to negotiate a common codec
with the CN during their session establishment. No special behavior
of the MN is required.
If the MN sees that a common codec does not exist, it executes the
transfer through an intermediate transcoding service. Rather than
establishing a direct media session between the CN and the local
device, separate sessions are established between the transcoder and
each of them, with the transcoder translating between the streams.
The Mobile Node discovers available transcoders through some means,
including SLP.
Using transcoding services in SIP is defined in [18] using third-
party call control. In MNC mode, the Mobile Node establishes one
media session between the transcoder and the CN, and one between the
transcoder and the local device. This differs from the normal
transcoding case, where one party establishes a media session between
itself and the transcoder and one between the transcoder and the
other party. The MN starts by sending an INVITE request to the local
device with no body; it receives in the response the list of codecs
that the device can use. It then repeats this for the CN, and
receives its available codecs. It chooses one codec from each side,
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RFC 5631 SIP Session Mobility October 2009
along with the address and port of each device, and combines them in
an INVITE request sent to the transcoder. The transcoder responds
with the ports on which it will accept each stream. The appropriate
port information is sent individually to the CN and the local device.
Once the three sessions have been established, two media sessions
exist, and the transcoder translates between them. This flow is
shown in Figure 7.
AN Transcoder MN CN
(codec A) (codec B)
| |(1) INVITE no sdp | |
|<---------------------------------------| |
| |(2) 200 AN params | |
|--------------------------------------->| |
| | |(3) INVITE no sdp |
| | |---------------------->|
| | |(4) 200 OK CN params |
| | |<----------------------|
| |(5) INVITE AN, CN params | |
| |<---------------------------| |
| |(6) 200 OK TA, TB params | |
| |--------------------------->| |
| |(7) ACK | |
| |<---------------------------| |
| |(8) ACK TA params | |
|<---------------------------------------| |
| RTP | | |
|..........>| RTP | |
| |...................................................>|
| | | (9) ACK TB params |
| | |---------------------->|
| | | RTP |
| RTP |<...................................................|
|<..........| | |
| | | |
Figure 7. Transfer of a session in Mobile Node Control mode
through a transcoder to translate between native codecs
of CN and an audio node AN, where they share no common
codec.
In Session Handoff mode, the local device itself establishes a
session with the CN through the transcoder. After receiving the
REFER request, it uses the OPTIONS method to find the capabilities of
the CN. It will then use a common codec, if available, in the
session setup, or set up the transcoded session using third-party
call control as in [18].
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RFC 5631 SIP Session Mobility October 2009
6.2. Display Resolution and Bandwidth Differences
Other differences in device capabilities, such as display resolution
and bandwidth limitations, are also suggested to be reconciled during
transfer. For example, a mobile device, limited both in its display
size and bandwidth, will likely be receiving the video stream from
the other call participant at a low resolution and frame rate. When
the user transfers his video output to a large-screen display, he may
start viewing much higher-quality video at the higher native
resolution of the display and at a higher frame rate.
Changing the image resolution and frame rate requires no special
handling by the MN. An SDP format is defined [19] for specifying
these and other parameters for the H.263+ codec, for example. The
suitable image formats and corresponding MPIs (Minimum Picture
Interval, related to the frame rate) supported for the given codec
are listed following the media line, in order of preference. For
example, the following lines in SDP would indicate that a device
supports the H.263 codec (value 34) with the image sizes of 16CIF,
4CIF, CIF, and QCIF (with the MPI for each format following the "="):
m=video 60300 RTP/AVP 34
a=fmtp:34 16CIF=8;4CIF=6;CIF=4;QCIF=3
In MNC mode, the response by the local device (Figure 2, message 2)
to the initial INVITE request sent by the MN includes this line in
the SDP body, and the MN then includes it in the INVITE request sent
to the CN (3). In Session Handoff mode, the local device includes
this parameter in the INVITE request sent to the CN (Figure 4,
message 3) after receiving the REFER request. If the local device is
not configured to include the supported image sizes during session
establishment, the information could be made available through SLP.
The MN then includes it in the INVITE request sent to the CN in
Mobile Node Control mode. However, this information is not sent in
Session Handoff mode unless the local device was configured to send
it. In both modes, the MN sends its own resolution and frame rate
preferences in the body of the INVITE request sent to retrieve the
session.
7. Simultaneous Session Transfer
A session transfer may be carried out by one call participant after
the other participant has transferred the session on his side. If
the first transfer was done in MNC mode, a subset of the original
session media is now on local devices. The MN receives either a
re-INVITE from the other participant or an INVITE request from a
local device on the other side. This message carries the new media
parameters of the session. The MN, therefore, must send a re-INVITE
Shacham, et al. Informational [Page 27]
RFC 5631 SIP Session Mobility October 2009
to any local devices with these parameters. It then includes the
parameters returned from these devices in the 200 OK response. If
the first transfer was done in SH mode, the local device will
directly receive the session transfer message from the other party
and will follow the normal procedure for responding to an INVITE
request. If it is controlling other local devices for this session
as part of an MDS, it follows the procedure above, where the first
transfer was done in MNC mode.
It may occur that both participants attempt a transfer at the same
time. In MNC mode, each node initiates a session with a local
device, then sends a re-INVITE to the other node. Section 14.2 in
[1] mandates a 491 response when a re-INVITE is received for a dialog
once another re-INVITE has already been sent. Once both parties
receive this response, they each generate a random timer with
staggered intervals. Once its timer fires, each participant attempts
the re-INVITE again. The first to receive it from the other
participant responds to it with the SDP parameters of its local
device. Both participants then send an ACK request to their local
device containing the new parameters obtained from the other one
during the re-INVITE process.
In SH mode, if both participants attempt a transfer at the same time,
after one node sends a REFER request to the local device, it receives
the INVITE request from the local device on the other end. The
appropriate protocol definition could mandate that a 491 response be
sent in this case, as well. This response would be returned to the
referrer in a NOTIFY indicating the status of the referred session
establishment. The staggered timer solution described above could
work. The MN would cancel the REFER request sent to the local
device, then wait a random amount of time before sending it again.
8. Session Termination
Once a session has been transferred, the user may terminate it by
hanging up the current device, as he would do in a call originating
on that device. This should be true even when the session is using
several local devices. In MNC mode, when the user hangs up the
current device, a BYE request is sent to the controller. The
controller must then send a BYE request to each device used in the
transfer and a BYE request to the CN. An MDSM used for SH mode must
follow the same procedure. In SH mode, the current device has
previously initiated an ordinary session with the CN in response to
the REFER request, and the BYE it sends to the CN on hang-up requires
no special handling.
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9. Security Considerations
As this work is based heavily on the work in [2], [3], and [5], the
security considerations described in those documents apply. We
discuss here the particular issues of authorizing use of local
devices, providing media-level security following transfer, and the
issue of flooding attacks in MNC mode.
9.1. Authorization for Using Local Devices
It is necessary that the use of a local device be limited to
authorized parties. As stated earlier, this document assumes both
personal and public devices, and these have different authorization
policies. A personal device only accepts transfer requests from a
single identity, the device owner. Therefore, the most appropriate
means of access control is to maintain a list of identities
representing the device owner authorized to transfer sessions to the
device. As mentioned before, the device is configured with an AOR
representing its status as a transfer device, in addition to the
user's AOR. Only requests made to the device AOR follow the access
list, while incoming requests to the user's AOR are accepted from
anyone (provided that a white or blacklist or other policy does not
preclude their request from being accepted). The SIP-Identity header
[25] is used to securely identify the initiator of a SIP request.
That specification can be used in our use-cases when the local device
must ensure that the INVITE or REFER request in MNC or SH mode,
respectively, is indeed from the owner of the device.
Public devices accept transfer requests from a large number of
identities. Access lists may be used for this purpose.
Alternatively, since devices are often available to categories of
users, such as "manager" or "faculty member", an appropriate solution
may be to use trait-based authorization [23]. Using this mechanism,
a user may acquire, from a trusted authorization service, an
"assertion" of his user status and permissions. The assertion, or a
reference to it, is included in the request to use the device.
9.2. Maintaining Media Security During Session Mobility
9.2.1. Establishing Secure RTP Using SDP
Confidentiality, message authentication, and replay protection are
necessary in internet protocols, including those used for real-time
multimedia communications. The Secure Real-time Transfer Protocol
(SRTP) [14] provides these for RTP streams. Since SRTP may be used
to carry the media sessions of SIP devices, such as the MN and CN, we
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discuss how to ensure that the session continues to use SRTP
following the transfer to another device. This is also discussed in
less detail in [2].
The establishment of secure RTP communications through SDP is defined
by two documents. The "crypto" attribute [15] is a media-level
attribute whose value includes the desired cryptographic suite and
key parameters used to perform symmetric encryption on the RTP
packets. Since the key information is sent in the SDP body with no
dedicated encryption or integrity protection, a separate protocol
such as S/MIME must be used to protect the signaling messages.
Another document [16] specifies the "key-mgmt" attribute used to
provide parameters for a key management protocol, such as MIKEY.
Using this attribute, the two participants exchange keys encrypted by
a public or shared key, or negotiate a key using the Diffie-Hellman
method.
The use of cryptographic parameters in SDP does not change the
message flows described earlier in this document. For instance, in
MNC mode shown in Figure 2, the response from the local device (2)
will include, in addition to any supported media type, cryptographic
information for each type. This cryptographic information will be a
list of attribute lines describing the crypto suite and key
parameters using either of the two attributes mentioned. These lines
will be sent by the MN to the CN in the subsequent request (3). The
CN will choose a cryptographic method and return its own key
information in the response (4). Maintaining a secure media session
in SH mode requires the local device to negotiate a cryptographic
relationship in the session that it establishes following its receipt
of the REFER request.
It is noted in [2] that establishing media security in third party
call control depends on the cooperation of the controller. In this
document, the Mobile Node (MN) in Mobile Node Control mode (MNC) has
the role of controller in 3pcc, while in the Session Handoff (SH)
mode, MN uses the REFER method instead. The following is an excerpt
from that document:
End-to-end media security is based on the exchange of keying
material within SDP. The proper operation of these mechanisms
with third party call control depends on the controller behaving
properly. So long as it is not attempting to explicitly disable
these mechanisms, the protocols will properly operate between the
participants, resulting in a secure media session that even the
controller cannot eavesdrop or modify. Since third party call
control is based on a model of trust between the users and the
controller, it is reasonable to assume it is operating in a well-
behaved manner. However, there is no cryptographic means that can
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prevent the controller from interfering with the initial exchanges
of keying materials. As a result, it is trivially possibl[e] for
the controller to insert itself as an intermediary on the media
exchange, if it should so desire.
We note here that given the model presented in this document, where
the controller is operated by the same person that uses the local
device, i.e., the MN user, there is even more reason to believe that
the controller will be well-behaved and will not interfere with the
initial transfer of key exchanges.
9.2.2. Securing Media Using the Transport Layer
The exchange of media could alternatively be secured at the transport
layer, using either TLS or Datagram Transport Layer Security (DTLS)
[24]. The one consideration for use of these protocols in session
mobility would be assigning the client and server roles. In SH mode,
it may be assumed that the local device, the referee, would act as
the client, since it is initiating the signaling session with the CN.
However, in MNC mode, these roles would be unclear. The same problem
was mentioned above in establishing a secure connection for an MSRP
session transferred in MNC mode. This problem could be solved
through the use of Connection-Oriented Media (COMEDIA) [26], which
specifies the "setup" SDP attribute to negotiate these roles.
We describe here briefly how this is done. In the MNC exchange shown
in Figure 2, the local device chooses whether to specify a media
session over a secured transport in its response to the MN. If so,
it includes under the media line a "setup" attribute set to either
"active", "passive", or "actpass". This is sent on to the CN.
Assuming it agreed to such a session, it responds with a "setup"
attribute, as per the COMEDIA specifications. This is then sent by
the MN to the local device. If the local device and CN agreed on
their roles, the appropriate session could be established, through
which the media would be transmitted. Before they transmit media
between them, the CN and local device exchange messages to establish
the TLS or DTLS session. This same approach could be used to
establish an SRTP security context over DTLS, as per [31].
9.3. Flooding Attacks in MNC Mode
The MNC call flows in this document, where one device instructs
another device to send an RTP flow to a third one, present the
possibility of a flooding attack. This is a general problem that
relates to any use of 3pcc. In this document, it is only a concern
where the device is public, as described at the beginning of this
section, and a large group of people can transfer media to it, since
there may not be a very strong trust relationship between the device
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owner (e.g., an institution) and the users. Obviously, where a
device is private and only its owner can transfer to it, the concern
does not exist, given the use of the Identity header mentioned
earlier. A possible solution may be the use of ICE [27], since both
sides confirm that they want to receive each other's media.
10. Acknowledgments
We would like to acknowledge the helpful comments made about this
document by the SIP community, in particular Jon Peterson, Joerg Ott,
and Cullen Jennings.
11. References
11.1. 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] Rosenberg, J., Peterson, J., Schulzrinne, H., and G. Camarillo,
"Best Current Practices for Third Party Call Control (3pcc) in
the Session Initiation Protocol (SIP)", BCP 85, RFC 3725, April
2004.
[3] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
[4] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[5] Sparks, R., "The Session Initiation Protocol (SIP) Referred-By
Mechanism", RFC 3892, September 2004.
[6] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
Protocol (SIP) "Replaces" Header", RFC 3891, September 2004.
[7] Campbell, B., Ed., Mahy, R., Ed., and C. Jennings, Ed., "The
Message Session Relay Protocol (MSRP)", RFC 4975, September
2007.
[8] Jennings, C., Mahy, R., and A. Roach, "Relay Extensions for the
Message Sessions Relay Protocol (MSRP)", RFC 4976, September
2007.
[9] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, June 2005.
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RFC 5631 SIP Session Mobility October 2009
[10] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating
User Agent Capabilities in the Session Initiation Protocol
(SIP)", RFC 3840, August 2004.
[11] Camarillo, G., "Internet Assigned Number Authority (IANA)
Registration of the Message Media Feature Tag", RFC 4569, July
2006.
[12] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent URIs (GRUU) in the Session Initiation Protocol (SIP)",
RFC 5627, October 2009.
11.2. Informative References
[13] Rosenberg, J., Schulzrinne, H., and R. Mahy, Ed., "An INVITE-
Initiated Dialog Event Package for the Session Initiation
Protocol (SIP)", RFC 4235, November 2005.
[14] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
3711, March 2004.
[15] Andreasen, F., Baugher, M., and D. Wing, "Session Description
Protocol (SDP) Security Descriptions for Media Streams", RFC
4568, July 2006.
[16] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
Carrara, "Key Management Extensions for Session Description
Protocol (SDP) and Real Time Streaming Protocol (RTSP)", RFC
4567, July 2006.
[17] Guttman, E., Perkins, C., Veizades, J., and M. Day, "Service
Location Protocol, Version 2", RFC 2608, June 1999.
[18] Camarillo, G., Burger, E., Schulzrinne, H., and A. van Wijk,
"Transcoding Services Invocation in the Session Initiation
Protocol (SIP) Using Third Party Call Control (3pcc)", RFC
4117, June 2005.
[19] Ott, J., Bormann, C., Sullivan, G., Wenger, S., and R. Even,
Ed., "RTP Payload Format for ITU-T Rec. H.263 Video", RFC 4629,
January 2007.
[20] Schulzrinne, H. and E. Wedlund, "Application-Layer Mobility
Using SIP", ACM SIGMOBILE Mobile Computing and Communications
Review, Vol. 4, No. 3, July 2000.
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RFC 5631 SIP Session Mobility October 2009
[21] Campbell, B., Ed., Rosenberg, J., Schulzrinne, H., Huitema, C.,
and D. Gurle, "Session Initiation Protocol (SIP) Extension for
Instant Messaging", RFC 3428, December 2002.
[22] Lennox, J., Wu, X., and H. Schulzrinne, "Call Processing
Language (CPL): A Language for User Control of Internet
Telephony Services", RFC 3880, October 2004.
[23] Peterson, J., Polk, J., Sicker, D., and H. Tschofenig, "Trait-
Based Authorization Requirements for the Session Initiation
Protocol (SIP)", RFC 4484, August 2006.
[24] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", RFC 4347, April 2006.
[25] Peterson, J. and C. Jennings, "Enhancements for Authenticated
Identity Management in the Session Initiation Protocol (SIP)",
RFC 4474, August 2006.
[26] Yon, D. and G. Camarillo, "TCP-Based Media Transport in the
Session Description Protocol (SDP)", RFC 4145, September 2005.
[27] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
Protocol for Network Address Translator (NAT) Traversal for
Offer/Answer Protocols", Work in Progress, October 2007.
[28] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session
Traversal Utilities for NAT (STUN)", RFC 5389, October 2008.
[29] Cheshire, S. and M. Krochmal, "DNS-Based Service Discovery",
Work in Progress, September 2008.
[30] Cheshire, S. and M. Krochmal, "Multicast DNS", Work in
Progress, September 2008.
[31] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework for
Establishing an SRTP Security Context using DTLS", Work in
Progress, March 2009.
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Authors' Addresses
Ron Shacham
Columbia University
1214 Amsterdam Avenue, MC 0401
New York, NY 10027
USA
EMail: shacham@cs.columbia.edu
Henning Schulzrinne
Columbia University
1214 Amsterdam Avenue, MC 0401
New York, NY 10027
USA
EMail: hgs@cs.columbia.edu
Srisakul Thakolsri
DoCoMo Communications Laboratories Europe
Landsberger Str. 312
Munich 80687
Germany
EMail: thakolsri@docomolab-euro.com
Wolfgang Kellerer
DoCoMo Communications Laboratories Europe
Landsberger Str. 312
Munich 80687
Germany
EMail: kellerer@docomolab-euro.com
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