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rfc:rfc5631

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.

Shacham, et al. Informational [Page 9] 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:

Shacham, et al. Informational [Page 14] 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.

Shacham, et al. Informational [Page 15] RFC 5631 SIP Session Mobility October 2009

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.

Shacham, et al. Informational [Page 16] RFC 5631 SIP Session Mobility October 2009

       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

Shacham, et al. Informational [Page 17] 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.

Shacham, et al. Informational [Page 18] RFC 5631 SIP Session Mobility October 2009

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]

Shacham, et al. Informational [Page 19] RFC 5631 SIP Session Mobility October 2009

 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

Shacham, et al. Informational [Page 20] 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

Shacham, et al. Informational [Page 21] 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.

Shacham, et al. Informational [Page 22] 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

Shacham, et al. Informational [Page 23] 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.

Shacham, et al. Informational [Page 24] 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,

Shacham, et al. Informational [Page 25] 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].

Shacham, et al. Informational [Page 26] 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.

Shacham, et al. Informational [Page 28] RFC 5631 SIP Session Mobility October 2009

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

Shacham, et al. Informational [Page 29] RFC 5631 SIP Session Mobility October 2009

 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

Shacham, et al. Informational [Page 30] RFC 5631 SIP Session Mobility October 2009

    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

Shacham, et al. Informational [Page 31] RFC 5631 SIP Session Mobility October 2009

 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.

Shacham, et al. Informational [Page 32] 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.

Shacham, et al. Informational [Page 33] 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.

Shacham, et al. Informational [Page 34] RFC 5631 SIP Session Mobility October 2009

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

Shacham, et al. Informational [Page 35]

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