SIPPING R. Shacham Internet-Draft H. Schulzrinne Expires: January 7, 2006 Columbia University S. Thakolsri W. Kellerer DoCoMo Eurolabs July 6, 2005 Session Initiation Protocol (SIP) Session Mobility draft-shacham-sipping-session-mobility-01 Status of this Memo By submitting this Internet-Draft, each author represents that any applicable patent or other IPR claims of which he or she is aware have been or will be disclosed, and any of which he or she becomes aware will be disclosed, in accordance with Section 6 of BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt. The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on January 7, 2006. Copyright Notice Copyright (C) The Internet Society (2005). Abstract Session Mobility is the seamless transfer of media of an ongoing communication session from one device to another. This document describes the general methods and specifies the flows for providing this service as part of the Session Initiation Protocol (SIP). The basic steps involved in session mobility which are describe are Shacham, et al. Expires January 7, 2006 [Page 1] Internet-Draft SIP Session Mobility July 2005 service discovery to locate devices to use as transfer targets, for which the Service Location Protocol (SLP) is used, session transfer, and, sometimes, reconciliation of device capability differences. The described session mobility methods include the possibility of transferring any subset of the active media to one or more devices. Table of Contents 1. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Component overview . . . . . . . . . . . . . . . . . . . . . . 4 4. Service Location . . . . . . . . . . . . . . . . . . . . . . . 4 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 . . . . . . . . . . . . . . . . . . . . 7 5.1.4 Types of transfered media . . . . . . . . . . . . . . 8 5.2 Mobile Node Control Mode . . . . . . . . . . . . . . . . . 8 5.2.1 Transfer to a single local device . . . . . . . . . . 9 5.2.2 Transfer to multiple devices . . . . . . . . . . . . . 10 5.2.3 Retrieval of a Session . . . . . . . . . . . . . . . . 13 5.3 Session Handoff (SH) mode . . . . . . . . . . . . . . . . 14 5.3.1 Transfer to a single device . . . . . . . . . . . . . 14 5.3.2 Retrieval of a session . . . . . . . . . . . . . . . . 17 5.3.3 Transfer to multiple devices . . . . . . . . . . . . . 18 5.4 On Incoming Call . . . . . . . . . . . . . . . . . . . . . 20 6. Reconciling Device Capability Differences . . . . . . . . . . 21 6.1 Codec differences . . . . . . . . . . . . . . . . . . . . 21 6.2 Display resolution and bandwidth differences . . . . . . . 24 7. Session Termination . . . . . . . . . . . . . . . . . . . . . 24 8. Performance . . . . . . . . . . . . . . . . . . . . . . . . . 25 8.1 Disruption of Media During Transfer . . . . . . . . . . . 25 8.1.1 Media Streams . . . . . . . . . . . . . . . . . . . . 25 8.1.2 MSRP Sessions . . . . . . . . . . . . . . . . . . . . 26 8.2 Total Transfer Latency . . . . . . . . . . . . . . . . . . 27 9. Security Considerations . . . . . . . . . . . . . . . . . . . 28 9.1 Authorization for using local devices . . . . . . . . . . 28 9.2 Privacy concerns for input devices . . . . . . . . . . . . 28 9.3 Privacy concerns for output devices . . . . . . . . . . . 29 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . 29 11. Change History . . . . . . . . . . . . . . . . . . . . . . . 29 11.1 Changes from draft-shacham-sipping-session-mobility-00 . . 29 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 29 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 32 Intellectual Property and Copyright Statements . . . . . . . . 33 Shacham, et al. Expires January 7, 2006 [Page 2] Internet-Draft SIP Session Mobility July 2005 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 not mobility. The seamless transition between these devices allow them to be used concurrently or interchangeably in mid-session, combining the advantages of both into a single "virtual device." Since the Session Initiation Protocol (SIP) [1] has been chosen by the Third Generation Partnership Project (3GPP) as its standard for session establishment in the Internet Multimedia Subsystem (IMS) [2] and it is being deployed in hardware and software IP multimedia clients, it is natural to specify an architecture to provide this seamless, ubiquitous service using SIP. Such a SIP-based approach is first given in [17], where two different methods are proposed, third- party call control (3PCC) [3] and the REFER method [4]. This document expands on this concept, defining a complete system for session mobility. The aim of this system is to allow a Mobile Node to discover available devices and to include them in an active session. In order to accomplish this, two main components are defined: o Service Location - At all times, a user is aware of the devices which are available in his local area, along with their capabilities. o Session Mobility - While in a session with a remote participant, the user may transfer any subset of the active media services to one or more devices. As described above, this document describes Session Mobility for SIP. Service location, on the other hand, does not depend on any particular protocol. Many different models exist for providing this service, and this document discusses the tradeoffs involved in choosing between them. For our examples, however, we have decided to use the Service Location Protocol (SLP) [5]. 2. Requirements This session mobility framework seeks to fulfill the following requirements: o REQ 1: Interoperability - No special capabilities should be required of the remote participant in the call, as long as he is using a SIP-compliant device or there is a PSTN gateway between them. The device should be capable of handling a transfer by the mobile user utilizing only features specified in Requests For Comments (RFCs) and mature Internet Drafts. Shacham, et al. Expires January 7, 2006 [Page 3] Internet-Draft SIP Session Mobility July 2005 o REQ 2: Backward Compatibility - Both mobility-enhanced and basic devices should be available as destinations for transfer. Mobility-enhanced devices are those that are controlled by software that is enhanced to support special call handling and service discovery. Commercial IP phones and embedded devices are basic since they cannot be enhanced to support such capabilities. o REQ 3: 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 4: Seamlessness - Session transfer should be as seamless as possible. It should involve minimal disruption of the media flow and should not appear to the remote participant as a new call. 3. Component overview Session Mobility involves five basic components: The Correspondent Node (CN), the Mobile Node (MN), the local devices, an SLP 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 UA, or a POTS phone reachable through a PSTN gateway. The Mobile Node (MN) is a mobile device, containing a SIP User Agent (UA) for standard SIP call setup, as well as specialized SIP-handling capabilities for session mobility and an SLP [5] User Agent (UA) for service querying. 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 5.3.3. The SLP Directory Agent (DA) keeps track of devices based on their location and capabilities. SLP will be described in more detail in Section 4. SIP-based transcoding services [7] are used, when necessary, to translate between media streams, as described in Section 6. 4. Service Location A mobile node must be able to discover suitable devices in its vicinity. It is possible for devices in close proximity to be discovered by direct methods such as Bluetooth without the use of centralized servers. On the other hand, many centralized directory- based service discovery protocols exist, such as the Service Location Protocol (SLP). These protocols are general and are not based on Shacham, et al. Expires January 7, 2006 [Page 4] Internet-Draft SIP Session Mobility July 2005 physical locations of services, but they may be easily adapted by adding location attributes to the service description [18]. They have the advantage of allowing discovery of devices at different location granularities, such as at the room or building level, and in a location other than that of the device. They have the disadvantage of requiring mobile devices to discover their location in order to perform such queries. We have chosen to use a general protocol, SLP, for device discovery in this document. Many standard technologies may be used to update the mobile node of its location for use in an SLP query. Indoors, the node can receive its civic coordinates (ie., street address, room number, etc.) either directly or indirectly. With direct methods, the location is sent to the node itself by means such as a Bluetooth beacon. Indirect methods use external location sources such as swipe cards to update the user's location. The mobile node subscribes to the user's location through the mechanism in [11] and receives the location updates in the format standardized in [12]. Outdoors, a mobile device may use direct methods such as the Global Positioning System (GPS) [13] to update it on its geospatial coordinates. Online databases can translate these to civic coordinates. The choice of location technology is beyond the scope of this document. Once the node has obtained its location, it uses SLP to query for available devices in the vicinity that may be used as transfer destinations. 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. For the SIP devices in this framework, we assume a a service type called "sip-device," whose SIP URL, such as sip:audio_rm123@example.com or sip:audio@device1.example.com serves as its service URL. The first URL mentioned is an address of record that is used to connect to the device through the local proxy server, while the second URL includes the name of the host on which the service is provided, "device1.example.com," allowing a point-to-point session to be established with the device. Either of these models may be used, although the proxy model will likely make the device available after an IP address change more quickly than in the point- to-point model, where the DNS entry would take longer to be updated unless dynamic DNS were used. The service description also includes attributes specifying device characteristics (e.g., vendor, supported media codec, display resolution) and location parameters, such as street address and room number. SLP defines Service Agents (SAs) which send descriptions of services using the Service Registration (SrvReg) message, and User Agents (UA) which query for services using the Service Request (SrvRqst) message. SAs are co-located with SIP UAs on mobility-enhanced devices, while a separate SA is available to send SrvReg messages on behalf of basic Shacham, et al. Expires January 7, 2006 [Page 5] Internet-Draft SIP Session Mobility July 2005 devices, which do not have integrated SLP SAs. SLP provides two models for a UA to query for services. In the distributed model, the UA sends requests through multicast and SAs reply directly with the details of the service they provide. In the centralized model, a Directory Agent (DA) is used, to which the SAs register and from which the UAs request services. For our examples, we use the centralized model, though either could be used. The SA registers its service description to the 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 the 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. 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 which are not in the current room. A SrvRply message is sent to the mobile device with a list of SIP URIs for all devices on the floor. A separate Attribute Request (AttrRqst) is then sent for each URL to get the attributes of the service, which include the room where the device is located. The MN displays for the user the available devices in the room, and their attributes. Figure 1 shows this protocol flow. 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. Shacham, et al. Expires January 7, 2006 [Page 6] Internet-Draft SIP Session Mobility July 2005 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. Transfer means to move the session on the current device to one or more other devices. Retrieval means to remotely transfer a session currently on another device to the local device. This may mean to return 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 retrieve a session to a device that had not previously carried it. For example, a participant in an audio call on his IP 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 Session media may either be transferred completely to a single device or be split across multiple devices. For instance, a user may only wish to transfer the video of his session while maintaining the audio on his PDA. Alternatively, he may find separate video and audio devices and wish to transfer one media service 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 mode and Session Handoff mode. 5.1.3.1 Mobile Node Control mode In Mobile Node Control mode, the Mobile Node uses third-party call control [3]. 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 session with the CN. The shortcoming of this approach is that it requires the MN to remain active to maintain the sessions. Shacham, et al. Expires January 7, 2006 [Page 7] Internet-Draft SIP Session Mobility July 2005 5.1.3.2 Session Handoff (SH) mode A user may need to transfer a session completely because the battery on his mobile device is running out. Alternatively, the user of a stationary device who leaves the area and wishes to transfer the session to his mobile device, he 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 case, Session Handoff mode, which completely transfers the session signaling and media to another device, is useful. We have found Mobile Node Control 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 transfered 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 messaging of text, and possibly other MIME types, that are suitable for SIP endpoints. RTP may be used to transport text payloads based on [22]. Any examples given for audio and video will work identically for text, as only the payloads differ. For the transfer of whole messages, either the SIP MESSAGE method [23] or the Message Session Relay Protocol (MSRP) [20] may be used. MESSAGE is used to send individual page-mode messages. The messages are not associated as part of 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. These messages, therefore, are forwarded by the controlling node to the appropriate device. MSRP, on the other hand, is based on sessions that are established like the real-time media sessions previously described. As such, transfering them is similar to transfering other media sessions. However, this document will point out where special handling is necessary for these types of sessions. 5.2 Mobile Node Control Mode Shacham, et al. Expires January 7, 2006 [Page 8] Internet-Draft SIP Session Mobility July 2005 5.2.1 Transfer to a single local device AN MN CN |(1) INVITE CN params | | |<---------------------| RTP | |............................................>| |(2) 200 AN params | | |--------------------->| | | |(3) INVITE AN params | | |--------------------->| | RTP | | |<............................................| | |(4) 200 OK | | |<---------------------| | |(5) ACK | | |--------------------->| |(6) ACK | | |<---------------------| | | | | | | | Figure 2 : Mobile Node Control mode flow for transfer to a single device. Figure 2 shows the message flow for transfering a session to a single device. It follows Third Party Call Control Flow I specified in [3] which is recommended as long as the endpoints will immediately answer. However, the SDP content here differs somewhat from that flow. Namely, message (1) includes the SDP parameters of the CN, currently being used in the session. Although these may change following the INVITE message sent to the CN (3), there is a performance advantage to initially sending the CN parameters. Following the reinvitation of the CN (3), the CN will redirect its media streams to the address and port given for the AN in message (3). If the AN receives an empty SDP body in message (1), it will be unaware of the new sender, and will not play the content of the RTP packets with the new SSRC, until it receives message (6). During this lapse, not only will media not be played on any device, but the media segment sent will be lost completely. The MN sends a SIP INVITE request to the local device used for the transfer, requesting that a new session be established. The local device's response contains an SDP body that includes the address and ports it will use for any media. The MN updates the session with the CN by sending an INVITE message (re-INVITE) containing the local device's media parameters in the SDP body, as follows: Shacham, et al. Expires January 7, 2006 [Page 9] Internet-Draft SIP Session Mobility July 2005 v=0 c= IN IP4 av_device.example.com m=audio 4400 RTP/AVP 0 m=video 5400 RTP/AVP 34 The CN sends 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 sends an ACK message to the local device, which includes these parameters in the body if they have changed. The MN has established separate SIP session with the CN and the local device, but a media flow has been established between the CN and the local device. 5.2.2 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 the media parameters it receives in their responses. For instance, in order to transfer an audio and video call to two devices, it creates an audio session with one device and a video session with another, and combines the SDP bodies from both to reINVITE the CN, as follows: v=0 m=audio 4400 RTP/AVP 0 c= IN IP4 audio_dev.example.com m=video 5400 RTP/AVP 34 c= IN IP4 video_dev.example.com Shacham, et al. Expires January 7, 2006 [Page 10] Internet-Draft SIP Session Mobility July 2005 VN AN MN CN | |(1) INVITE CN params| | | |<-------------------| RTP Audio | | |...........................................>| | |(2) 200 AN params | | | |------------------->| | | |(3) INVITE CN params| | |<---------------------------------------| RTP Video | |...............................................................>| | |(4) 200 VN params | | | |------------------->| | | | |(5) INVITE AN/VN params| | | |---------------------->| | | RTP Audio | | | RTP Video |<...........................................| |<...............................................................| | | |(6) 200 OK | | | |<----------------------| | | |(7) ACK | | | |---------------------->| | |(8) ACK | | | |<-------------------| | | |(9) ACK | | |<---------------------------------------| | | | | | | | | | 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 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 video display and camera into two different unidirectional media sessions, using the "sendonly" and "recvonly" parameters, respectively. Using their responses, which contain the opposite parameter, the MN constructs the following SDP body to re-INVITE the CN: m=video 8900 RTP/AVP 34 a=recvonly c=IN IP4 display.example.com m=video 8800 RTP/AVP 34 a=sendonly Shacham, et al. Expires January 7, 2006 [Page 11] Internet-Draft SIP Session Mobility July 2005 c=IN IP4 camera.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 its response to the MN, who will then send a response to the CN. The message sequence for transfering an MSRP message session using MNC mode is identical to that used for audio or video, in Figure 2, although the contents of the messages differ. To simplify the example, we assume that an MSRP session, with no other media, is being transfered to a local messaging device, MSGN. An empty INVITE request (1) is sent to the local messaging node, MSGN, as follows: INVITE sip:msgn@msgn.example.com SIP/2.0 To: From: ;tag=786 Call-ID: 893rty@mn.example.com Content-Type: application/sdp The messaging node responds with all of its media capabilities, as follows (2): SIP/2.0 200 OK To: ;tag=087js From: ;tag=786 Call-ID: 893rty@mn.example.com Content-Type: application/sdp v=0 c=IN IP4 msgn.example.com m=message 12000 msrp/tcp * a=accept-types:text/plain a=path:msrp://msgn.example.com:12000/kjhd37s2s2;tcp The same request is then sent by the MN to the CN, but containing the path given in the MSGN response above (3). The CN responds with its own path(4). The MN then includes this in the ACK that it sends to the MSGN (6). MSRP sessions are carried over a reliable connection, using TCP or Shacham, et al. Expires January 7, 2006 [Page 12] Internet-Draft SIP Session Mobility July 2005 TLS. Therefore, unlike in the case of real-time media, this connection must be established. According to the current MSRP draft, the initiator of a message session, known as the "offerer", must be the active endpoint, opening the TCP connection between them. In this transfer scenario, the offerer is the MN, who is on neither end of the desired TCP connection. As such, neither end point will establish the connection. Therefore, one of two solutions must be used. A negotiation mechanism may be added to allow the active endpoint role to be assigned during MSRP session setup. The aforementioned draft leaves open the possiblity of negotiating this role, but has left out previously specified methods because of complexity. Alternatively, the CN and the local device may use MSRP relays [21], so that 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), authenticate, and receive the path of the relay. In its response to the MN, it will include the entire path that must be traversed to it, including its relay, in the path attribute. For instance, the response from the MSGN will look as follows: SIP/2.0 200 OK To: ;tag=087js From: ;tag=786 Call-ID: 893rty@mn.example.com Content-Type: application/sdp v=0 c=IN IP4 msgn.example.com m=message 12000 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. 5.2.3 Retrieval of a Session The MN may later retrieve the session by sending a re-INVITE 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. Shacham, et al. Expires January 7, 2006 [Page 13] Internet-Draft SIP Session Mobility July 2005 5.3 Session Handoff (SH) mode 5.3.1 Transfer to a single device Session Handoff mode uses the SIP REFER method [4]. This message is a request sent by a "referer" 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" [14] header is used to give the referer'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 S/MIME to authenticate the REFER request. Figure 4 shows the flow for transferring a session. AN MN CN |(1) REFER | | |<----------------------------| | |(2) 202 Accepted | | |---------------------------->| | |(3) INVITE, Replaces | | |-------------------------------------------------->| | RTP | |<..................................................| |(4) 200 OK | | |<--------------------------------------------------| | RTP | |..................................................>| |(5) ACK | | |-------------------------------------------------->| |(6) NOTIFY | | |---------------------------->| | |(7) 200 OK | | |<----------------------------| | | |(6) BYE | | |-------------------->| | |(7) 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: Shacham, et al. Expires January 7, 2006 [Page 14] Internet-Draft SIP Session Mobility July 2005 REFER sip:av@local_device.example.com SIP/2.0 To: From: Refer-To: Referred-By: [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 following the URI are callee capabilities. Here they are used to inform the referee that it should initiate an audio and video session with the CN. Also included is the "Replaces" header which is to be included in the INVITE request. The "Replaces" header identifies an existing session that should be replaced by the new session. Here, the local device requests that the CN replace its current session with the MN with the new session. According to [15], the CN should only accept a request to replace a session by certain authorized groups 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 must provide 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. Shacham, et al. Expires January 7, 2006 [Page 15] Internet-Draft SIP Session Mobility July 2005 AN MN CN |(1) REFER | | |<----------------------------| | |(2) 202 Accepted | | |---------------------------->| | |(3) REFER | | |---------------------------->| | | |(4) INVITE, Replaces | | |-------------------->| | | RTP | | |<....................| | |(5) 200 OK | | |<--------------------| | | RTP | | |....................>| | |(6) ACK | | |-------------------->| | (7) BYE | | |<--------------------------------------------------| | (8) 200 OK | | |-------------------------------------------------->| | | | | | | Figure 5 : Session Handoff mode flow for session retrieval. Once the local device has established a session with the CN, it sends a NOTIFY request to the MN, as specified in [4]. 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 5.3.2. Once a session is transfered, 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) must indicate 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 field tag, "message" has already been defined [27]. 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 [20] or will set up the session through its relay if Shacham, et al. Expires January 7, 2006 [Page 16] Internet-Draft SIP Session Mobility July 2005 configured to do so. 5.3.2 Retrieval of a session Figure 5 shows the flow for retrieval by the MN of a session currently on a local device. 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 (4): INVITE sip:cn@host1.macrosoft.com SIP/2.0 To: From: Replaces: 1@local_device.example.com;to-tag=aaa;from-tag=bbb Referred-By: [S/MIME authentication body] 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, and the user may also wish to execute the transfer while running out of the office with mobile device in hand. In order to prompt the REFER from the Mobile Node, a "nested REFER," [14] a REFER request for another REFER, is sent. 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 target INVITE request. The REFER request from the local device to the MN (3) is composed as follows: REFER sip:mn@example.com SIP/2.0 To: From: Refer-To: Referred-By: [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 Shacham, et al. Expires January 7, 2006 [Page 17] Internet-Draft SIP Session Mobility July 2005 have this message sent to it, the MN must send the following REFER request (1): REFER sip:av@local_device.example.com SIP/2.0 To: From: Refer-To: The "Refer-To" header specifies the MN as the refer target and that the referral be in the form of a REFER request. The header field specifies that the REFER request should contain a "Refer-To" header containing the URI of the CN. That URI, itself, should contain 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 should contain a "Replaces" header. If the MN had previously transfered the session to the local device, it would have received 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 must get these parameters by subscribing to the Dialog Event Package [26] 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 must not accept the REFER request from anybody except for 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. 5.3.3 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. Shacham, et al. Expires January 7, 2006 [Page 18] Internet-Draft SIP Session Mobility July 2005 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." It then chooses a single SIP URI to address it, such as sip:a_v@dev.example.com, and registers the service in SLP. We refer to the controlling device as the Multi-Device System Manager (MDSM). In a system that includes at least one mobility-enhanced device, it 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 must act as MDSM. Once the MN discovers this system, it may hand off a session by sending a REFER request to the MDSM URI. When the device receives the request sent to this URI, it uses third-party call control to set up media sessions between the CN and each device in the system (including itself). Specifically, it invites each local device into a separate session, and uses their media parameters to invite the CN into a session. Figure 6 shows the transfer of a session to a multi- device system. The Audio Node (AN) has previously discovered the Video Node (VN), and created a multi-device system. The REFER request sent to sip:a_v@an.example.com prompts the Audio Node to invite the Video Node 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 Node. Shacham, et al. Expires January 7, 2006 [Page 19] Internet-Draft SIP Session Mobility July 2005 VN AN MN CN | |(1) REFER | | | |<--------------------| | | |(2) 202 Trying | | | (3) INVITE No SDP |-------------------->| | |<-------------------| | | | (4) 200 OK VN SDP | | | |------------------->| | | | |(5) INVITE AN/VN SDP, Replaces | | |--------------------------------->| | | RTP Audio | | |<.................................| | | RTP Video | |<......................................................| | |(6) 200 OK CN SDP | | |<---------------------------------| | | RTP Audio | | (7) ACK CN SDP |.................................>| |<-------------------| | | | RTP Video | | | |......................................................>| | |(8) ACK | | | |--------------------------------->| | | |(9) BYE | | | |----------->| | | |(10) 200 OK | | | |<-----------| | | | | | | | | Figure 6 : Session handoff to a multi-device system. 5.4 On Incoming Call The examples presented above have involved an established session which 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 redirection. Mobile Node Control 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. Expires January 7, 2006 [Page 20] Internet-Draft SIP Session Mobility July 2005 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 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 caller 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 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 would alone need to ensure the proper routing of the call. The desired behavior could be achieved by not registering the desk phone and having all calls, whether video or audio-only go through the PC. The PC would register two contacts, one representing itself and the phone as a single audio-video device, and another representing only the phone, but using the PC's host address. A third solution would be to register both devices, give a higher priority to the PC UA, and use CPL (the "proxy" node) [29] to specify that routing should be done to the set of user devices in sequence, rather than in parallel. Since all calls would first be proxied to the PC as long as it were online, it would need to redirect any request that included audio in its SDP. 6. Reconciling Device Capability Differences Session mobility sometimes involves the transfer of a session between devices with differing 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 which must be taken into account when carrying out session transfer. 6.1 Codec differences Before executing a session transfer, the device must check 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 Shacham, et al. Expires January 7, 2006 [Page 21] Internet-Draft SIP Session Mobility July 2005 device's media capabilities, or may be included as SLP service attributes. Since the OPTIONS method is standard, it should be used to query the CN, while SLP should 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 should be carried out so that it becomes the codec used in the new media session. In Mobile Node Control Mode, the flow is identical, but the SDP bodies include the desired codec. 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. If the CN and the local device are found to have a common codec, the transfer should be carried out so that it becomes the codec used in the new media session. In Mobile Node Control Mode, the flow is identical, but the SDP bodies include the desired codec. 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. If a common codec does not exist, the MN must execute the transfer through an intermediate transcoding service, which need not be geographically local. 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 may discover available transcoders through SLP. Current efforts [7] use third-party call control for transcoding. The standard case involves a controller, party A, that initiates a media session with party B through a transcoder. The controller invites the transcoder into a session and provides the media parameters of itself and B. The transcoder responds with a "200 OK" that includes its own media parameters, namely the ports on which it will receive each stream. The controller then establishes a separate session with B, in which it gives it the address and port of the transcoder as the destination of its media. It also establishes its own media session with the transcoder. Once both sessions are established, two media streams have been established through the transcoder. Shacham, et al. Expires January 7, 2006 [Page 22] Internet-Draft SIP Session Mobility July 2005 AN Transcoder MN CN (codec A) (codec B) | |(1) INVITE AN, CN params | | | |<---------------------------| | | |(2) 200 A, B params | | | |--------------------------->| | | |(3) INVITE A params | | |<---------------------------------------| | | RTP | | | |..........>| | | | |(4) 200 AN params | | |--------------------------------------->| | | |(5) ACK | | |<---------------------------------------| | | | |(5) INVITE B params | | | |---------------------->| | | | RTP | | |<...................................................| | | |(6) 200 OK CN params | | | |<----------------------| | | |(7) ACK | | | |---------------------->| | |(8) ACK AN, CN 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 device AN, where they share no common codec. In Mobile Node Control mode, the Mobile Node establishes a media session between the transcoder and the CN, and one between the transcoder and the local device. The initial INVITE sent by the MN to the transcoder includes a session description referring to the CN and the local device, rather than including its own parameters as in the standard case. It then establishes a separate session with each of those nodes. Once the three sessions have been established, two media sessions exist, and the transcoder translates between them. This flow is shown in Figure 7. In Session Handoff mode, the local device itself must establish 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 Shacham, et al. Expires January 7, 2006 [Page 23] Internet-Draft SIP Session Mobility July 2005 call control as in [7]. 6.2 Display resolution and bandwidth differences Other differences in device capabilities, such as display resolution and bandwidth limitations, should also be reconciled during tranfer. 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 framerate. 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 framerate. Changing the image resolution and framerate requires no special handling by the MN. An SDP format is defined [8] for specifying these and other parameters for the H.263+ codec. The suitable image formats and corresponding MPIs (Minimum Picture Interval, related to the framerate) 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 Mobile Node Control mode, the response by the local device (Figure 2, message 1) to the initial INVITE request sent by the MN would include this line in the SDP body, and the MN would then include it in the INVITE request sent to the CN (3). In Session Handoff mode, the local device would include this parameter in the INVITE request sent to the CN (Figure 4, 3) after receiving the REFER request. If the local device is not mobility-enhanced, and is, therefore, not configured to include the supported image sizes during session establishment, the information could be made available through SLP. The MN would then include it in the INVITE request sent to the CN in mobile node control mode. However, this information would not be sent in Session Handoff mode unless the local device were configured to send it. In both modes, the MN would send its own resolution and framerate preferences in the body of the INVITE request sent to retrieve the session. 7. 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 Shacham, et al. Expires January 7, 2006 [Page 24] Internet-Draft SIP Session Mobility July 2005 current device, a BYE 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. A MDSM used for SH mode must follow the same procedure. In SH mode, the current device has previously inititated an ordinary session with the CN in response the the REFER request, and the BYE it sends to the CN on hang-up requires no special handling. 8. Performance In order to enjoy the advantages of session mobility, the transfer should minimize the disruption of service, and should be quick. Below, we analyze the expected performance of our system, based on these criteria. 8.1 Disruption of Media During Transfer The most critical disruption is the "dead time" between tear-down of the existing media stream and the establishment of the new stream. Transfer could cause a segment of media to be unplayed, which we refer to as "lapse." This occurs when one side begins sending media to a device not ready to display it, because the necessary session establishment has not been done. It may also occur if the device involved in the current session stops sending media before the new device begins sending media. Even if there is no lapse, there may be a delay during which one or both sides do not receive media on any device. The flows already shown ensure that this delay is no longer than the time it takes a single packet to travel between the remote participant and the local network, and that the lapse is either non- existent or negligibly small. Therefore, the mobile user need not warn the participant at the Correspondent Node: "Wait for me until I complete this transfer." 8.1.1 Media Streams In the Mobile Node Control mode flow of Figures 2 and 3, even though the CN receives a new destination address for its media in messages 3 and 5, respectively, it will not stop receiving or playing the incoming flows from the MN until it starts receiving media streams from the local devices, initiated (Figure 2 message 1, Figure 3 messages 1 and 3). While the MN has no way of knowing when this happens, it can wait a safe amount of time, such as a second, before it stops sending. Therefore, the CN will not experience any disruption of media flow. The mobile user may experience a negligibly short delay in incoming media service. The CN stops sending media to the MN and starts sending to the local devices after receiving the INVITE request with new media parameters, as shown in the figure. Since the local devices are already listening for the Shacham, et al. Expires January 7, 2006 [Page 25] Internet-Draft SIP Session Mobility July 2005 media following the INVITE requests received by the MN, they will begin displaying the media as soon as they receive it. Therefore, no media lapse will occur. The only delay will be the time it takes for the RTP packets to travel to the devices. During some of that time, as well, old media packets will still be received by the MN. The delay will, therefore, be extremely short. In Session Handoff mode (Fig. 5), also, there will be no media lapse. After the CN receives the INVITE request (3) from the local device, it immediately redirects the media from the MN to the local device. Once the media stream reaches the local device, it will immediately begin displaying it. Therefore, all media sent by the CN will be displayed on some device for the mobile user. There may be a negligibly short delay between the time the MN stops receiving the media and the local device begins displaying it. The CN will not experience any lapse or delay, since it will display the media from the MN until it starts receiving the streams from the local device, following message 4. 8.1.2 MSRP Sessions While latency in messaging sessions is not a serious issue, given their timescale as compared to that of real-time media, lapse is very serious, as it amounts to the omission of an entire message that may fully contain essential information. Therefore, we show that this will not occur. However, it should be mentioned that, unlike real- time media, messaging sessions do not send any packets unless a participant explicitly orders them sent. Even if it were possible for a packet to be lost during the window of time that the transfer occurs, this will only be an issue if one of the participants actually sends a message during that window. As described above, MNC mode operates in a similar way when transfering message sessions. However, as described in Section 5.2.2, MSRP requires the establishment of a reliable connection between the two parties, which are, in our case, the CN and the local device. This connection may be established by one of the sides, such as the CN, given a negotiation mechanism. Alternatively, the local device and CN may both use relays, leaving the relays to handle the connections between them. Given each model, we analyze the possibility of messages sent by the CN being missed by the mobile user and messages sent by the mobile user being missed by the CN. This is done for MNC mode, but the same analysis may be used for SH mode. If the user at the CN indicates that a message should be sent once the transfer starts, it may be sent as part of the existing MSRP session or as part of the new session. Therefore, it will either be Shacham, et al. Expires January 7, 2006 [Page 26] Internet-Draft SIP Session Mobility July 2005 sent through the existing TCP connection to the MN or it will be sent through the new TCP connection established with the local device, after being buffered if necessary. In the first case, the MN will receive the message, as TCP's [24] graceful close will not allow the MN to close the connection until then. In the second case, the local device will receive the message after the TCP connection has been established. The mobile user may attempt to send a message from the MN after the transfer has begun. If the TCP connection has already been torn down, the user interface should disallow such a message by the user, as it has nowhere to send it. If the connection is still open the MN may send the message, and the graceful close will ensure that it will be received by the CN. When the CN wishes to close the connection, it will send a FIN segment, but will continue receiving packets from the MN until it receives the FIN/ACK segment. The MN will only send the FIN/ACK once it has received positive acknowledgments for any messages it has sent. If the user attempts to send a message on the local device before the new TCP connection has been established, the message will be temporarily buffered and sent once the connection is made. If relays are used, once the CN receives the INVITE request from the MN, it will use the same connection to its relay to authenticate and create a new MSRP session. There will be two sessions, one between the CN and the MN, and one between the CN and the local device. Since all connections will remain open, the CN will be able to receive messages associated with either session. Once the CN receives the INVITE request from the MN, it should start sending messages to the local device, in case the MN quickly shuts down. 8.2 Total Transfer Latency A less critical, but important, concern is the total delay in transferring a session, from the time that the mobile user makes the request until he begins to receive the CN's media streams on the new device. Firstly, the user may be away from the current device when transfering to a second one. For example, the mobile user may be retrieving a session onto his mobile device while walking out of the office. Even if the user has both devices in front of him, the latency should be small enough to provide seamless use of the devices. Furthermore, a long delay may lead the mobile user to believe that the transfer is not working, as mentioned regarding ordinary telephone call setup in [19]. Therefore, we give an estimate of the transfer latency in a typical network. Both models presented in this document use flows that consist of signaling between the local environment (MN and local devices) and Shacham, et al. Expires January 7, 2006 [Page 27] Internet-Draft SIP Session Mobility July 2005 the CN, which may traverse a long distance, and signaling within the local environment. Previous work [19] has measured transcontinental call setup delay in SIP to be below one second. We assume that the network-layer packet loss and latency over the wireless links connecting the MN, and possibly the CN and local devices, will not significantly affect this figure. Therefore, we use this figure for call setup between the local environment and the CN. Delay in session creation with the local devices depends on the route taken. If the sessions are established through the proxy servers and the mobile user's proxy server is far away, the setup may require a triangular route to be traversed. If, on the other hand, either the mobile user's proxy is local or peer-to-peer setup is done, without the proxy, setup time should be negligible. We assume that this setup time is very short. In Mobile Node Control mode, the call flow consists of session creation with each local device, followed by a session update with the CN, which has the same signaling as a normal call setup. We therefore estimate that the transfer delay should not be much longer than a second. In Session Handoff mode, as well, the call flow consists of a single call setup between the local network and the CN, and signaling between the MN and the local devices, such as the REFER request. Here, too, the transfer should not take much longer than a second. 9. Security Considerations Many security concerns must be addressed in the local device environment. Here we give ways to handle two such concerns, the unauthorized use of devices and the use of input devices for surveillance [18]. 9.1 Authorization for using local devices Public devices generally have a group of users who are authorized to use and to transfer their sessions to them. It is essential that any other users are not allowed access, in order not to limit the usage of authorized users. Many methods may be used for authentication of users, including Authentication, Authorization and Accounting (AAA) in SIP [16]. 9.2 Privacy concerns for input devices Input devices such as cameras and microphones could be used for surveillance. This concern can be mitigated in two ways. First of all, the remote control of devices could be disallowed by requiring proof that the user is actually in the location of the device. This could be done by requiring an authentication token that is only Shacham, et al. Expires January 7, 2006 [Page 28] Internet-Draft SIP Session Mobility July 2005 available locally [18]. Such a token would regularly change and would be passed to the mobile device by a low-power Bluetooth beacon, with its ray restricted to a single room. This token would then be used in Digest Authentication. In order to keep a local user from transferring an ongoing session, leaving the room and eavesdropping, the device should also contain an LED to warn other users that a session is currently active. 9.3 Privacy concerns for output devices The ability of a call participant to transfer parts of a session to output devices threatens the privacy of the other participant. Especially in the case of video and messaging, the remote participant will be unaware that his stream is being sent to a different device, where it may be viewed by people besides the other participant. Extensions to SIP Caller Preferences [28] and SDP [9] are specified in [30] which allow a call participant to specify the required level of privacy of the session. This may serve to disallow the other party to transfer the session to a less private device, and even force the other party to retieve a session when the content becomes more private. 10. IANA Considerations This document has no actions for IANA. 11. Change History 11.1 Changes from draft-shacham-sipping-session-mobility-00 o Discussed in 5.3.1 how MN receives To, From tags and Call-ID of transfered session, and in 5.3.2 how it receives these if it had not previously held the session. o Defined in subsection 5.1.4 the different media that may be transfered, introducing the three types of messaging o Specified in 5.2.2 and 5.3.2 how transfer flows apply to messaging using SIP MESSAGE and MSRP o Added subsection 5.4 on incoming call o Added section 7 on hangup behavior o Divided up subsection 8.1 to discuss disruption for transfer of continuous media (8.1.1) and MSRP sessions (8.1.2) o Added Section 9.3 about privacy concerns for output devices 12. References [1] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Sparks, R., Handley, A., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002. Shacham, et al. Expires January 7, 2006 [Page 29] Internet-Draft SIP Session Mobility July 2005 [2] "3GPP: TS 23.228: IP Multimedia Subsystem (IMS) (Stage 2), Release 5", September 2002. [3] Rosenberg, J., Peterson, J., Schulzrinne, H., and G. Camarillo, "Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)", RFC 3725, April 2004. [4] Sparks, R., "The Session Initiation Protocol (SIP) Refer Method", RFC 3515, April 2003. [5] Gutman, E., Perkins, C., Veizades, J., and M. Day, "Service Location Protocol, Version 2", RFC 2608, June 1999. [6] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", RFC 3550, July 2003. [7] Camarillo, G., Burger, E., Schulzrinne, H., and A. Van Wijk, "Transcoding Services Invocation in the Session Initiation Protocol Using Third Party Call Control (3pcc)", IETF Internet Draft (Work in Progress), September 2004. [8] Ott, J., Sullivan, G., Wenger, S., and R. Even, "RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+), IETF Internet Draft (Work in progress)", December 2004. [9] Handley, M. and V. Jacobson, "SDP: Session Description Protocol", RFC 2327, April 1998. [10] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with the Session Description Protocol (SDP)", RFC 3264, June 2002. [11] Roach, A., "Session Initiation Protocol (SIP)-Specific Event Notification", RFC 3265, June 2002. [12] Peterson, J., "A Presence-based GEOPRIV Location Object Format, IETF Internet Draft (Work in Progress)", September 2004. [13] "Global Positioning System Standard Positioning Service Specification, 2nd Edition", June 1995. [14] Sparks, R., "The Session Initiation Protocol (SIP) Referred-By Mechanism", RFC 3892, September 2004. [15] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation Protocol (SIP) 'Replaces' Header", RFC 3891, September 2004. [16] Loughney, J. and G. Camarillo, "Authentication, Authorization Shacham, et al. Expires January 7, 2006 [Page 30] Internet-Draft SIP Session Mobility July 2005 and Accounting Requirements for the Session Initiation Protocol (SIP)", RFC 3702, February 2004. [17] Schulzrinne, H. and E. Wedlund, "Application-Layer Mobility Using SIP, ACM Mobile Computing and Communications Review, Vol.4, No.3", July 2000. [18] Berger, S., Schulzrinne, H., Sidiroglou, S., and X. Wu, "Ubiquitous Computing Using SIP, ACM NOSSDAV", 2003. [19] Eyers, T. and H. Schulzrinne, "Predicting Internet Telephony Call Setup Delay, in Proceedings of First IP Telephony Workshop, Berlin, Germany", April 2000. [20] Campbell, B., Mahy, R., and C. Jennings, "The Message Session Relay Protocol", IETF Internet Draft (Work in Progress), February 2005. [21] Jennings, C. and R. Mahy, "Relay Extensions for the Message Session Relay Protocol", IETF Internet Draft (Work in Progress), June 2005. [22] Hellstrom, G., "RTP Payload for Text Conversation", RFC 2793, May 2000. [23] Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C., and D. Gurle, "Session Initiation Protocol (SIP) Extension for Instant Messaging", RFC 3428, December 2002. [24] Postel, J., "Transmission Control Protocol", RFC 793, May 1981. [25] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)", RFC 3840, August 2004. [26] Rosenberg, J., Schulzrinne, H., and R. Mahy, "An INVITE Initiated Dialog Event Package for the Session Initiation Protocol (SIP)", IETF Internet Draft (Work in progress), April 2005. [27] Camarillo, G., "Internet Assigned Number Authority (IANA) Registration of the Message Media Feature Tag", IETF Internet Draft (Work in progress) Session Initiation Protocol (SIP), June 2005. [28] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller Preferences for the Session Initiation Protocol (SIP)", RFC 3841, August 2004. Shacham, et al. Expires January 7, 2006 [Page 31] Internet-Draft SIP Session Mobility July 2005 [29] Lennox, J., Wu, X., and H. Schulzrinne, "Caller Preferences for the Session Initiation Protocol (SIP)", RFC 3880, October 2004. [30] Shacham, R., Schulzrinne, H., Kellerer, W., and S. Thakolsri, "Specifying Media Privacy Requirements in the Session Initiation Protocol (SIP)", IETF Internet Draft (Work in progress), June 2005. Authors' Addresses Ron Shacham Columbia University 1214 Amsterdam Avenue, MC 0401 New York, NY 10027 USA Email: rs2194@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 Eurolabs Landsberger Str. 312 Munich 80687 Germany Email: thakolsri@docomolab-euro.com Wolfgang Kellerer DoCoMo Eurolabs Landsberger Str. 312 Munich 80687 Germany Email: kellerer@docomolab-euro.com Shacham, et al. Expires January 7, 2006 [Page 32] Internet-Draft SIP Session Mobility July 2005 Intellectual Property Statement The IETF takes no position regarding the validity or scope of any Intellectual Property Rights or other rights that might be claimed to pertain to the implementation or use of the technology described in this document or the extent to which any license under such rights might or might not be available; nor does it represent that it has made any independent effort to identify any such rights. Information on the procedures with respect to rights in RFC documents can be found in BCP 78 and BCP 79. Copies of IPR disclosures made to the IETF Secretariat and any assurances of licenses to be made available, or the result of an attempt made to obtain a general license or permission for the use of such proprietary rights by implementers or users of this specification can be obtained from the IETF on-line IPR repository at http://www.ietf.org/ipr. 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Acknowledgment Funding for the RFC Editor function is currently provided by the Internet Society. Shacham, et al. Expires January 7, 2006 [Page 33]