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<rfc category="info" docName="draft-niccolini-sipping-siphandover-05" ipr="full3978">

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you can add the attributes updates="NNNN" and obsoletes="NNNN" 
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  <!-- ***** FRONT MATTER ***** -->

  <front>

<title abbrev="SIP-based vertical handover">Requirements for vertical handover of multimedia sessions using SIP
</title>

<author initials="S." surname="Niccolini" fullname="Saverio Niccolini">
<organization abbrev="NEC">Network Laboratories, NEC Europe Ltd.</organization>
<address>
<postal>
   <street>Kurfuersten-Anlage 36</street>
   <city>Heidelberg</city>
   <code>69115</code>
   <country>Germany</country>
</postal>
<phone>+49 (0) 6221 43 42 118</phone>
<email>saverio.niccolini@netlab.nec.de</email>
<uri>http://www.netlab.nec.de</uri>
</address>
</author>

<author initials="S." surname="Salsano" fullname="Stefano Salsano">
<organization abbrev="Univ. of Rome Tor Vergata">DIE, University of Rome "TorVergata"</organization>
<address>
<postal>
   <street>Via Politecnico, 1</street>
   <city>Rome</city>
   <code>00156</code>
   <country>Italy</country>
</postal>
<phone>+39 06 7259 7770</phone>
<email>stefano.salsano@uniroma2.it</email>
<uri>http://netgroup.uniroma2.it/Stefano_Salsano</uri>
</address>
</author>

<author initials="H." surname="Izumikawa" fullname="Haruki Izumikawa">
<organization abbrev="KDDI Labs">KDDI Labs</organization>
<address>
<postal>
   <street> Postfach 330440 </street>
   <city> Bremen </city>
   <code>28334 </code>
   <country>Germany</country>
</postal>
<phone>+49-421/21863908 </phone>
<email>izumikawa@kddilabs.jp</email>
</address>
</author>

<author initials="R." surname="Lillie" fullname="Ross Lillie">
<organization abbrev="Motorola Labs">Motorola Labs</organization>
<address>
<postal>
   <street>1301 East Algonquin Road, IL02/2240</street>
   <city>Schaumburg, IL</city>
   <code>60196</code>
   <country>US</country>
</postal>
<phone>+1 847 576 0012</phone>
<email>ross.lillie@motorola.com</email>
</address>
</author>

<author initials="L." surname="Veltri" fullname="Luca Veltri">
<organization abbrev="Univ. of Parma">DII, University of Parma</organization>
<address>
<postal>
   <street> Parco Area delle Scienze 181/A</street>
   <city>Parma</city>
   <code>43100</code>
   <country>Italy</country>
</postal>
<phone>+39 0521 90 5768</phone>
<email>luca.veltri@unipr.it</email>
<uri>http://www.tlc.unipr.it/veltri</uri>
</address>
</author>

<author initials="Y." surname="Kishi" fullname="Yoji Kishi">
<organization abbrev="KDDI Labs">KDDI Labs</organization>
<address>
<postal>
   <street>2-1-15 Ohara</street>
   <city>Fujimino</city>
   <code>356-8502</code>
   <country>Japan</country>
</postal>
<email>kishi@kddilabs.jp</email>
</address>
</author>

<date year="2008"/>

<!-- If the month and year are both specified and are the current ones, xml2rfc will fill in the current day for you. If only the current year is specified, xml2rfc will fill in the current day and month for you. If the year is not the current one, it is necessary to specify at least a month (xml2rfc assumes day="1" if not specified for the purpose of calculating the expiry date).  With drafts it is normally sufficient to specify just the year. -->

<!-- Meta-data Declarations -->

<area>Real-time Applications and Infrastructure</area>

<workgroup>SIPPING Working Group</workgroup>

<!-- WG name at the upperleft corner of the doc,
IETF is fine for individual submissions.  
	If this element is not present, the default is "Network Working Group",
which is used by the RFC Editor as a nod to the history of the IETF. -->

<keyword>draft</keyword>
<keyword>SIP</keyword>
<keyword>session mobility</keyword>
<keyword>vertical handover</keyword>
<keyword>media bi-casting</keyword>

<abstract>
<t>A vertical handover occurs in heterogeneous networks when a session media is moved among different access network technologies within the same device. This document analyses the issue of handling the vertical handover using the Session Initiation Protocol <xref target="SIP">(SIP)</xref>.</t>

</abstract>
  </front>

  <middle>
<section title="Introduction">
<t>Let us consider a terminal (hereafter named "Mobile Host" or MH), that is possibly equipped with different network interfaces (i.e. a subset of WiFi, Bluetooth, GPRS, 3G, 3.5G (HSDPA), fixed Ethernet, WiMax). A user needs to be able to access the Internet using the same MH irrespective the type of network it connects to. Such multimode terminals are already in the market, and we will be gradually get surrounded by a heterogeneous network where several access networks (ANs) supplement each other. Each interface of the MH will receive an IP address from the corresponding AN it connects to. Therefore the mobile host will have a set of different IP addresses and will have to select which one to use when running multimedia sessions with correspondent terminals. While the mobile host moves, the "selected" interface may become not available due to loss of signal, or could suffer high packet loss or packet delay. Under these circumstances, the MH would like to switch to another interface (using a different IP address) keeping the running sessions active, and might adjust the service quality to be optimal depending on  circumstances.  Even with a single interface the connected access network can become not available anymore and the terminal could connect to another Access Network (in this case on the same technology), which provides a different IP address.  If the switch to the new AN is fast enough, the MH could also be interested in keeping the running session active.</t>

<t>This problem can be addressed with many different approaches, at all the different levels of the protocol stack from link layer to application layer. For a review and comparison of these different approaches, see for example <xref target="DUTTA07"/> and <xref target="XiaomingFu"/>. We are interested here in "application level" mobility solutions. The main advantage of application level mobility solutions is that they do not require any support at the networking level and below from the different access networks, which only needs to provide plain IP connectivity. Application level mobility can conduct flexible mobility management, which is another advantage. Within IETF, discussions on using SIP for mobility management date back several years ago and have been mostly carried on in SIPPING WG. For example <xref target="itsumo"/> analysed preliminary requirements and identified issues that need to be resolved in order to develop a mobility management mechanism in a SIP environment. . Furthermore, session mobility using SIP <xref target="SHACHAM"/> has been discussed in SIPPING WG, and is now awaiting processing and publishing as RFC.
</t>

<t>This document addresses issues and requirements regarding a SIP based "application level" mobility solution, focusing on "terminal mobility". Although "terminal mobility" could be considered as a sub-set of "session mobility", we believe that some requirements (e.g. vertical handovers, fast switching) are not adequately covered by the mobility features of current SIP specifications and therefore they need a careful consideration within the SIPPING WG.</t> 

<t>The relevance of this topic is also confirmed by the ongoing work within 3GPP on "Voice Call Continuity" <xref target="3GPP.23.806"/> and "Multimedia Session Continuity" <xref target="3GPP.23.893"/>, which is providing specification of vertical handover solutions using SIP. A discussion of mobility requirements and solution within SIPPING WG provides the chance to consider this 3GGP work.</t>
</section>

<section title="Terminology" anchor="terminology">

<t>This section presents a few terms used throughout the document.</t>

<t>
<list style="symbols">
<t>Vertical handover: A vertical handover refers to changing a MH's point of attachment between different types of access networks, e.g. GPRS and WiFi. The feature of a vertical handover is that the transmission rate and latency may well differ significantly before and after handover, respectively. </t>

<t>Network level mobility: Network level mobility refers to managing an MH's  mobility at the network layer. Mobile IP <xref target="MIP">(MIP)</xref> and  Mobile IPv6 <xref target="MIP6">(MIPv6)</xref> are well-known as network level mobility. One of the merits of network level mobility is making the mobility  of an MH transparent to an application by keeping the same IP address.  This type of mobility management is useful to run existing applications. However, an ubiquitous deployment of MIP/MIPv6 is needed if a MH's mobility is globally supported. In addition, it is difficult for an application to recognize an available bandwidth  and delay before or during handoff and to adjust a transmitting rate or buffer space  to a target network since network level mobility makes the MH's mobility transparent to the upper layer. </t> 

<t>Application level mobility: Application level mobility refers to managing an MH's mobility at the application layer. SIP mobility <xref target="SCHULZRINNE"/> is well-known as application level mobility. Application level mobility has its merits in terms of its ease of deployment. This does not require whole networks to be changed. Therefore, the mobility support can be more and more rapidly deployable using application level mobility. Furthermore, SIP has an affinity for applications, which can make handoff more flexible.  The feature can fit heterogeneous networks where an available bandwidth and delay may well differ significantly among access networks. </t> 

<t>Terminal mobility: Terminal Mobility allows a device to move between IP subnets, while continuing to be reachable for incoming requests and maintaining sessions across subnet changes. (Definition from <xref target="SCHULZRINNE"/>)</t> 

<t>Session mobility: Session mobility allows a user to maintain a media session even while changing terminals. (Definition from <xref target="SCHULZRINNE"/>)</t> 

</list>
</t>

</section>


<section title="Scenario for vertical handover" anchor="scenario for vertical handover">
<t>In this document, we focus on Terminal Mobility. The figure below shows a Mobile Host that wants to communicate with a "Correspondent Host" (CH). The Mobile Host can connect to different Access Networks (AN1, AN2, AN3 are represented in the figure). The different ANs could have different wireless or wired technologies and difference bandwidth/delay, and the Mobile Host could be connected to more than one Access Network at the same time if it has more than one physical network interface. Note that the Access Networks can provide public or private addresses to the mobile host (in most typical scenarios the Access Networks are likely to provide private IP addresses). For example in the figure below AN 1 and AN 3 provide a private address (as shown by the NAT box), while AN2 provides a public address. Similarly, the Correspondent Host can have a public address (like CH 1 in the figure) or a private IP address (like CH 2 in the figure). </t>

<figure align="center"
title="Figure 1 - Network architecture for vertical handover">
<artwork><![CDATA[
                            +-------+
                            |  AN1  |-----+
                        ----|       | NAT |                +--------+
                       /    +-------+-----+                |Corresp.|
          +-------+                         __________     | Host 1 |
          | Mobile|         +-------+      /          \    +--------+
          | Host  |         |  AN2  |     /            \
          +-------+     ----|       |    |   INTERNET   |
                            + - - - +     \            /
                                           \__________/
                       \    +-------+                      +--------+
                        ----|  AN3  |-----+          +-----|Corresp.|
                            |       | NAT |          | NAT | Host 2 |
                            +-------+-----+          +-----+--------+
]]>
</artwork>

</figure>

<t>The goal of the handover mechanism is to let the MH roam among different Access Networks in a seamless way.  Therefore we are addressing the issue of "Terminal Mobility" as defined in previous section. The mobility management mechanism should consider the roaming of the MH both "off call" and during an active call.  The MH should be able to dynamically choose among the available ANs the one that better suits its needs (e.g. perceived quality of media flows and cost) in a given moment.  It is important to notice that this draft does not address the criteria and tools for selection of the "best" access network, it only details the issues and the requirements regarding the mobility management and handover execution mechanism.</t> 
</section>

<section title="Requirements for Vertical Handovers" anchor="requirements">
<t>In this session we discuss a set of requirements that a mobility management solution based on SIP should have. The requirements are divided into two types, i.e., mandatory requirements and optional requirements.</t>

<t>Mandatory Requirements</t>

<t>
<list style="symbols">
<t>The solution should take care of handover as fast as possible.  The goal is to provide a "seamless" handover with no perception from the user point of view.  If it is not possible to provide a truly seamless solution, the impairment should be minimized. For example, voip application is used to evaluate for the performance of SIP-based terminal mobility in <xref target="Salsano07"/> and also video-phone is used in <xref target="IZUMIKAWA"/>. According to the ITU requirement<xref target="ITUT"/>, one-way delay and packet loss rate for conversational voice and video-phone are required to be less than 150 ms and 3%, and 150 ms and 1%, respectively. </t>

<t>The handover solution should not require a support in the different access network.  The access networks are only required to provide IP connectivity (either with public or private addresses) for the forwarding of signalling SIP packets and media RTP packets.</t>

<t>Correspondent Host (CH) (which in general are not moving, but they are communicating with a moving terminal) should be a basic User Agent (UA) which only has basic SIP capabilities. If this requirement is not fulfilled there is the need to change all SIP terminals to support the handovers of Mobile Host.</t>

<t>The handover solution should be compatible with NATted networks, i.e. it should interoperate gracefully with NAT traversal mechanisms for SIP signaling and for session media flows.</t>

</list>
</t>
<t>Optional Desirable Requirements</t>
<t>
<list style="symbols">
<t>The solution may support a "forward handover" (i.e. in which all the procedure is performed on the new target Access Network). This is important if the connection on the old Access Network is suddenly broken.  If possible (i.e. if the connection with the old access network does not break suddenly) the solution could exploit the communication on the old access network in order to better control the handover procedure.  Soft handovers (i.e. where the two active connections can be exploited in the same moment to send the session data) could be exploited.</t>

<t>The switch of the "active" interface during a SIP transaction may be supported.  As an example the terminal should be able to send (receive) an INVITE on the currently active interface, switch to another interface and receive (send) the 200 OK on the other interface.</t>

<t>Decoupling of "user level" registration and mobility and "terminal level" mobility may be allowed.  As an example a user  with AOR "sip:user@domain.com" should be allowed to use different  terminals (i.e.  Mobile Hosts supporting the handover solutions  as well as normal SIP terminals).  These terminals can be used in  sequence or at the same time depending on the capability of his  own "home" registrar/proxy server, and this is independent of the vertical handover solution which takes care of the mobility  of only one specific Mobile Host.  A concrete example for this  requirement is to support a user "sip:user@domain.com" who owns  three Mobile Hosts (one could be his phone, one his PDA, one his  laptop) and two fixed terminals (his desktop and his home VoIP phone).   The vertical handover solution takes care of the mobility of the phone, PDA and laptop as three separate Mobile Hosts (which can also be all active in the same time).</t>

<t>Providing privacy with respect to user location and user movements
would be preferable.</t>

<t>Existing user agents for the Mobile Host may inter-work with  the handover procedure without the need to be updated.  It would be desirable to reuse existing SIP clients (User Agents) without updating them to support the terminal mobility.</t>

<!--<t>Media streams bi-casting would be realized to leverage current or emerging practices in the SIP. In other words, a new method or an extra header for media streams bi-casting can not be required.</t>-->

<t>It is desirable that the service quality is adjusted to optimal levels according to the access network after every handover.</t>

<t>Non-RTP media such as IM or file transfer using MSRP may be supported.</t>

</list>
</t>

</section>

<section title="Taxonomy of possible approaches" anchor="taxonomy">

<t>The application level terminal mobility solutions based on SIP can be classified in "Correspondent host based" or "Intermediate Element based". In addition, a session mobility is introduced just for reference.</t>

<t>
<list style="symbols">
<t>Correspondent Host based terminal mobility solutions</t>
</list>
</t>
<t>RFC 3261 <xref target="SIP"/> has a built in mechanism for mobility management. The "off-call" mobility management consists in the Registration process.  The "on-call" handover is performed using RE-INVITE messages towards the Corresponding Node <xref target="SCHULZRINNE"/>.  No intermediate entities are directly involved in the handover process.  This has the advantage that no additional procedures for the handover need to be implemented in intermediate elements, and that there is no additional load in the networks due to the handovers.  On the other hand, the procedure requires that the Corresponding Node (which in general is not a mobile host) supports the RE-INVITE mechanism.  A second drawback is that the handover delay is directly proportional to the end-to-end delay, and this could be higher with respect to the delay occurring between a mobile node and an intermediate element.</t>


<t>
<list style="symbols">
<t>Intermediate Element based terminal mobility solutions</t>
</list>
</t>
<t>In order to overcome the drawbacks of the Correspondent Host based solutions, "intermediate" entities that take an active role in the handover can be introduced.  Several proposals can be found in the literature, but to our knowledge no internet draft has been proposed in this respect.  Hereafter we mention some of the existing proposals.  In <xref target="Banerjee"/>, intermediate entities are used only to speed up the handover process, but the handover procedure still involves the Corresponding Node as well.  A similar approach is followed in <xref target="PROFITIS"/>, which also deals with location based selection of the "optimal" intermediate entity and of wireless access points.  In <xref target="Salsano08"/> the intermediate entities fully handle the user mobility, hiding the mobility to the Corresponding Nodes.  In  <xref target="IZUMIKAWA"/>, the intermediate entities are used to support MH's mobility as well as adjusting service quality to the MH's target access network. <xref target="DUTTA04"/> also describes different ways "intermediate element"-based approach can expedite handover for single-interface-based terminals. </t>

<t>
<list style="symbols">
<t>Session mobility solutions</t>
</list>
</t>
<t>According to <xref target="SHACHAM"/> session mobility is the transfer of media of an ongoing communication session from one device to another. <xref target="REFER"/>, based on the previous work in <xref target="SCHULZRINNE"/> has defined a framework for session mobility that allows a mobile node to discover available devices and to include them in an active session. <xref target="REFER"/> has demonstrated the suitability of employing either 3PCC <xref target="THIRD">(3PCC)</xref> or SIP's REFER <xref target="REFER"/> method as suitable mechanisms for session mobility between mobile devices. </t>

<t>The problem of session mobility is more general and more complex than the problem of terminal mobility that is addressed by this requirement analysis. It is likely that a solution for the session mobility problem can also solve the terminal mobility problem, but it also needs to consider several aspects that are not relevant to terminal mobility. Just to give two examples: 1) device discovery; 2) signaling procedures to communicate the intention or need to transfer the session from the original device and the target devices. These two aspects are not needed in a solution for terminal mobility, as all the network interfaces are local to the terminal and they do not need to be discovered nor there is the need to communicate the session transfer from one interface to another. On the other hand, there are some specific requirements that could be taken into consideration in terminal mobility. One example is the avoidance of media disruption during the handover. The gap of the media stream in terminal mobility case (on the same terminal) would cause more severe degradation in the user's experience than that in the session mobility case. Such gap should be made minimum or avoided during the handover in terminal mobility.</t> 

<t>For the above reason, we believe that requirements for terminal mobility should be addressed in a separate context than session mobility. Obviously solutions for session mobility could become a part of the solution for terminal mobility</t>
 
   </section>
   
   <section title="Conclusions" anchor="conclusions">
<t>As a concluding remark, we believe that it is important to consider a new solution for vertical handover that meets the set of requirements that has been analysed.  This solution will help providing seamless handover to SIP based application with a better performance and overcoming some shortcomings of the current solution based on <xref target="SIP"/>.</t>
</section>
   
   <section title="Security considerations" anchor="security">
<t>The security considerations should be taken into account in the design of the handover solution, so that no new additional security issues will be introduced.</t>
</section>

<section anchor="IANA" title="IANA Considerations">
<t>This memo includes no request to IANA.</t>

</section>

<section title="Acknowledgments" anchor="Acknowledgments">
     <t>The authors would like to thank a number of people that
     recently contributed to the development of this draft in
     a more clear direction pointing out the issues that need
     to be addressed to advance this document.
     Acknowledgement go to people in the SIPPING working group,
     including: Ashutosh Dutta, Salvatore Loreto, Henning
     Schulzrinne and Henry Sinnreich.</t>
</section>


  </middle>

  <!--  *****BACK MATTER ***** -->

  <back>
<references title="Normative References">
<reference anchor="SIP">

<front>
<title>SIP: Session Initiation Protocol</title>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
<organization/>
</author>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne">
<organization/>
</author>
<author initials="G." surname="Camarillo" fullname="G. Camarillo">
<organization/>
</author>
<author initials="A." surname="Johnston" fullname="A. Johnston">
<organization/>
</author>
<author initials="J." surname="Peterson" fullname="J. Peterson">
<organization/>
</author>

<author initials="R." surname="Sparks" fullname="R. Sparks">
<organization/>
</author>
<author initials="M." surname="Handley" fullname="M. Handley">
<organization/>
</author>
<author initials="E." surname="Schooler" fullname="E. Schooler">
<organization/>
</author>
<date year="2002" month="June"/>
<abstract>
<t>This document describes Session Initiation Protocol (SIP), an application-layer
  control (signaling) protocol for creating, modifying, and terminating sessions with
  one or more participants. These sessions include Internet telephone calls, multimedia
  distribution, and multimedia conferences. [STANDARDS TRACK] </t>
</abstract>
</front>

<seriesInfo name="RFC" value="3261"/>
<format type="TXT" octets="647976" target="ftp://ftp.isi.edu/in-notes/rfc3261.txt"/>
</reference>


<reference anchor="MIP">

<front>
<title>Mobility Support in IPv6</title>
<author initials="C." surname="Perkins" fullname="Charles E. Perkins">
<organization/>
</author>
<date year="2002" month="August"/>
<abstract>
<t>This document specifies protocol enhancements that allow transparent
   routing of IP datagrams to mobile nodes in the Internet.  Each mobile
   node is always identified by its home address, regardless of its
   current point of attachment to the Internet.  While situated away
   from its home, a mobile node is also associated with a care-of
   address, which provides information about its current point of
   attachment to the Internet.  The protocol provides for registering
   the care-of address with a home agent.  The home agent sends
   datagrams destined for the mobile node through a tunnel to the care-
   of address.  After arriving at the end of the tunnel, each datagram
   is then delivered to the mobile node. [STANDARDS TRACK]</t>
</abstract>
</front>

<seriesInfo name="RFC" value="3344"/>
<format type="TXT" octets="241041" target="http://www.ietf.org/rfc/rfc3344.txt"/>
</reference>


<reference anchor="MIP6">

<front>
<title>IP Mobility Support for IPv4</title>
<author initials="D." surname="Johnson" fullname="D. Johnson">
<organization/>
</author>
<author initials="C." surname="Perkins" fullname="Charles E. Perkins">
<organization/>
</author>
<author initials="J." surname="Arkko" fullname="Charles E. Perkins">
<organization/>
</author>
<date year="2004" month="June"/>
<abstract>
<t>This document specifies a protocol which allows nodes to remain
   reachable while moving around in the IPv6 Internet.  Each mobile node
   is always identified by its home address, regardless of its current
   point of attachment to the Internet.  While situated away from its
   home, a mobile node is also associated with a care-of address, which
   provides information about the mobile node's current location.  IPv6
   packets addressed to a mobile node's home address are transparently
   routed to its care-of address.  The protocol enables IPv6 nodes to
   cache the binding of a mobile node's home address with its care-of
   address, and to then send any packets destined for the mobile node
   directly to it at this care-of address.  To support this operation,
   Mobile IPv6 defines a new IPv6 protocol and a new destination option.
   All IPv6 nodes, whether mobile or stationary, can communicate with
   mobile nodes. [STANDARDS TRACK]</t>
</abstract>
</front>

<seriesInfo name="RFC" value="3775"/>
<format type="TXT" octets="393514" target="http://www.ietf.org/rfc/rfc3775.txt"/>
</reference>
<!--
<reference anchor="SDP">

<front>
<title>SDP: Session Description Protocol</title>
<author initials="M." surname="Handley" fullname="M. Handley">
<organization/>
</author>
<author initials="V." surname="Jacobson" fullname="V. Jacobson">
<organization/>
</author>
<author initials="C." surname="Perkins" fullname="C. Perkins">
<organization/>
</author>
<date year="2006" month="July"/>
<abstract>
<t>This memo defines the Session Description Protocol (SDP). SDP is intended for
  describing multimedia sessions for the purposes of session announcement, session
  invitation, and other forms of multimedia session initiation. [STANDARDS TRACK]</t>
</abstract>
</front>

<seriesInfo name="RFC" value="4566"/>
<format type="TXT" octets="108820" target="ftp://ftp.isi.edu/in-notes/rfc4566.txt"/>
</reference>
-->
<reference anchor="THIRD">

<front>
<title>Best Current Practices for Third Party Call Control (3pcc) in the Session
Initiation Protocol (SIP)</title>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
<organization/>
</author>
<author initials="J." surname="Peterson" fullname="J. Peterson">
<organization/>
</author>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne">
<organization/>
</author>
<author initials="G." surname="Camarillo" fullname="Gonzalo Camarillo">
<organization/>
</author>
<date year="2004" month="April"/>
<abstract>

<t>Third party call control refers to the ability of one entity to create a call in
  which communication is actually between other parties. Third party call control is
  possible using the mechanisms specified within the Session Initiation Protocol (SIP).
  However, there are several possible approaches, each with different benefits and
  drawbacks. This document discusses best current practices for the usage of SIP for
  third party call control. This document specifies an Internet Best Current Practices
  for the Internet Community, and requests discussion and suggestions for improvements.
</t>
</abstract>
</front>

<seriesInfo name="BCP" value="85"/>
<seriesInfo name="RFC" value="3725"/>
<format type="TXT" octets="77308" target="ftp://ftp.isi.edu/in-notes/rfc3725.txt"/>
</reference>

<reference anchor="REFER">
<front>
<title>The Session Initiation Protocol (SIP) Refer Method</title>
<author initials="R." surname="Sparks" fullname="R. Sparks">
<organization/>
</author>
<date year="2003" month="April"/>
<abstract>
<t>This document defines the REFER method. This Session Initiation Protocol (SIP)
  extension requests that the recipient REFER to a resource provided in the request. It
  provides a mechanism allowing the party sending the REFER to be notified of the
  outcome of the referenced request. This can be used to enable many applications,
  including call transfer. In addition to the REFER method, this document defines the
  refer event package and the Refer-To request header. [STANDARDS TRACK] </t>
</abstract>
</front>

<seriesInfo name="RFC" value="3515"/>
<format type="TXT" octets="47788" target="ftp://ftp.isi.edu/in-notes/rfc3515.txt"/>
</reference>
<!--
<reference anchor="CASNER">
<front>
<title>A Session Description Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol
(RTCP) Bandwidth</title>
<author initials="S" surname="Casner" fullname="Stephen L. Casner">
<organization/>
</author>
<date year="2003" month="July"/>
<abstract>
<t>This document specifies an Internet standards track protocol for the Internet
  community, and requests discussion and suggestions for improvements. Please refer to
  the current edition of the "Internet Official Protocol Standards" (STD 1) for the
  standardization state and status of this protocol. Distribution of this memo is
  unlimited. [STANDARDS TRACK] </t>
</abstract>
</front>

<seriesInfo name="RFC" value="3556"/>
<format type="TXT" octets="15310" target="ftp://ftp.isi.edu/in-notes/rfc3556.txt"/>
</reference>

<reference anchor="TRANSCODE1">
<front>
<title>Transcoding Services Invocation in the Session Initiation Protocol (SIP) Using
Third Party Call Control (3pcc)</title>
<author initials="G." surname="Camarillo" fullname="Gonzalo Camarillo">
<organization/>
</author>
<author initials="E." surname="Burger" fullname="Eric Burger">
<organization/>
</author>
<author initials="H." surname="Schulzrinne" fullname="Henning Schulzrinne">
<organization/>
</author>
<author initials="A." surname="Wijk" fullname="Arnoud van Wijk">
<organization/>
</author>
<date year="2005" month="June"/>
<abstract>
<t>This memo provides information for the Internet community. It does not specify an
  Internet standard of any kind. Distribution of this memo is unlimited.[Informational]
</t>
</abstract>
</front>

<seriesInfo name="RFC" value="4117"/>
<format type="TXT" octets="37951" target="ftp://ftp.isi.edu/in-notes/rfc4117.txt"/>
</reference>

<reference anchor="GROUP">
<front>
<title>Grouping of Media Lines in the Session Description Protocol (SDP) </title>
<author initials="G." surname="Camarillo" fullname="Gonzalo Camarillo">
<organization/>
</author>
<author initials="J." surname="Holler" fullname="J. Holler">
<organization/>
</author>
<author initials="H." surname="Schulzrinne" fullname="Henning Schulzrinne">
<organization/>
</author>
<date year="2002" month="December"/>
<abstract>
<t>This document defines two Session Description Protocol (SDP) attributes: "group" and
  "mid". They allow to group together several "m" lines for two different purposes: for
  lip synchronization and for receiving media from a single flow (several media streams)
  that are encoded in different formats during a particular session, on different ports
  and host interfaces.[STANDARDS TRACK] </t>
</abstract>
</front>

<seriesInfo name="RFC" value="3388"/>
<format type="TXT" octets="39365" target="ftp://ftp.isi.edu/in-notes/rfc3388.txt"/>
</reference>

<reference anchor="UPDATE">

<front>
<title>The Session Initiation Protocol (SIP) UPDATE Method</title>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
<organization/>
</author>
<date year="2002" month="October"/>
</front>

<seriesInfo name="RFC" value="3311"/>
<format type="TXT" octets="28125" target="ftp://ftp.isi.edu/in-notes/rfc3311.txt"/>
</reference>
-->
</references>

<references title="Informative References">
 

<reference anchor="Banerjee">
<front>
<title>Seamless SIP-Based Mobility for Multimedia Applications</title>
<author initials="N" surname="Banerjee" fullname="N. Banerjee">
<organization/>
</author>
<author initials="A" surname="Acharya" fullname="A. Acharya">
<organization/>
</author>
<author initials="S. K." surname="Das" fullname="S. K. Das">
<organization/>
</author>

<date month="March/April" year="2006"/>
</front>
<seriesInfo name="IEEE Network" value=""/>
</reference>

<reference anchor="SCHULZRINNE">
<front>
<title>Application-Layer Mobility Using SIP</title>
<author initials="H" surname="Schulzrinne">
<organization/>
</author>
<author initials="E" surname="Wedlund">
<organization/>
</author>
<date month="July" year="2000"/>
</front>
<seriesInfo name="ACM Mobile Computing and Communications Review" value="Vol.4, No.3"/>
</reference>

<reference anchor="SHACHAM">
<front>
<title>Session Initiation Protocol (SIP) Session Mobility</title>

<author initials="R" surname="Shacham" fullname="Ron Shacham">
<organization/>
</author>

<date month="November" day="18" year="2007"/>

<abstract>
<t>Session mobility is the transfer of media of an ongoing communication session from
  one device to another. This document describes the general methods and specifies the
  signaling and media flows 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.</t>
</abstract>

</front>

<seriesInfo name="Internet-Draft" value="draft-shacham-sipping-session-mobility-05"/>
<format type="TXT"
target="http://www.ietf.org/internet-drafts/draft-shacham-sipping-session-mobility-05.txt"
/>
</reference>
<!--
<reference anchor="TRANSCODE2">
<front>
<title>Framework for Transcoding with the Session Initiation Protocol (SIP)</title>

<author initials="G" surname="Camarillo" fullname="Gonzalo Camarillo">
<organization/>
</author>

<date month="December" day="1" year="2006"/>

<abstract>
<t>This document defines a framework for transcoding with SIP. This framework includes
  how to discover the need for transcoding services in a session and how to invoke those
  transcoding services. Two models for transcoding services invocation are discussed:
  the conference bridge model and the third party call control model. Both models meet
  the requirements for SIP regarding transcoding services invocation to support deaf,
  hard of hearing, and speech-impaired individuals.</t>
</abstract>

</front>

<seriesInfo name="Internet-Draft" value="draft-ietf-sipping-transc-framework-05"/>
<format type="TXT"
target="http://www.ietf.org/internet-drafts/draft-ietf-sipping-transc-framework-05.txt"/>
</reference>
-->
<reference anchor="Salsano07">
<front>
<title>Architecture and testbed implementation of vertical handovers based on SIP Session Border Controllers</title>
<author initials="S" surname="Salsano" fullname="S. Salsano">
<organization/>
</author>
<author initials="L" surname=" Veltri " >
<organization/>
</author>
<author initials="A" surname=" Polidoro " >
<organization/>
</author>
<author initials="A" surname=" Ordine " >
<organization/>
</author>
<date month="November" year="2007"/>
</front>
<seriesInfo name="Wireless Personal Communications, Springer" value=""/>
</reference>

<reference anchor="Salsano08">
<front>
<title>SIP-based Mobility Management in Next Generation Networks</title>
<author initials="S" surname="Salsano" fullname="S. Salsano">
<organization/>
</author>
<author initials="C" surname="Mingardi" >
<organization/>
</author>
<author initials="S" surname="Niccolini" >
<organization/>
</author>
<author initials="A" surname="Polidoro" >
<organization/>
</author>
<author initials="L" surname="Veltri" >
<organization/>
</author>
<date month="April" year="2008"/>
</front>
<seriesInfo name="IEEE Wireless Communication" value=""/>
</reference>

<reference anchor="itsumo">
<front>
<title>Mobility Management in a SIP Environment Requirements, Functions and Issues</title>
<author initials="F" surname="Vakil" fullname="F. Vakil">
<organization/>
</author>
<author initials="A" surname="Dutta" fullname="A. Dutta">
<organization/>
</author>
<author initials="J-C" surname="Chen" fullname="J-C. ">
<organization/>
</author>
<author initials="S" surname="Baba" fullname="S. Baba">
<organization/>
</author>
<author initials="N" surname="Nakajima" fullname="N. Nakajima">
<organization/>
</author>
<author initials="Y" surname="Shobatake" fullname="Y. Shobatake">
<organization/>
</author>
<author initials="H" surname="Schulzrinne" fullname="H. Schulzrinne">
<organization/>
</author>
<date month="December" year="2000"/>
</front>
<seriesInfo name="Internet-Draft" value="draft-itsumo-sip-mobility-req-02.txt"/>
</reference>


<reference anchor="PROFITIS">
<front>
<title>PROFITIS: architecture for location-based vertical handovers supporting real-time applications</title>
<author initials="S" surname="Tsiakkouris" fullname="S. Tsiakkouris">
<organization/>
</author>
<author initials="I" surname="Tsiakkouris" fullname="I. Tsiakkouris">
<organization/>
</author>
<date month="April" year="2006"/>
</front>
<seriesInfo name="25th IEEE International Performance, Computing, and Communications Conference" value="(IPCCC 2006)"/>
</reference>

<reference anchor="IZUMIKAWA">
<front>
<title>User-centric Seamless Handover Scheme for Realtime Applications</title>
<author initials="H" surname="Izumikawa" fullname="Haruki Izumikawa">
<organization/>
</author>
<author initials="T" surname="Fukuhara" fullname="Tadayuki Fukuhara">
<organization/>
</author>
<author initials="T" surname="Matsunaka" fullname="Takashi Matsunaka">
<organization/>
</author>
<author initials="K" surname="Sugiyama" fullname="Keizo Sugiyama">
<organization> KDDI R&amp;D Laboratories, Inc. </organization>
</author>
<date month="September" year="2007"/>
</front>
<seriesInfo
name="IEEE Internation Symposium on Personal, Indoor and
Mobile Radio Communications"
value="(PIMRC'07)"/>

</reference>

<reference anchor="XiaomingFu">
<front>
<title>A review of Mobility Support Paradigms for the Internet</title>
<author initials="D" surname="Le">
<organization/>
</author>
<author initials="X" surname="Fu">
<organization/>
</author>
<author initials="D" surname="Hogrefe">
<organization/>
</author>
<date month="1st quarter" year="2006"/>
</front>
<seriesInfo name="IEEE Communications Surveys & Tutorials" value="Vol.8, No.1"/>
</reference>


<reference anchor="DUTTA07">
<front>
<title>Generalized Modeling Framework for Handoff Analysis</title>
<author initials="A" surname="Dutta">
<organization/>
</author>
<author initials="B" surname="Lyles">
<organization/>
</author>
<author initials="H" surname="Schulzrinne">
<organization/>
</author>
<author initials="T" surname="Chiba">
<organization/>
</author>
<author initials="H" surname="Yokota">
<organization/>
</author>
<author initials="A" surname="Idoue">
<organization/>
</author>
<date month="September" year="2007"/>
</front>
<seriesInfo name="18th Annual IEEE International Symposium on Personal, Indoor and Mobile Radio Communications" value="(PIMRC 2007)"/>
</reference>

<reference anchor="DUTTA04">
<front>
<title> Fast-handoff Schemes for Application Layer Mobility Management </title>
<author initials="A" surname="Dutta">
<organization/>
</author>
<author initials="S" surname="Madhani">
<organization/>
</author>
<author initials="W" surname="Chen">
<organization/>
</author>
<author initials="H" surname="Schulzrinne">
<organization/>
</author>
<date month="September" year="2004"/>
</front>
<seriesInfo name="15th Annual IEEE International Symposium on Personal, Indoor and Mobile Radio Communications" value="(PIMRC 2004)"/>
</reference>


<reference anchor='3GPP.23.806'>
<front>
<title>Voice call continuity between Circuit Switched (CS) and IP Multimedia Subsystem (IMS) Study</title>
<author><organization>3GPP</organization></author>
<date day='15' month='December' year='2005' />
</front>

<seriesInfo name='3GPP TR' value='23.806 7.0.0' />
<format type='HTML' target='http://www.3gpp.org/ftp/Specs/html-info/23806.htm' />
</reference>


<reference anchor='3GPP.23.893'>
<front>
<title>Feasibility study on multimedia session continuity; Stage 2</title>
<author><organization>3GPP</organization></author>
<date day='09' month='June' year='2008' />
</front>

<seriesInfo name='3GPP TR' value='23.893 8.0.0' />
<format type='HTML' target='http://www.3gpp.org/ftp/Specs/html-info/23893.htm' />
</reference>

<reference anchor='ITUT'>
<front>
<title> ITU-T G.1010 End-user multimedia QoS categories </title>
<author><organization>ITU-T</organization></author>
<date year='2001' />
</front>

</reference>


<!--
<reference anchor="SALSANO">
<front>
<title>A solution for vertical handover of multimedia sessions using SIP</title>
<author initials="S" surname="Salsano" fullname="Stefano Salsano">
<organization> Univ. of Rome Tor Vergata </organization>
</author>
<author initials="N" surname="Niccolini" fullname="Saverio Niccolini">
<organization> NEC </organization>
</author>
<author initials="V" surname="Veltri" fullname="Luca Veltri">
<organization> Univ. of Parma </organization>
</author>
<author initials="P" surname="Polidoro" fullname="Andrea Polidoro">
<organization> Univ. of Rome Tor Vergata </organization>
</author>
<date month="August" day="28" year="2007"/>
<abstract>
<t>This document proposes a solution for handling vertical handovers among different
  network technologies using SIP, fulfilling a set of requirements discussed in the
  document "Requirements for vertical handover of multimedia sessions using SIP"
  (draft-niccolini-sipping-siphandover-01). The solution requires a new header field
  (named "Handover") and a new parameter in the Via header field (named "MMID").</t>
</abstract>
</front>
<seriesInfo name="Internet-Draft" value="draft-salsano-sipping-siphandover-solution-01"/>
<format type="TXT"
target="http://www.ietf.org/internet-drafts/draft-salsano-sipping-siphandover-solution-01.txt"
/>
</reference>
-->
</references>
  </back>

</rfc>



