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<rfc category="info" docName="draft-niccolini-sipping-siphandover-04" ipr="full3978">

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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>2-1-15 Ohara</street>
               <city>Fujimino</city>
               <code>356-8502</code>
               <country>Japan</country>
            </postal>
            <phone>+81-49-278-7866</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 
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	 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),
      possibly equipped with different network interfaces (i.e. a subset of
      WiFi, Bluetooth, GPRS, 3G, fixed Ethernet, WiMax).  A user will be able
      to access the Internet using the same MH.  This seems to be the trend for 
      the foreseeable future, i.e. we will be 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.  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 has been addressed with different approaches.  One
      approach is based on "network level" mobility solutions like Mobile
      IP or MobileIPv6.  Another approach is based on "application level"
      mobility solution.  The main advantage of application level mobility
      solutions is that they do not require any support at the network
      level 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. This document details 
      the issues and the requirements regarding an "application level" mobility 
      solution, in particular considering the solution that exploits the SIP protocol.
      The aim of SIP based application level mobility is to keep active a
      multimedia session that has been established with SIP, while the
      terminal switches from one network interface to another or it changes
      IP address on the same interface.</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 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 an 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 lavel 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>
                        
        </list>
      </t>

    </section>


    <section title="Scenario for vertical handover" anchor="scenario for vertical handover">
      <t>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 techologies 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.  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 mechanims.</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 be 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.</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 an 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 currenlty 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 (service) 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 procedure for the handover needs to be
      implemented in network elements, and that there is no additional load
      in the networks due to the handovers.  On the other hand, the
      procedure requirese 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 entity.</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.</t>
      
      <t>
        <list style="symbols">
          <t>Session (service) mobility solutions</t>
      </list>
      </t>
      <t>Concerning session (service) mobility using SIP, <xref target="SCHULZRINNE"/>
      has examined through the use of both Third-Party Call Control 
        <xref target="THIRD">(3PCC)</xref> as well as SIP's REFER <xref target="REFER"/> 
        method as possible solutions.  In addition, more recently, <xref
          target="SHACHAM"/> has expanded upon this work and demonstrated the suitability of
        employing either 3PCC or SIP's REFER request as suitable mechanisms for session 
        (service) mobility between mobile devices. </t>

      <t>Although such session (service) mobility <xref target="SCHULZRINNE"/> <xref
          target="SHACHAM"/> can maintain a media session even while changing terminal
          through the use of 3PCC <xref target="THIRD"/> or SIP's REFER <xref target="REFER"/> 
          method, there can be a service interruption during the hard swithing (handover). 
          Therefore, it is necessary to look into a fast/seamless handover for SIP-based 
          terminal mobility where the existing media session seamlessly continues on the 
          same device after a handover.</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>

  </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="et" surname="al" fullname="et al.">
            <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="" surname="" fullname="et al.">
            <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="" surname="" fullname="et al.">
            <organization/>
          </author>
          <date month="April" year="2008"/>
        </front>
        <seriesInfo name="IEEE Wireless Communication" value=""/>
      </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="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>
