path: root/spec/Call_Stream_Interface_Media.xml

<?xml version="1.0" ?>
<node name="/Call_Stream_Interface_Media"
  xmlns:tp="http://telepathy.freedesktop.org/wiki/DbusSpec#extensions-v0">
  <tp:copyright>Copyright © 2009-2010 Collabora Ltd.</tp:copyright>
  <tp:copyright>Copyright © 2009-2010 Nokia Corporation</tp:copyright>
  <tp:license xmlns="http://www.w3.org/1999/xhtml">
    <p>This library is free software; you can redistribute it and/or
      modify it under the terms of the GNU Lesser General Public
      License as published by the Free Software Foundation; either
      version 2.1 of the License, or (at your option) any later version.</p>

    <p>This library is distributed in the hope that it will be useful,
      but WITHOUT ANY WARRANTY; without even the implied warranty of
      MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
      Lesser General Public License for more details.</p>

    <p>You should have received a copy of the GNU Lesser General Public
      License along with this library; if not, write to the Free Software
      Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
      02110-1301, USA.</p>
  </tp:license>

  <interface name="org.freedesktop.Telepathy.Call1.Stream.Interface.Media">
    <tp:added version="0.25.2">(as stable API)</tp:added>
    <tp:requires interface="org.freedesktop.Telepathy.Call1.Stream"/>

    <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
      <p>This interface deals with how to connect a stream to an
      endpoint.  It contains all that is required to describe the
      local endpoint, to succesfully establish a connection. While a
      call is established, one may try to connect to multiple remote
      endpoints at the same time. This is called forking in the SIP
      jargon.  Informations related to the connections are on the
      <tp:dbus-ref
      namespace="ofdT.Call1.Stream">Endpoint</tp:dbus-ref>
      objects. Once the call is established, there MUST be a single
      endpoint left.</p>

      <h4>ICE restarts</h4>

      <p>If the CM wants to do an ICE restart, then the
        <tp:member-ref>ICERestartPending</tp:member-ref> property is set,
        and the <tp:member-ref>ICERestartRequested</tp:member-ref> signal is
        emitted. The streaming implementation should then call
        <tp:member-ref>SetCredentials</tp:member-ref> again. This will trigger
        the actual ICE restart, and cause
        <tp:member-ref>LocalCandidates</tp:member-ref> to be cleared.</p>

      <p>For more information on ICE restarts see
        <a href="http://tools.ietf.org/html/rfc5245#section-9.1.1.1">RFC 5245
        section 9.1.1.1</a></p>
    </tp:docstring>

    <tp:enum type="u" name="Stream_Flow_State">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        The type of <tp:member-ref>SendingState</tp:member-ref>
        and <tp:member-ref>ReceivingState</tp:member-ref>.
      </tp:docstring>
      <tp:enumvalue suffix="Stopped" value="0">
        <tp:docstring>
          No data is flowing (or expected to be flowing) at this time.
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Pending_Start" value="1">
        <tp:docstring>
          The streaming implementation has been told to start or receiving,
          but has not yet indicated that it is doing so.
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Pending_Stop" value="2">
        <tp:docstring>
          The streaming implementation has been told to stop sending or
          receiving data, but it has not yet indicated that it has done so.
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Started" value="3">
        <tp:docstring>
          The streaming implementation is successfully sending or receiving
          data, and everything is going swimmingly.
        </tp:docstring>
      </tp:enumvalue>
    </tp:enum>

    <property name="SendingState" tp:name-for-bindings="Sending_State"
        type="u" tp:type="Stream_Flow_State" access="read">
      <tp:docstring>
        Indicates whether the streaming implementation is/should be sending
        media for this stream. The streaming implementation should be able to
        rely on reading this value and listening to
        <tp:member-ref>SendingStateChanged</tp:member-ref> to
        determine whether it should be sending media or not. It should not
        need to listen to the Hold interfaces on the Call/Content.
        Feedback on success should be given via
        <tp:member-ref>CompleteSendingStateChange</tp:member-ref>. Failures
        should be reported via <tp:member-ref>ReportSendingFailure</tp:member-ref>.
      </tp:docstring>
    </property>

    <signal name="SendingStateChanged"
        tp:name-for-bindings="Sending_State_Changed">
      <tp:docstring>
        Change notification for <tp:member-ref>SendingState</tp:member-ref>.
        Note that this information is duplicated onto the Stream interface, so
        that UIs can ignore the Media interface, and streaming implementations
        can ignore everything but the media interface.
      </tp:docstring>
      <arg name="State" type="u" tp:type="Stream_Flow_State">
        <tp:docstring>
          The new value of SendingState.
        </tp:docstring>
      </arg>
    </signal>

    <method name="CompleteSendingStateChange"
        tp:name-for-bindings="Complete_Sending_State_Change">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        <p>Called in response to
          <tp:member-ref>SendingStateChanged</tp:member-ref>(Pending_*, *) to
          indicate that the media state has successfully progressed from
          Pending_{Start, Stop, Pause} to the corresponding non-pending
          state.</p>
      </tp:docstring>
      <arg name="State" type="u" tp:type="Stream_Flow_State" direction="in">
        <tp:docstring>
          The new (non-pending) value of SendingState.
        </tp:docstring>
      </arg>
      <tp:possible-errors>
        <tp:error name="org.freedesktop.Telepathy.Error.InvalidArgument">
          <tp:docstring>
            The state change made no sense, and was ignored by the CM. The
            most likely cause for this is a race-condition between the CM
            emitting a new state change and the streaming implementation
            responding to the previous state change.
          </tp:docstring>
        </tp:error>
      </tp:possible-errors>
    </method>

    <method name="ReportSendingFailure"
        tp:name-for-bindings="Report_Sending_Failure">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        Can be called at any point to indicate a failure in the outgoing
        portion of the stream.
      </tp:docstring>
      <arg name="Reason" type="u" tp:type="Call_State_Change_Reason"
        direction="in"/>
      <arg name="Error" type="s" tp:type="DBus_Error_Name" direction="in"/>
      <arg name="Message" type="s" direction="in"/>
    </method>

    <property name="ReceivingState" tp:name-for-bindings="Receiving_State"
        type="u" tp:type="Stream_Flow_State" access="read">
      <tp:docstring>
        The counterpart of <tp:member-ref>SendingState</tp:member-ref>.
        Indicates whether the streaming implementation is/should be expecting
        to receive media for this stream. The CM should only tell the
        streaming implementation to stop receiving if it has been told to put
        the stream on hold, or the stream has been removed from the call.
      </tp:docstring>
    </property>

    <signal name="ReceivingStateChanged"
        tp:name-for-bindings="Receiving_State_Changed">
      <tp:docstring>
        Change notification for <tp:member-ref>ReceivingState</tp:member-ref>.
      </tp:docstring>
      <arg name="State" type="u" tp:type="Stream_Flow_State">
        <tp:docstring>
          The new value of ReceivingState.
        </tp:docstring>
      </arg>
    </signal>

    <method name="CompleteReceivingStateChange"
        tp:name-for-bindings="Complete_Receiving_State_Change">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        <p>Called in response to
          <tp:member-ref>ReceivingStateChanged</tp:member-ref>(Pending_*, *) to
          indicate that the media state has successfully progressed from
          Pending_{Start, Stop, Pause} to the corresponding non-pending
          state.</p>
      </tp:docstring>
      <arg name="State" type="u" tp:type="Stream_Flow_State" direction="in">
        <tp:docstring>
          The new (non-pending) value of ReceivingState.
        </tp:docstring>
      </arg>
      <tp:possible-errors>
        <tp:error name="org.freedesktop.Telepathy.Error.InvalidArgument">
          <tp:docstring>
            The state change made no sense, and was ignored by the CM. The
            most likely cause for this is a race-condition between the CM
            emitting a new state change and the streaming implementation
            responding to the previous state change.
          </tp:docstring>
        </tp:error>
      </tp:possible-errors>
    </method>

    <method name="ReportReceivingFailure"
        tp:name-for-bindings="Report_Receiving_Failure">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        Can be called at any point to indicate a failure in the incoming
        portion of the stream.
      </tp:docstring>
      <arg name="Reason" type="u" tp:type="Call_State_Change_Reason"
        direction="in"/>
      <arg name="Error" type="s" tp:type="DBus_Error_Name" direction="in"/>
      <arg name="Message" type="s" direction="in"/>
    </method>

    <method name="SetCredentials" tp:name-for-bindings="Set_Credentials">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        <p>Used to set the username fragment and password for streams that have
          global credentials.</p>
      </tp:docstring>
      <arg name="Username" type="s" direction="in">
        <tp:docstring>
          The username to use when authenticating on the stream.
        </tp:docstring>
      </arg>
      <arg name="Password" type="s" direction="in">
        <tp:docstring>
          The password to use when authenticating on the stream.
        </tp:docstring>
      </arg>
    </method>

    <tp:enum type="u" name="Call_Stream_Candidate_Type">
      <tp:docstring>
        The network topology that an IP candidate represents. This can
        sometimes be used to infer what kind of performance characteristics
        (latency, bandwith, etc) can be expected of connections made to this
        candidate.
      </tp:docstring>
      <tp:enumvalue suffix="None" value="0">
        <tp:docstring>
          This is not an IP candidate. This is a reserved value, and should
          not be seen on the bus.
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Host" value="1">
        <tp:docstring>
          This candidate represents a direct connection to the host, as its
          address is taken directly the host's IP stack.
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Server_Reflexive" value="2">
        <tp:docstring>
          This candidate probably represents a connection to the host through
          a NAT device, as its address was discovered by sending a binding
          request to a STUN server or similar.
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Peer_Reflexive" value="3">
        <tp:docstring>
          This candidate probably represents a good route between the host and
          its peer, as its address was discovered by sending a STUN binding
          request to one of the candidates advertised by the peer.
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Relay" value="4">
        <tp:docstring>
          This candidate represents the address of a relay server (usually
          somewhere on the public internet). This candidate is the most likely
          to work, but all media will go via a relay server, so latency is
          likely to be higher than other types of candidate.
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Multicast" value="5">
        <tp:docstring>
          This candidate represents a Multicast group. This value should only
          appear if the Stream's <tp:member-ref>Transport</tp:member-ref> is
          set to Multicast.
        </tp:docstring>
      </tp:enumvalue>
    </tp:enum>


    <tp:mapping name="Candidate_Info">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        <p>Extra information about the candidate. Allowed and mandatory keys
          depend on the transport protocol used. The following keys are commenly
          used:</p>

        <dl>
          <dt><code>type</code> - u</dt>
          <dd>The type of candidate
            (<tp:type>Call_Stream_Candidate_Type</tp:type>)</dd>

          <dt><code>foundation</code> - s</dt>
          <dd>The foundation of this candidate</dd>

          <dt><code>protocol</code> - u</dt>
          <dd>Underlying protocol of the candidate
            (<tp:type>Media_Stream_Base_Proto</tp:type>) </dd>

          <dt><code>priority</code> - u</dt>
          <dd>Priority of the candidate (should be a number between 0 and
            65535). Most ICE implementations will prefer the highest priority
            candidate pair that manages to connect. For backwards
            compatibility with non-ICE SIP clients, the lowest priority
            candidate may be sent as a raw UDP fallback candidate.
            It is recommended that a relay candidate is used as the lowest
            priority candidate if possible. If both IPv4 and IPv6 raw udp
            fallback candidates are available, they should be set to the
            same priority and advertised to the CM at the same time. The CM
            will decide which to advertise to the remote end.</dd>

          <dt><code>base-ip</code> - s</dt>
          <dd>The underlying Host address where media sent to this
            (non-host-type) candidate will eventually arrive.</dd>

          <dt><code>base-port</code> - u</dt>
          <dd>The underlying Host port where media sent to this
            (non-host-type) candidate will eventually arrive.</dd>

          <dt><code>username</code> - s</dt>
          <dd>Username of this candidate
            (only if credentials are per candidate)</dd>

          <dt><code>password</code> - s</dt>
          <dd>Password of this candidate
            (only if credentials are per candidate)</dd>

          <dt><code>ttl</code> - u</dt>
          <dd>The TTL mandated for RTP/RTCP packets sent to a multicast group
            (only valid for Multicast Streams)</dd>
        </dl>
      </tp:docstring>
      <tp:member name="Key" type="s">
        <tp:docstring>One of the well-known keys documented here, or an
          implementation-specific key.</tp:docstring>
      </tp:member>
      <tp:member name="Value" type="v">
        <tp:docstring>The value corresponding to that key.</tp:docstring>
      </tp:member>
    </tp:mapping>

    <tp:enum type="u" name="Stream_Component">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        Media streams can use more than one UDP socket: one for RTP (data)
        and one for RTCP (control). Most of the time, they are adjacent
        to each other, but some protocols (xmpp) signal each port separately.
      </tp:docstring>
      <tp:enumvalue suffix="Unknown" value="0">
        <tp:docstring>
          The stream transport type is unknown or not applicable
          (should not appear over dbus).
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Data" value="1">
        <tp:docstring>
          This is the high-traffic data socket, containing the audio/video
          data for the stream.
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Control" value="2">
        <tp:docstring>
          This is the low-traffic control socket, usually containing feedback
          about packet loss etc.
        </tp:docstring>
      </tp:enumvalue>
    </tp:enum>

    <tp:struct name="Candidate" array-name="Candidate_List">
      <tp:docstring>A Stream Candidate.</tp:docstring>
      <tp:member name="Component" type="u" tp:type="Stream_Component">
        <tp:docstring>The component number.</tp:docstring>
      </tp:member>
      <tp:member name="IP" type="s">
        <tp:docstring>The IP address to use.</tp:docstring>
      </tp:member>
      <tp:member name="Port" type="u">
        <tp:docstring>The port number to use.</tp:docstring>
      </tp:member>
      <tp:member name="Info" type="a{sv}" tp:type="Candidate_Info">
        <tp:docstring>Additional information about the candidate.</tp:docstring>
      </tp:member>
    </tp:struct>

    <method name="AddCandidates" tp:name-for-bindings="Add_Candidates">
      <tp:docstring>
        Add candidates to the
        <tp:member-ref>LocalCandidates</tp:member-ref> property and
        signal them to the remote contact(s). Note that connection managers
        MAY delay the sending of candidates until
        <tp:member-ref>FinishInitialCandidates</tp:member-ref> is called.
      </tp:docstring>
      <arg name="Candidates" direction="in"
        type="a(usua{sv})" tp:type="Candidate[]">
        <tp:docstring>
          The candidates to be added.
        </tp:docstring>
      </arg>
    </method>

    <method name="FinishInitialCandidates"
      tp:name-for-bindings="Finish_Initial_Candidates">
      <tp:docstring>
        This indicates to the CM that the initial batch of candidates
        has been added, and should now be processed/sent to the remote side.
        <tp:rationale>
          Protocols supporting Raw UDP SHOULD wait for FinishInitialCandidates,
          and then set the lowest priority candidate as the Raw UDP candidate.
        </tp:rationale>
      </tp:docstring>

      <tp:possible-errors>
        <tp:error name="org.freedesktop.Telepathy.Error.NotAvailable">
          <tp:docstring>
            The minimal required candidates have not been set. For
            example, for an RTP protocol, at least one candidate on the
            component 1 (RTP) must have been set.
          </tp:docstring>
        </tp:error>
      </tp:possible-errors>
    </method>

    <tp:enum type="u" name="Stream_Transport_Type">
      <tp:changed version="0.21.2">WLM_8_5 was removed</tp:changed>
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        A transport that can be used for streaming.
      </tp:docstring>
      <tp:enumvalue suffix="Unknown" value="0">
        <tp:docstring>
          The stream transport type is unknown or not applicable
          (for streams that do not have a configurable transport).
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Raw_UDP" value="1">
        <tp:docstring>
          Raw UDP, with or without STUN. All streaming clients are assumed to
          support this transport, so there is no handler capability token for
          it in the <tp:dbus-ref namespace="ofdT.Channel.Type"
          >Call1</tp:dbus-ref> interface.
          [This corresponds to "none" or "stun" in the old Media.StreamHandler
          interface.]
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="ICE" value="2">
        <tp:docstring>
          Interactive Connectivity Establishment, as defined by RFC
          5245. Note that this value covers ICE-UDP only.
          [This corresponds to "ice-udp" in the old
          Media.StreamHandler interface.]
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="GTalk_P2P" value="3">
        <tp:docstring>
          Google Talk peer-to-peer connectivity establishment, as implemented
          by libjingle 0.3.
          [This corresponds to "gtalk-p2p" in the old Media.StreamHandler
          interface.]
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="WLM_2009" value="4">
        <tp:docstring>
          The transport used by Windows Live Messenger 2009 or later, which
          resembles ICE draft 19.
          [This corresponds to "wlm-2009" in the old Media.StreamHandler
          interface.]
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="SHM" value="5">
        <tp:added version="0.21.2"/>
        <tp:docstring>
          Shared memory transport, as implemented by the GStreamer
          shmsrc and shmsink plugins.
        </tp:docstring>
      </tp:enumvalue>
      <tp:enumvalue suffix="Multicast" value="6">
        <tp:added version="0.21.5"/>
        <tp:docstring>
          Multicast transport.
        </tp:docstring>
      </tp:enumvalue>
    </tp:enum>

    <property name="Transport" tp:name-for-bindings="Transport"
        type="u" tp:type="Stream_Transport_Type" access="read" tp:immutable="yes">
      <tp:docstring>
        The transport for this stream.
      </tp:docstring>
    </property>

    <property name="LocalCandidates" tp:name-for-bindings="Local_Candidates"
      type="a(usua{sv})" tp:type="Candidate[]" access="read">
      <tp:docstring>
        [FIXME]. Change notification is via the
        <tp:member-ref>LocalCandidatesAdded</tp:member-ref> signal.
      </tp:docstring>
    </property>

    <signal name="LocalCandidatesAdded"
      tp:name-for-bindings="Local_Candidates_Added">
      <tp:docstring>
        Emitted when local candidates are added to the
        <tp:member-ref>LocalCandidates</tp:member-ref> property.
      </tp:docstring>
      <arg name="Candidates" type="a(usua{sv})" tp:type="Candidate[]">
        <tp:docstring>
          Candidates that have been added.
        </tp:docstring>
      </arg>
    </signal>

    <tp:struct name="Stream_Credentials">
      <tp:docstring>A username and password pair.</tp:docstring>

      <tp:member name="Username" type="s">
        <tp:docstring>The username.</tp:docstring>
      </tp:member>

      <tp:member name="Password" type="s">
        <tp:docstring>The password.</tp:docstring>
      </tp:member>
    </tp:struct>

    <property name="LocalCredentials" tp:name-for-bindings="Local_Credentials"
      type="(ss)" tp:type="Stream_Credentials" access="read">
      <tp:docstring>
        The local credentials are sent to the remote site over the
        signalling protocol. They are used in ICE to make sure that
        the connectivity checks come from the right peer. Change
        notification is via the
        <tp:member-ref>LocalCredentialsChanged</tp:member-ref> signal.

        This property will be a pair of empty strings if ICE has not yet been
        started.
      </tp:docstring>
    </property>

    <signal name="LocalCredentialsChanged"
      tp:name-for-bindings="Local_Credentials_Changed">
      <tp:changed version="0.21.2">renamed from LocalCredentailsSet</tp:changed>
      <tp:docstring>
        Emitted when the value of
        <tp:member-ref>LocalCredentials</tp:member-ref> changes to a non-empty
        value. This should only happen when the streaming implementation calls
        <tp:member-ref>SetCredentials</tp:member-ref>, so this signal is
        mostly useful for debugging.
      </tp:docstring>
      <arg name="Username" type="s" />
      <arg name="Password" type="s" />
    </signal>

    <signal name="RelayInfoChanged"
      tp:name-for-bindings="Relay_Info_Changed">
      <tp:docstring>
        Emitted when the value of
        <tp:member-ref>RelayInfo</tp:member-ref> changes.
      </tp:docstring>
      <arg name="Relay_Info" type="aa{sv}" tp:type="String_Variant_Map[]" />
    </signal>

    <signal name="STUNServersChanged"
      tp:name-for-bindings="STUN_Servers_Changed">
      <tp:docstring>
        Emitted when the value of
        <tp:member-ref>STUNServers</tp:member-ref> changes.
      </tp:docstring>
      <arg name="Servers" type="a(sq)" tp:type="Socket_Address_IP[]" />
    </signal>

    <property name="STUNServers" tp:name-for-bindings="STUN_Servers"
      type="a(sq)" tp:type="Socket_Address_IP[]" access="read">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        <p>The IP addresses of possible STUN servers to use for NAT
          traversal, as dotted-quad IPv4 address literals or RFC2373
          IPv6 address literals.  Change notification is via the
          <tp:member-ref>STUNServersChanged</tp:member-ref>
          signal. The IP addresses MUST NOT be given as DNS hostnames.</p>

        <tp:rationale>
          High-quality connection managers already need an asynchronous
          DNS resolver, so they might as well resolve this name to an IP
          to make life easier for streaming implementations.
        </tp:rationale>
      </tp:docstring>
    </property>

    <property name="RelayInfo" type="aa{sv}" access="read"
      tp:type="String_Variant_Map[]" tp:name-for-bindings="Relay_Info">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        <p>A list of mappings describing TURN or Google relay servers
          available for the client to use in its candidate gathering, as
          determined from the protocol. Well-known map keys are:</p>

        <dl>
          <dt><code>ip</code> - s</dt>
          <dd>The IP address of the relay server as a dotted-quad IPv4
            address literal or an RFC2373 IPv6 address literal. This MUST NOT
            be a DNS hostname.

            <tp:rationale>
              High-quality connection managers already need an asynchronous
              DNS resolver, so they might as well resolve this name to an IP
              and make life easier for streaming implementations.
            </tp:rationale>
          </dd>

          <dt><code>type</code> - s</dt>
          <dd>
            <p>Either <code>udp</code> for UDP (UDP MUST be assumed if this
              key is omitted), <code>tcp</code> for TCP, or
              <code>tls</code>.</p>

            <p>The precise meaning of this key depends on the
              <tp:member-ref>Transport</tp:member-ref> property: if
              Transport is ICE, <code>tls</code> means
              TLS over TCP as referenced by ICE draft 19, and if
              Transport is GTalk_P2P, <code>tls</code> means
              a fake SSL session over TCP as implemented by libjingle.</p>
          </dd>

          <dt><code>port</code> - q</dt>
          <dd>The UDP or TCP port of the relay server as an ASCII unsigned
            integer</dd>

          <dt><code>unique-id</code> - s</dt>
          <dd>A string identifying the relay server. If two RelayInfo entries
            have the same unique-id, but different <code>type</code>s, there
            is usually little point in connecting to both. Use
            <code>priority</code> to determine which version to prefer in this
            case. Can also be used by the streaming implementation to avoid
            connecting to the same relay multiple times if relaying is
            required for both audio and video.</dd>

          <dt><code>priority</code> - u</dt>
          <dd>A number determining which version of a server to prefer (if
            multiple are present with the same <code>unique-id</code>,
            the one with the highest priority should be used, or the streaming
            implementation should use the one whose <code>type</code> has the
            most desirable properties)</dd>

          <dt><code>username</code> - s</dt>
          <dd>The username to use</dd>

          <dt><code>password</code> - s</dt>
          <dd>The password to use</dd>

          <dt><code>component</code> - u</dt>
          <dd>The component number to use this relay server for, as an
            ASCII unsigned integer; if not included, this relay server
            may be used for any or all components.

            <tp:rationale>
              In ICE draft 6, as used by Google Talk, credentials are only
              valid once, so each component needs relaying separately.
            </tp:rationale>
          </dd>
        </dl>

        <tp:rationale>
          <p>An equivalent of the gtalk-p2p-relay-token property on
            MediaSignalling channels is not included here. The connection
            manager should be responsible for making the necessary HTTP
            requests to turn the token into a username and password.</p>
        </tp:rationale>

        <p>The type of relay server that this represents depends on
          the value of the <tp:member-ref>Transport</tp:member-ref>
          property. If Transport is ICE, this is a TURN server;
          if Transport is GTalk_P2P, this is a Google relay server;
          otherwise, the meaning of RelayInfo is undefined.</p>

        <p>If relaying is not possible for this stream, the list is
          empty.</p>

        <p>Change notification is given via the
          <tp:member-ref>RelayInfoChanged</tp:member-ref> signal.</p>
      </tp:docstring>
    </property>

    <signal name="ServerInfoRetrieved"
      tp:name-for-bindings="Server_Info_Retrieved">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        <p>Signals that the initial information about STUN and Relay servers
          has been retrieved, i.e. the
          <tp:member-ref>HasServerInfo</tp:member-ref> property is
          now true.</p>
      </tp:docstring>
    </signal>

    <property name="HasServerInfo" type="b"
        tp:name-for-bindings="Has_Server_Info" access="read">
      <tp:docstring xmlns="http://www.w3.org/1999/xhtml">
        <p>True if all the initial information about STUN servers and Relay
          servers has been retrieved. Change notification is via the
          <tp:member-ref>ServerInfoRetrieved</tp:member-ref> signal.</p>

        <tp:rationale>
          Streaming implementations that can't cope with STUN and
          relay servers being added later SHOULD wait for this
          property to become true before proceeding.
        </tp:rationale>
      </tp:docstring>
    </property>

    <signal name="EndpointsChanged"
      tp:name-for-bindings="Endpoints_Changed">
      <tp:docstring>
        Emitted when the <tp:member-ref>Endpoints</tp:member-ref> property
        changes.
      </tp:docstring>
      <arg name="Endpoints_Added" type="ao">
        <tp:docstring>
          Endpoints that were added.
        </tp:docstring>
      </arg>
      <arg name="Endpoints_Removed" type="ao">
        <tp:docstring>
          Endpoints that no longer exist.
        </tp:docstring>
      </arg>
    </signal>

    <property name="Endpoints" tp:name-for-bindings="Endpoints"
      type="ao" access="read">
      <tp:docstring>
        <p>The list of <tp:dbus-ref namespace="ofdT.Call1.Stream"
          >Endpoint</tp:dbus-ref> objects that exist for this
          stream.</p>

        <p>Change notification is via the
          <tp:member-ref>EndpointsChanged</tp:member-ref> signal.</p>
      </tp:docstring>
    </property>

    <signal name="ICERestartRequested"
      tp:name-for-bindings="ICE_Restart_Requested">
      <tp:docstring>
        Emitted when the remote side requests an ICE restart (e.g. third party
        call control, when the remote endpoint changes). The streaming
        implementation should call
        <tp:member-ref>SetCredentials</tp:member-ref> again.
      </tp:docstring>
    </signal>

    <property name="ICERestartPending"
      tp:name-for-bindings="ICE_Restart_Pending" access="read" type="b">
      <tp:docstring>
        State recovery for <tp:member-ref>ICERestartRequested</tp:member-ref>.
        Set when the signal is emitted, and unset when
        <tp:member-ref>SetCredentials</tp:member-ref> is called.
        Useful for debugging.
      </tp:docstring>
    </property>

    <method name="Fail" tp:name-for-bindings="Fail">
      <tp:docstring>
        Signal an unrecoverable error for this stream, and remove it. If all
        streams are removed from a content, then it will also be removed.
      </tp:docstring>
      <arg direction="in" name="Reason" type="(uuss)"
        tp:type="Call_State_Reason">
        <tp:docstring>
          A structured reason for stream removal.
        </tp:docstring>
      </arg>
    </method>
  </interface>
</node>
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