diff options
Diffstat (limited to 'qt4/spec/Media_Stream_Handler.xml')
| -rw-r--r-- | qt4/spec/Media_Stream_Handler.xml | 725 |
1 files changed, 0 insertions, 725 deletions
diff --git a/qt4/spec/Media_Stream_Handler.xml b/qt4/spec/Media_Stream_Handler.xml deleted file mode 100644 index 123ea8be7..000000000 --- a/qt4/spec/Media_Stream_Handler.xml +++ /dev/null @@ -1,725 +0,0 @@ -<?xml version="1.0" ?> -<node name="/Media_Stream_Handler" xmlns:tp="http://telepathy.freedesktop.org/wiki/DbusSpec#extensions-v0"> - <tp:copyright> Copyright (C) 2005-2008 Collabora Limited </tp:copyright> - <tp:copyright> Copyright (C) 2005-2008 Nokia Corporation </tp:copyright> - <tp:copyright> Copyright (C) 2006 INdT </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.Media.StreamHandler"> - - <tp:struct name="Media_Stream_Handler_Candidate" - array-name="Media_Stream_Handler_Candidate_List"> - <tp:member type="s" name="Name"/> - <tp:member type="a(usuussduss)" name="Transports" - tp:type="Media_Stream_Handler_Transport[]"/> - </tp:struct> - - <tp:struct name="Media_Stream_Handler_Transport" - array-name="Media_Stream_Handler_Transport_List"> - <tp:member type="u" name="Component_Number"/> - <tp:member type="s" name="IP_Address"/> - <tp:member type="u" name="Port"/> - <tp:member type="u" tp:type="Media_Stream_Base_Proto" name="Protocol"/> - <tp:member type="s" name="Subtype"/> - <tp:member type="s" name="Profile"/> - <tp:member type="d" name="Preference_Value"/> - <tp:member type="u" tp:type="Media_Stream_Transport_Type" - name="Transport_Type"/> - <tp:member type="s" name="Username"/> - <tp:member type="s" name="Password"/> - </tp:struct> - - <tp:struct name="Media_Stream_Handler_Codec" - array-name="Media_Stream_Handler_Codec_List"> - <tp:docstring> - Information about a codec supported by a client or a peer's client. - </tp:docstring> - - <tp:member type="u" name="Codec_ID"> - <tp:docstring> - The codec's payload identifier, as per RFC 3551 (static or dynamic) - </tp:docstring> - </tp:member> - <tp:member type="s" name="Name"> - <tp:docstring>The codec's name</tp:docstring> - </tp:member> - <tp:member type="u" tp:type="Media_Stream_Type" name="Media_Type"> - <tp:docstring>Type of stream this codec supports</tp:docstring> - </tp:member> - <tp:member type="u" name="Clock_Rate"> - <tp:docstring>Sampling frequency in Hertz</tp:docstring> - </tp:member> - <tp:member type="u" name="Number_Of_Channels"> - <tp:docstring>Number of supported channels</tp:docstring> - </tp:member> - <tp:member type="a{ss}" name="Parameters" tp:type="String_String_Map"> - <tp:docstring>Codec-specific optional parameters</tp:docstring> - </tp:member> - </tp:struct> - - <property name="STUNServers" tp:name-for-bindings="STUN_Servers" - type="a(sq)" tp:type="Socket_Address_IP[]" access="read"> - <tp:added version="0.17.22"/> - <tp:docstring> - The IP addresses of possible STUN servers to use for NAT traversal, as - dotted-quad IPv4 address literals or RFC2373 IPv6 address literals. - This property cannot change once the stream has been created, so there - is no change notification. The IP addresses MUST NOT be given as DNS - hostnames. - - <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="CreatedLocally" tp:name-for-bindings="Created_Locally" - type="b" access="read"> - <tp:added version="0.17.22"/> - <tp:docstring> - True if we were the creator of this stream, false otherwise. - <tp:rationale> - This information is needed for some nat traversal mechanisms, such - as ICE-UDP, where the creator gets the role of the controlling agent. - </tp:rationale> - </tp:docstring> - </property> - - <property name="NATTraversal" tp:name-for-bindings="NAT_Traversal" - type="s" access="read"> - <tp:added version="0.17.22"/> - <tp:docstring xmlns="http://www.w3.org/1999/xhtml"> - <p>The transport (NAT traversal technique) to be used for this - stream. Well-known values include:</p> - - <dl> - <dt>none</dt> - <dd>Raw UDP, with or without STUN, should be used. If the - <tp:member-ref>STUNServers</tp:member-ref> property is non-empty, - STUN SHOULD be used.</dd> - - <dt>stun</dt> - <dd>A deprecated synonym for 'none'.</dd> - - <dt>gtalk-p2p</dt> - <dd>Google Talk peer-to-peer connectivity establishment should be - used, as implemented in libjingle 0.3.</dd> - - <dt>ice-udp</dt> - <dd>Interactive Connectivity Establishment should be used, - as defined by the IETF MMUSIC working group.</dd> - - <dt>wlm-8.5</dt> - <dd>The transport used by Windows Live Messenger 8.5 or later, - which resembles ICE draft 6, should be used.</dd> - - <dt>wlm-2009</dt> - <dd>The transport used by Windows Live Messenger 2009 or later, - which resembles ICE draft 19, should be used.</dd> - </dl> - - <p>This property cannot change once the stream has been created, so - there is no change notification.</p> - </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. 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>NATTraversal</tp:member-ref> property: if - NATTraversal is <code>ice-udp</code>, <code>tls</code> means - TLS over TCP as referenced by ICE draft 19, and if - NATTraversal is <code>gtalk-p2p</code>, <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>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>NATTraversal</tp:member-ref> - property. If NATTraversal is ice-udp, this is a TURN server; - if NATTraversal 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>This property cannot change once the stream has been created, so - there is no change notification.</p> - </tp:docstring> - </property> - - <signal name="AddRemoteCandidate" - tp:name-for-bindings="Add_Remote_Candidate"> - <arg name="Candidate_ID" type="s"> - <tp:docstring> - String identifier for this candidate - </tp:docstring> - </arg> - <arg name="Transports" type="a(usuussduss)" - tp:type="Media_Stream_Handler_Transport[]"> - <tp:docstring> - Array of transports for this candidate with fields, - as defined in NewNativeCandidate - </tp:docstring> - </arg> - <tp:docstring> - Signal emitted when the connection manager wishes to inform the - client of a new remote candidate. - </tp:docstring> - </signal> - <signal name="Close" tp:name-for-bindings="Close"> - <tp:docstring> - Signal emitted when the connection manager wishes the stream to be - closed. - </tp:docstring> - </signal> - <method name="CodecChoice" tp:name-for-bindings="Codec_Choice"> - <arg direction="in" name="Codec_ID" type="u"/> - <tp:docstring> - Inform the connection manager of codec used to receive data. - </tp:docstring> - </method> - <method name="Error" tp:name-for-bindings="Error"> - <arg direction="in" name="Error_Code" type="u" tp:type="Media_Stream_Error"> - <tp:docstring> - ID of error, from the MediaStreamError enumeration - </tp:docstring> - </arg> - <arg direction="in" name="Message" type="s"> - <tp:docstring> - String describing the error - </tp:docstring> - </arg> - <tp:docstring> - Inform the connection manager that an error occured in this stream. The - connection manager should emit the StreamError signal for the stream on - the relevant channel, and remove the stream from the session. - </tp:docstring> - </method> - <tp:enum name="Media_Stream_Error" type="u"> - <tp:enumvalue suffix="Unknown" value="0"> - <tp:docstring> - An unknown error occured. - </tp:docstring> - </tp:enumvalue> - <tp:enumvalue suffix="EOS" value="1"> - <tp:docstring> - The end of the stream was reached. - </tp:docstring> - <tp:deprecated version="0.17.27"> - This error has no use anywhere. In Farsight 1 times, it was used to - indicate a GStreamer EOS (when the end of a file is reached). But - since this is for live calls, it makes no sense. - </tp:deprecated> - </tp:enumvalue> - <tp:enumvalue suffix="Codec_Negotiation_Failed" value="2"> - <tp:added version="0.17.27"/> - <tp:docstring> - There are no common codecs between the local side - and the other particpants in the call. The possible codecs are not - signalled here: the streaming implementation is assumed to report - them in an implementation-dependent way, e.g. Farsight should use - GstMissingElement. - </tp:docstring> - </tp:enumvalue> - <tp:enumvalue suffix="Connection_Failed" value="3"> - <tp:added version="0.17.27"/> - <tp:docstring> - A network connection for the Media could not be established or was - lost. - </tp:docstring> - </tp:enumvalue> - <tp:enumvalue suffix="Network_Error" value="4"> - <tp:added version="0.17.27"/> - <tp:docstring> - There was an error in the networking stack - (other than the connection failure). - </tp:docstring> - </tp:enumvalue> - <tp:enumvalue suffix="No_Codecs" value="5"> - <tp:added version="0.17.27"/> - <tp:docstring> - There are no installed codecs for this media type. - </tp:docstring> - </tp:enumvalue> - <tp:enumvalue suffix="Invalid_CM_Behavior" value="6"> - <tp:added version="0.17.27"/> - <tp:docstring> - The CM is doing something wrong. - </tp:docstring> - </tp:enumvalue> - <tp:enumvalue suffix="Media_Error" value="7"> - <tp:added version="0.17.27"/> - <tp:docstring> - There was an error in the media processing stack. - </tp:docstring> - </tp:enumvalue> - </tp:enum> - <method name="NativeCandidatesPrepared" - tp:name-for-bindings="Native_Candidates_Prepared"> - <tp:docstring> - Informs the connection manager that all possible native candisates - have been discovered for the moment. - </tp:docstring> - </method> - <method name="NewActiveCandidatePair" - tp:name-for-bindings="New_Active_Candidate_Pair"> - <arg direction="in" name="Native_Candidate_ID" type="s"/> - <arg direction="in" name="Remote_Candidate_ID" type="s"/> - <tp:docstring> - Informs the connection manager that a valid candidate pair - has been discovered and streaming is in progress. - </tp:docstring> - </method> - <method name="NewActiveTransportPair" - tp:name-for-bindings="New_Active_Transport_Pair"> - <arg direction="in" name="Native_Candidate_ID" type="s"/> - <arg direction="in" name="Native_Transport" type="(usuussduss)" - tp:type="Media_Stream_Handler_Transport"/> - <arg direction="in" name="Remote_Candidate_ID" type="s"/> - <arg direction="in" name="Remote_Transport" type="(usuussduss)" - tp:type="Media_Stream_Handler_Transport"/> - <tp:docstring> - <p>Informs the connection manager that a valid transport pair - has been discovered and streaming is in progress. Component - id MUST be the same for both transports and the pair is - only valid for that component.</p> - - <tp:rationale> - <p>The connection manager might need to send the details of - the active transport pair (e.g. c and o parameters of SDP - body need to contain address of selected native RTP transport - as stipulated by RFC 5245). However, the candidate ID might - not be enough to determine these info if the transport was - found after <tp:member-ref>NativeCandidatesPrepared</tp:member-ref> - has been called (e.g. peer reflexive ICE candidate). </p> - </tp:rationale> - - <p>This method must be called before - <tp:member-ref>NewActiveCandidatePair</tp:member-ref>.</p> - - <tp:rationale> - <p>This way, connection managers supporting this method can - safely ignore subsequent - <tp:member-ref>NewActiveCandidatePair</tp:member-ref> call.</p> - </tp:rationale> - - <p>Connection managers SHOULD NOT implement this method unless - they need to inform the peer about selected transports. As a - result, streaming implementations MUST NOT treat errors raised - by this method as fatal.</p> - - <tp:rationale> - <p>Usually, connection managers only need to do one answer/offer - round-trip. However, some protocols give the possibility to - to send an updated offer (e.g. ICE defines such mechanism to - avoid some race conditions and to properly set the state of - gateway devices).</p> - </tp:rationale> - </tp:docstring> - </method> - <tp:enum name="Media_Stream_Base_Proto" type="u"> - <tp:enumvalue suffix="UDP" value="0"> - <tp:docstring>UDP (User Datagram Protocol)</tp:docstring> - </tp:enumvalue> - <tp:enumvalue suffix="TCP" value="1"> - <tp:docstring>TCP (Transmission Control Protocol)</tp:docstring> - </tp:enumvalue> - </tp:enum> - <method name="NewNativeCandidate" - tp:name-for-bindings="New_Native_Candidate"> - <arg direction="in" name="Candidate_ID" type="s"> - <tp:docstring> - String identifier for this candidate - </tp:docstring> - </arg> - <arg direction="in" name="Transports" type="a(usuussduss)" - tp:type="Media_Stream_Handler_Transport[]"> - <tp:docstring xmlns="http://www.w3.org/1999/xhtml"> - Array of transports for this candidate, with fields: - <ul> - <li>component number</li> - <li>IP address (as a string)</li> - <li>port</li> - <li>base network protocol (one of the values of MediaStreamBaseProto)</li> - <li>proto subtype (e.g. RTP)</li> - <li>proto profile (e.g. AVP)</li> - <li>our preference value of this transport (double in range 0.0-1.0 - inclusive); 1 signals the most preferred transport</li> - <li>transport type, one of the values of MediaStreamTransportType</li> - <li>username if authentication is required</li> - <li>password if authentication is required</li> - </ul> - </tp:docstring> - </arg> - <tp:docstring> - Inform this MediaStreamHandler that a new native transport candidate - has been ascertained. - </tp:docstring> - </method> - <tp:enum name="Media_Stream_Transport_Type" type="u"> - <tp:enumvalue suffix="Local" value="0"> - <tp:docstring> - A local address - </tp:docstring> - </tp:enumvalue> - <tp:enumvalue suffix="Derived" value="1"> - <tp:docstring> - An external address derived by a method such as STUN - </tp:docstring> - </tp:enumvalue> - <tp:enumvalue suffix="Relay" value="2"> - <tp:docstring> - An external stream relay - </tp:docstring> - </tp:enumvalue> - </tp:enum> - <method name="Ready" tp:name-for-bindings="Ready"> - <arg direction="in" name="Codecs" type="a(usuuua{ss})" - tp:type="Media_Stream_Handler_Codec[]"> - <tp:docstring> - Locally-supported codecs. - </tp:docstring> - </arg> - <tp:docstring> - Inform the connection manager that a client is ready to handle - this StreamHandler. Also provide it with info about all supported - codecs. - </tp:docstring> - </method> - <method name="SetLocalCodecs" tp:name-for-bindings="Set_Local_Codecs"> - <arg name="Codecs" type="a(usuuua{ss})" direction="in" - tp:type="Media_Stream_Handler_Codec[]"> - <tp:docstring> - Locally-supported codecs - </tp:docstring> - </arg> - <tp:docstring xmlns="http://www.w3.org/1999/xhtml"> - <p>Used to provide codecs after Ready(), so the media client can go - ready for an incoming call and exchange candidates/codecs before - knowing what local codecs are available.</p> - - <p>This is useful for gatewaying calls between two connection managers. - Given an incoming call, you need to call - <tp:member-ref>Ready</tp:member-ref> to get the remote codecs before - you can use them as the "local" codecs to place the outgoing call, - and hence receive the outgoing call's remote codecs to use as the - incoming call's "local" codecs.</p> - - <p>In this situation, you would pass an empty list of codecs to the - incoming call's Ready method, then later call SetLocalCodecs on the - incoming call in order to respond to the offer.</p> - </tp:docstring> - </method> - <signal name="RemoveRemoteCandidate" - tp:name-for-bindings="Remove_Remote_Candidate"> - <arg name="Candidate_ID" type="s"> - <tp:docstring> - String identifier for remote candidate to drop - </tp:docstring> - </arg> - <tp:deprecated version="0.17.18"> - There is no case where you want to release candidates (except - for an ICE reset, and there you'd want to replace then all, - using <tp:member-ref>SetRemoteCandidateList</tp:member-ref>). - </tp:deprecated> - <tp:docstring> - Signal emitted when the connection manager wishes to inform the - client that the remote end has removed a previously usable - candidate. - - <tp:rationale> - It seemed like a good idea at the time, but wasn't. - </tp:rationale> - </tp:docstring> - </signal> - <signal name="SetActiveCandidatePair" - tp:name-for-bindings="Set_Active_Candidate_Pair"> - <arg name="Native_Candidate_ID" type="s"/> - <arg name="Remote_Candidate_ID" type="s"/> - <tp:docstring> - Emitted by the connection manager to inform the client that a - valid candidate pair has been discovered by the remote end - and streaming is in progress. - </tp:docstring> - </signal> - <signal name="SetRemoteCandidateList" - tp:name-for-bindings="Set_Remote_Candidate_List"> - <arg name="Remote_Candidates" type="a(sa(usuussduss))" - tp:type="Media_Stream_Handler_Candidate[]"> - <tp:docstring> - A list of candidate id and a list of transports - as defined in NewNativeCandidate - </tp:docstring> - </arg> - <tp:docstring> - Signal emitted when the connection manager wishes to inform the - client of all the available remote candidates at once. - </tp:docstring> - </signal> - <signal name="SetRemoteCodecs" tp:name-for-bindings="Set_Remote_Codecs"> - <arg name="Codecs" type="a(usuuua{ss})" - tp:type="Media_Stream_Handler_Codec[]"> - <tp:docstring> - Codecs supported by the remote peer. - </tp:docstring> - </arg> - <tp:docstring> - Signal emitted when the connection manager wishes to inform the - client of the codecs supported by the remote end. - If these codecs are compatible with the remote codecs, then the client - must call <tp:member-ref>SupportedCodecs</tp:member-ref>, - otherwise call <tp:member-ref>Error</tp:member-ref>. - </tp:docstring> - </signal> - <signal name="SetStreamPlaying" tp:name-for-bindings="Set_Stream_Playing"> - <arg name="Playing" type="b"/> - <tp:docstring> - If emitted with argument TRUE, this means that the connection manager - wishes to set the stream playing; this means that the streaming - implementation should expect to receive data. If emitted with argument - FALSE this signal is basically meaningless and should be ignored. - - <tp:rationale> - We're very sorry. - </tp:rationale> - </tp:docstring> - </signal> - <signal name="SetStreamSending" tp:name-for-bindings="Set_Stream_Sending"> - <arg name="Sending" type="b"/> - <tp:docstring> - Signal emitted when the connection manager wishes to set whether or not - the stream sends to the remote end. - </tp:docstring> - </signal> - <signal name="StartTelephonyEvent" - tp:name-for-bindings="Start_Telephony_Event"> - <arg name="Event" type="y" tp:type="DTMF_Event"> - <tp:docstring> - A telephony event code. - </tp:docstring> - </arg> - <tp:docstring> - Request that a telephony event (as defined by RFC 4733) is transmitted - over this stream until StopTelephonyEvent is called. - </tp:docstring> - </signal> - <signal name="StartNamedTelephonyEvent" - tp:name-for-bindings="Start_Named_Telephony_Event"> - <tp:added version="0.21.2"/> - <arg name="Event" type="y" tp:type="DTMF_Event"> - <tp:docstring> - A telephony event code as defined by RFC 4733. - </tp:docstring> - </arg> - <arg name="Codec_ID" type="u"> - <tp:docstring> - The payload type to use when sending events. The value 0xFFFFFFFF - means to send with the already configured event type instead of using - the specified one. - </tp:docstring> - </arg> - <tp:docstring> - Request that a telephony event (as defined by RFC 4733) is transmitted - over this stream until StopTelephonyEvent is called. This differs from - StartTelephonyEvent in that you force the event to be transmitted - as a RFC 4733 named event, not as sound. You can also force a specific - Codec ID. - </tp:docstring> - </signal> - <signal name="StartSoundTelephonyEvent" - tp:name-for-bindings="Start_Sound_Telephony_Event"> - <tp:added version="0.21.2"/> - <arg name="Event" type="y" tp:type="DTMF_Event"> - <tp:docstring> - A telephony event code as defined by RFC 4733. - </tp:docstring> - </arg> - <tp:docstring> - Request that a telephony event (as defined by RFC 4733) is transmitted - over this stream until StopTelephonyEvent is called. This differs from - StartTelephonyEvent in that you force the event to be transmitted - as sound instead of as a named event. - </tp:docstring> - </signal> - <signal name="StopTelephonyEvent" - tp:name-for-bindings="Stop_Telephony_Event"> - <tp:docstring> - Request that any ongoing telephony events (as defined by RFC 4733) - being transmitted over this stream are stopped. - </tp:docstring> - </signal> - <method name="StreamState" tp:name-for-bindings="Stream_State"> - <arg direction="in" name="State" type="u" tp:type="Media_Stream_State"/> - <tp:docstring> - Informs the connection manager of the stream's current state, as - as specified in Channel.Type.StreamedMedia::ListStreams. - </tp:docstring> - </method> - - <method name="SupportedCodecs" tp:name-for-bindings="Supported_Codecs"> - <arg direction="in" name="Codecs" type="a(usuuua{ss})" - tp:type="Media_Stream_Handler_Codec[]"> - <tp:docstring> - Locally supported codecs. - </tp:docstring> - </arg> - <tp:docstring> - Inform the connection manager of the supported codecs for this session. - This is called after the connection manager has emitted SetRemoteCodecs - to notify what codecs are supported by the peer, and will thus be an - intersection of all locally supported codecs (passed to Ready) - and those supported by the peer. - </tp:docstring> - </method> - - <method name="CodecsUpdated" tp:name-for-bindings="Codecs_Updated"> - <arg direction="in" name="Codecs" type="a(usuuua{ss})" - tp:type="Media_Stream_Handler_Codec[]"> - <tp:docstring> - Locally supported codecs, which SHOULD be the same as were previously - in effect, but possibly with different parameters. - </tp:docstring> - </arg> - <tp:docstring> - Inform the connection manager that the parameters of the supported - codecs for this session have changed. The connection manager should - send the new parameters to the remote contact. - - <tp:rationale> - This is required for H.264 and Theora, for example. - </tp:rationale> - </tp:docstring> - </method> - - <signal name="SetStreamHeld" tp:name-for-bindings="Set_Stream_Held"> - <tp:docstring xmlns="http://www.w3.org/1999/xhtml"> - <p>Emitted when the connection manager wishes to place the stream on - hold (so the streaming client should free hardware or software - resources) or take the stream off hold (so the streaming client - should reacquire the necessary resources).</p> - - <p>When placing a channel's streams on hold, the connection manager - SHOULD notify the remote contact that this will be done (if - appropriate in the protocol) before it emits this signal.</p> - - <tp:rationale> - <p>It is assumed that relinquishing a resource will not fail. - If it does, the call is probably doomed anyway.</p> - </tp:rationale> - - <p>When unholding a channel's streams, the connection manager - SHOULD emit this signal and wait for success to be indicated - via HoldState before it notifies the remote contact that the - channel has been taken off hold.</p> - - <tp:rationale> - <p>This means that if a resource is unavailable, the remote - contact will never even be told that we tried to acquire it.</p> - </tp:rationale> - </tp:docstring> - <tp:added version="0.17.3"/> - - <arg name="Held" type="b"> - <tp:docstring> - If true, the stream is to be placed on hold. - </tp:docstring> - </arg> - </signal> - - <method name="HoldState" tp:name-for-bindings="Hold_State"> - <tp:docstring> - Notify the connection manager that the stream's hold state has - been changed successfully in response to SetStreamHeld. - </tp:docstring> - <tp:added version="0.17.3"/> - <arg direction="in" name="Held" type="b"> - <tp:docstring> - If true, the stream is now on hold. - </tp:docstring> - </arg> - </method> - - <method name="UnholdFailure" tp:name-for-bindings="Unhold_Failure"> - <tp:docstring> - Notify the connection manager that an attempt to reacquire the - necessary hardware or software resources to unhold the stream, - in response to SetStreamHeld, has failed. - </tp:docstring> - <tp:added version="0.17.3"/> - </method> - - <tp:docstring> - Handles signalling the information pertaining to a specific media stream. - A client should provide information to this handler as and when it is - available. - </tp:docstring> - </interface> -</node> -<!-- vim:set sw=2 sts=2 et ft=xml: --> |