diff options
Diffstat (limited to 'spec/Media_Stream_Handler.xml')
| -rw-r--r-- | spec/Media_Stream_Handler.xml | 89 |
1 files changed, 87 insertions, 2 deletions
diff --git a/spec/Media_Stream_Handler.xml b/spec/Media_Stream_Handler.xml index c9ae78f..123ea8b 100644 --- a/spec/Media_Stream_Handler.xml +++ b/spec/Media_Stream_Handler.xml @@ -337,6 +337,53 @@ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.</ 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> @@ -513,9 +560,9 @@ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.</ </signal> <signal name="StartTelephonyEvent" tp:name-for-bindings="Start_Telephony_Event"> - <arg name="Event" type="y"> + <arg name="Event" type="y" tp:type="DTMF_Event"> <tp:docstring> - A telephony event code as defined by RFC 4733. + A telephony event code. </tp:docstring> </arg> <tp:docstring> @@ -523,6 +570,44 @@ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.</ 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> |