Purge done and outdated TODO items

20070507131533-5b6ca-17d26682459d4eba0570aeda43f553de2a919a3a.gz

1 files changed, 4 insertions, 79 deletions

diff --git a/TODO b/TODO
index e4984cb..e864267 100644
--- a/TODO
+++ b/TODO
@@ -11,37 +11,22 @@ Feature Roadmap
 Critical todo items
 -------------------
 
-- when making outbound sessions with multiple media, only 
-  first media is succesfully set to playing state
-    - signals are emitted correctly, but they do not seem to have
-      the correct effect
-- DONE: (works for me) segfault handling an offer that has fewer media
-  than locally available (audio offer, when audio+video locally available)
+empty
 
 Account settings
 ----------------
 
-- note: requires modifications to data/sofiasip.manager, sip-connection.c
-  as well as to sip-connection-manager.c
-- ability to toggle whether to modify local contact (discover binding)
-    - whether to use rport and/or STUN and re-register with the updated
-      contact
-- additional set of username, realm and password for authentication
-    - to authenticate to PSTN gateways, etc where registrations credentials
-      are not sufficient
-    - also needed if the service provider uses a separate username 
-      for authentication (different from user part of the public SIP address)
+- alternative username for registrar authentication if differs from
+  the URI username
 - ability to disable known difficult-to-implement features
     - early media with PRACK
 - ability to disable use of outbound proxy
     - any use cases for this?
-    - see sip-connection.c:sip_connection_connect()
 
 Connection management
 ---------------------
 
 - implement Connection.AdvertiseCapabilities()
-- check if we already have a connection to a requested account
 
 Media sessions
 --------------
@@ -72,7 +57,7 @@ Presence and messaging
 - implement Connection.Interface.Presense.GetStatuses()
 - implement Connection.Interface.Presense.RequestContactInfo()
 - implement Connection.Interface.Presense.SetLastActivityTime()
-- ConnectionInterfacePresence; RequestPresence:
+- (obsolete?) ConnectionInterfacePresence; RequestPresence:
     - Response to SUBSCRIBE initiated by nua_glib_subscribe() emits a signal
       subscribe-answered from nua_glib but there is no signal for
       telepathy-sofiasip client, i.e. client cannot be informed if subscribe was
@@ -100,64 +85,4 @@ General
     - status 2006-12-05: 24 XXXs 
     - status 2006-12-15: 20 XXXs 
     - status 2006-12-18: 19 XXXs
-
-Past todo items
----------------
-
-- DONE: un-REGISTER does not exit
-- DONE: unsuccessful REGISTER not handled correctly
-- DONE: 3rd message (sent or recvd) causes "Permission denied" because of bad handle code
-- DONE: 'message-sent' emitted after 200 OK
-- DONE: 'send-error' emitted if message delivery failed
-- DONE: "Permission denied" shown when starting a chat by updating tp-sofiasip dbus API
-- DONE: various XXX-SIPify items (code copied from telepathy-gabble but not yet 
-  converted to SIP) items around the codebase
-- DONE: all places marked with "#if 0" should be resolved
-- DONE: upgrade to tp-0.13 interfaces
-    - VoipEngine -> StreamEngine
-    - remaining FooHandle -> Foohandles changes
-- DONE: move from nua_glib to nua
-    - better API for extensions (new methods, custom headers, nua
-      identity, different presence usage scenarios, etc, etc)
-    - less maintenance (nua_glib+nua vs nua)...?
-- DONE: BYE is not correcly sent when Dbus connection dies
-    - it tries to send it, but process exits before BYE is completed
-    - segfault due to invalid handle
-        - 0x0804b8ab in cb_status_changed (conn=0x8082c38, data=0x2) at sip-connection-manager.c:70
-        - 70 g_hash_table_remove (connman->priv->connections, conn);
-- DONE: update handles code to use gheap.h (and not use quarks)
-- DONE: verify session cleanup (make a test case that repeatedly
-  creates and destroys media sessions)
-- DONE: test that signaling for local alerts works
-- DONE: the conn.mgr should parse the SDP and create a matching number 
-  of sip-media-stream instance (otherwise we get an assert 
-  from sip-media-session:sip_media_session_set_remote_info())
-- DONE: currently in auto-answer mode, should wait until client 
-  modifies the local_pending_members set
-- DONE: correct handling incomning call hold
-    - verified with 0.3.5 (remote client N80ie)
-- DONE: call that fails with a 404 response is not properly handled
-    - channel disconnected but no proper error given to the UI
-- DONE: add an option to stream-engine interface to select non-jingle mode
-  of operation
-- DONE: specifying keepalive method
-- DONE: setting to override first-hop transport selection
-    - "transport", with possible settings of "udp", "tcp", "tcp/tls", "auto"
-    - "proxy" setting has been replaced by "transport", "proxy-host", "port"
-- DONE: specifying keepalive method frequency
-- DONE: rename "contact" to "bind-url"
-    -> removed contact altogether, instead use "address", "proxy" and
-       "registrar" to determine the set of required transports and 
-       local sockets to activate
-- READY: account settings 
-    - ability to set all key connection parameters
-- READY: solid registrations
-    - login and logout initiated by TP UIs 
-    - ability to support multiple accounts 
-- READY: inbound and outbound audio calls
-    - interoperability with PSTN gateways and SIP compliant clients
-- READY: sending and receiving instant messages (SIP MESSAGE)
-- READY: outbound and inbound audio/video calls
-- READY: basic SIP presence (avail/not-avail)
-    - not real presence
   
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