diff options
| -rw-r--r-- | TODO | 83 |
1 files changed, 4 insertions, 79 deletions
@@ -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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