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===============================================================
TODO / telepathy-sofiasip
===============================================================
Feature Roadmap
---------------
- re-registration on network change detection
- re-offer media streams on network change detection (handover)
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)
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)
- 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
--------------
- if multiple network interfaces are present, the wrong IP address
may be offered in a c= SDP media line.
- prevent an endless authentication loop when the server responds
with 401 or 407 repeatedly.
- verify correct operation with 100rel/PRACK
- verify correct operation with a remote node utilizing
early media
- implement code for removing group members
- see sip-media-channel.c:priv_media_channel_remove_member()
- implement StreamHandler.CodecChoice()
Presence and messaging
-----------------------
- check that content type of receivng messages is "text/plain" before
forwarding; or parse the MIME payload and try to find a "text/plain" part
- send back an error code for unsupported message body types
- make sure that body character sets other than ASCII and UTF-8 are supported
or at least detected
- implement Connection.Interface.Presense.AddStatus()
- implement Connection.Interface.Presense.SetStatus()
- implement Connection.Interface.Presense.RemoveStatus()
- implement Connection.Interface.Presense.ClearStatus()
- implement Connection.Interface.Presense.GetStatuses()
- implement Connection.Interface.Presense.RequestContactInfo()
- implement Connection.Interface.Presense.SetLastActivityTime()
- 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
successful.
Test programs
-------------
- tp_caller is an ugly hack, major refactoring needed
- unit tests for basic functionality (creation and removal of
conn.mgr etc objects, registration, calling to itself, etc)
Plugin interface
----------------
- mechanism to dynamically load handlers for new types of
channels, and/or new types of connections
General
-------
- various XXX items in the source codes (a generic todo item)
- status 2006-11-26: 37 XXXs
- status 2006-12-04: 36 XXXs
- 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
|
