SIP SIMPLE Client SDK

[[TOC(gui_progress, depth=3)]]

= Audio sessions =

 1. ~~ Incoming sessions do not display '''session information''' and the control buttons are missing~~

  1. Not getting a SessionDidStart notification from middleware on incoming sessions

 1. ~~ When '''rejecting a session''' it creates an entry in the drawer~~

 1. '''Cancel''' is not handled for incoming sessions, the pop-up remains active 

 1. ~~'''Maximize drawer''' when it opens the first time, it seem to open only 80% now ~~

 1. Play '''ringtones'''

 1. ~~ A terminated session must '''disappear from the drawer''' automatically after 5 seconds ~~

 1. ~~ Remaining sessions must '''shuffle to the top''' when a session ends~~

 1. ~~ Toggle hold does not change hold '''button color''' ~~

 1. ~~ Update session audio information with '''Hold by remote''' when held by remote party ~~

 1. ~~ Only one audio session can be '''active at a time''', all other existing audio sessions must be put on hold~~

 1. ~~ Highlight the '''active session''' with a bold border, click on a session to select it as the active session~~

 1. When the active session is selected, what is typed on the keyboard must be transmitted as '''DTMF tones'''

 1. ~~ When a call is connected move keyboard focus to it ~~

 1. ~~ When a session has ended, display it for '''5 more seconds''' then hide it and shuffle the deck  ~~

 1. ~~ Add an item '''Show audio sessions''' to the '''Session menu''' to show the drawer, if it was closed  (activated the View -> Toggle Sessions Drawer) ~~

 1. Remove the '''i''' icon, display a black triangle instead (black triangle?). Maybe is beter to just click on audio info to get a pop up with more information?

 1. ~~ '''Scroll bar''' appears as result of multiple sessions, but it does not disappear when the number of session decreases~~

 1. ~~ '''Hangup All''' is not implemented~~

 1. Display icon with lock when sRTP is active

 1. Disable Hangup All button if no session is active

 1. Disable Conference button if less than two sessions active

= History sessions =

 1. Use the same session drawer to display previous sessions

 1. Keep same layout as active session with the following changes:

   1. Instead of audio session information display the negotiated streams (Audio|Chat|Audio+Chat etc...)

   1. Instead of duration display '''YYYY-MM-DD HH:MM  Duration: HH:MM:SS''' 

   1. Add a button to call back in place of current session buttons

 1. Add a Session menu item to toggle the drawer content between active session and history sessions

 1. Show most recent sessions on top

= Conferencing =

Only one conference is possible to make the interface easy to use.

To start a conference: 

 1. Send a re-INVITE to a) take each call off hold and b) set the is-focus attribute of the Contact header

 1. Set the conference bridge to mix audio between all parties

 1. Disable the hold buttons, they do not operate while in conference

 1. Draw the selection rectangle around the sessions part of the conference

To stop a conference

 1. Send a re-INVITE to a) take each call an leave only one off hold and b) disable is-focus attribute of the Contact header

 1. Set the conference bridge to stop mixing audio

 1. Enable the hold buttons

 1. Draw the selection rectangle around the selected session

New sessions while in a conference

 1. New sessions are added bellow the conference

 1. A session can be dragged in and out of the conference

 1. When switching from a conference to another single session, do not use hold

= Chat sessions =

 1. '''Close chat''' button does not close the window

 1. '''No BYE''' is sent when session ends

 1. Append chat messages to '''chat.history_directory/sip_account/YYYYMMDD-recipient.txt'''

 1. For each message that appears in the history window:

  1. Display a rounded thin rectangle with the '''recipient''' on the left and '''full date''' on the right side of the text in history with either light blue or grey background depending on which party is typing it, display the full FROM headers

  1. Then display the content of the message without a rectangle

 1. Play an audible message when a message is received, based on '''silent''' and '''general.message_received_sound''' settings

 1. Play an audible message when a message is sent, based on silent and '''general.message_sent_sound''' settings

 1. '''Throttle playback''' of audible notifications to maximum one every 3 seconds 

 1. Display the number of '''un-read messages''' in non-visible tabs in a red circle over the tab name

 1. Display a status bar on the bottom of the window and '''show the connection information''' in it instead of the main window

 1. When dragging a recipient tab outside of the window '''spawn a new chat window''' and move the session into it

 1. When an incoming request has '''chat only''' and the from header is in the contact list, automatically accept the session

 1. Make horizontal delimiter drag-able

 1. When start typing send '''is-composing'''. This requires implementing the is-composing payload parser, see http://sipsimpleclient.com/ticket/40. If user has deleted the typed text before sending it update is-composing. 

 1. Add a + button (add contact) in the window toolbar if the buddy does not exit

= Main interface =

 1. When selecting a different account, mark it as '''default''' using the settings api

 1. '''Remove account''' from list when is disabled in preferences  

 1. On startup, '''select the default account''' in the account list

 1. The SIP accounts are '''not de-registered''' on quit

 1. Add a '''minimize button''' to collapse the buddy list

 1. Hide the Add button + sign when the interface is collapsed

 1. Mute button is not implemented 

 1. If red button is pressed the main window disappears and is no option to bring it back

 1. No dock icon available

= Contacts =

 1. When add contact without domain part always append curent domain

 1. '''Delete the contact''' when pressing delete

 1. After dialing a found contact, display back the buddy list

 1. Show search results from the system address book

 1. Add right click for contact:

  1. Start audio session

  1. Start chat session

  1. Transfer file

  1. Offer local desktop

  1. Request remote desktop

  1. Edit contact

  1. Delete contact

= Preferences =

 1. '''Change audio devices''' does not enter effect unless rebooting, call set_devices() to activate the new settings

 1. '''Stun server''' addresses cannot be set, they seem to inherit the results found in DNS  

 1. '''general.rtp.local_ip''': Can't set option local_ip illegal local IP address value: auto

 1. '''general.rtp.port_range''': (AttributeError: 'PortRangeOption' object has no attribute 'save')

 1. '''general.sip.transports''' cannot be saved

 1. '''general.audio.codec_list''' cannot be saved and displays duplicate codecs

 1. '''account.audio.codec_list''' cannot be saved and displays duplicate codecs

 1. '''account.msrp.relay''' cannot be set: global name 'MSRPRelayAddres' is not defined

 1. When '''disable account''' in preferences, it does not get removed from the main interface account lists

 1. Properly '''align''' horizontally the Advanced settings

 1. Display text to the '''left of the checkbox''' instead of the right

 1. '''Delete account Enabled checkbox''', leave only the checkbox in the account list

 1. '''De-register''' the account and remove it from the account list in main interface when disable the account

 1. When '''enable an account''', select it in the main interface account list

 1. Add a '''Reset option''' to reset advanced settings to default values

 1. Display the '''bonjour account''' always at the end of the list

 1. '''bonjour account''' has only Display name as main property.  There is not sip address or password

 1. Cannot '''delete Bonjour''' account: (TypeError: cannot delete BonjourAccount instance with default id). Do  not allow the user to attempt to delete it. Bonjour can only be enabled/disabled but not deleted.

 1. For '''account.ringtone''' advanced setting, the default setting must be inherited from '''general.ringtone.inbound'''

 1. '''general.message_received_sound''' does not fit in window

 1. Allow ordering of accounts in the account list

 1. While clicking on account I got this (cannot reproduce): (TypeError: 'NSAutoreleasePool' object is not iterable)

 1. Check if the wav file is in the right format before saving the settings related to ringtones

 1. Check if the TLS options set by the user are valid before saving them (by loading them outside the engine using gnutls library)

Display SIP registration state in '''account.advanced.registration''' section.

= Engine =

 1. Set User Agent name to blink-version

 1. If '''engine is dead''', the message  '''no sip account is active''' is displayed but is unclear what the real reason is. An engine stop must be logged to general debug window. Not clear how to restart the engine, maybe we should restart the application until a good solution is found

 1. If the engine has stopped (because of some crash) the preferences do not show anymore:

{{{

Blink[26992:10b] <type 'exceptions.AttributeError'>: 'Engine' object has no attribute 'output_devices'

}}}

By setting the wrong TLS certificate files, the middleware does not start anymore:

{{{

[Session started at 2009-07-19 10:37:39 +0200.]

Traceback (most recent call last):

  File "/Users/adigeo/work/blink/build/Debug/Blink.app/Contents/Resources/BlinkAppDelegate.py", line 40,

   in applicationDidFinishLaunching_

    self.backend.init(options)

  File "/Users/adigeo/work/blink/build/Debug/Blink.app/Contents/Resources/SIPManager.py", line 108, in init

    self._app.start(config_backend=config_be)

  File "/Library/Python/2.5/site-packages/sipsimple/api.py", line 85, in start

    trace_sip=settings.logging.trace_sip,

  File "/Library/Python/2.5/site-packages/sipsimple/engine.py", line 75, in start

    self._ua = PJSIPUA(self._handle_event, 1.1.init_options)

  File "core.ua.pxi", line 66, in sipsimple.core.PJSIPUA.__init__ (sipsimple/core.c:23189)

  File "core.lib.pxi", line 106, in sipsimple.core.PJSIPEndpoint.__cinit__ (sipsimple/core.c:6231)

  File "core.lib.pxi", line 167, in sipsimple.core.PJSIPEndpoint._start_tls_transport (sipsimple/core.c:6884)

PJSIPError: Could not create TLS transport: Error loading/verifying SSL CA list file (PJSIP_TLS_ECACERT)

None

}}}

= Presence bar =

Make it as high as the contacts (two rows high).

 1. Show '''own photo''' to the left, align photo and size with buddy list photos

 1. Show '''Display name''' next to the photo on top row

 1. Show '''Presence activity''' next to name on top row

 1. Show '''Presence activity note''' on second row, editable text

= Debug window =

 1. Display all traces in a '''debug window''' with tabs for each debug type

 1. Add a '''General messages''' window to the Debug menu, first item

= GUI settings =

Remember the following settings between restarts:

 1. '''Debug window''' and the selected tab

 1. Status of '''contact groups''' - colapsed/expanded and their order

 1. SIP '''accounts order''' list

= Middleware =

 1. On startup, check if the audio device from the preferences '''is available''' in the detected devices list. If not, set the device to the system default device and display a warning to the user to set the devices in the preferences

 1. Check the validity of the '''TLS''' settings/files before starting the engine

 1. It seems that the '''stun servers''' are learned from the DNS and cannot be overwritten in settings, they show up automatically

 1. If I set the relay to abcd.com standard '''port 2855''' must be used, now is set to zero: relay = abcd.com:0;transport=tls

 1. Append '''dummy''' to the lists of available audio devices

 1. We need a solution to accept a session without a 180- ringing. For instance we can have automatic answer based on various preferences.

 1. Move '''general.audio.codec_list''' to '''general.rtp.audio_codecs'''

 1. Move '''account.audio.codec_list''' to '''account.rtp.audio_codecs'''

 1. Move '''account.audio.srtp_encryption''' to '''account.rtp.srtp_encryption'''

 1. Delete '''general.audio.playback_dtmf''' setting

 1. Add '''general.audio.mute''' setting

 1. Delete '''account.audio''' section

 1. Add '''account.sip''' section

 1. Move '''account.outbound_proxy''' to '''account.sip.outbound_proxy'''

 1. Move '''account.presence.publish_interval''' to '''account.sip.publish_interval'''

 1. Move '''account.presence.subscribe_interval''' to '''account.sip.subscribe_interval'''

 1. Move '''account.registration.interval''' to '''account.sip.register_interval'''

 1. Delete '''account.ice.use_stun''', it is always true

 1. Rename '''account.ice''' to '''account.nat_traversal'''

 1. Rename '''account.ice.enabled''' to '''account.nat_traversal.enable_ice'''

 1. Delete global.chat.accept_types

 1. Delete global.chat.accept_wrapped_types