SIP SIMPLE Client SDK

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

= Audio sessions =

 1. Incoming sessions to not display '''session information'''

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

 1. '''Cancel''' is not handled for incoming sessions

 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

 1. Remove the '''i''' icon, display a black triangle instead

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

 1. Draw a '''visible handle''' to pop up the session drawer

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

= Chat sessions =

 1. Cannot '''reject Chat''' session: (AttributeError: 'NoneType' object has no attribute 'setStream')

 1. Cannot '''accept Chat''' session: (AttributeError: 'NoneType' object has no attribute 'setStream')

 1. Change title of the chat window to '''Chat to remote@domain'''

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

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

 1. Display the Chat recipient in a '''tab at the bottom of the window'''

 1. In chat window, after entering a message the scrollbar '''moves up by itself'''

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

 1. When text is pasted, as opposed to typed on the keyboard, the '''pasted content''' is sent but not displayed

 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. '''Hide the empty scroll bar''' of the text input box

= Main interface =

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

 1. '''Remove account''' from lists when disabled in preferences  

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

 1. Display '''active account in bold text'''

 1. Remove the blue '''is-selected''' rectangle that appears around the active account on startup

 1. Mute button does not work

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

= Contacts =

 1. When searching contacts, do not display '''Add button''' if the contact already exists

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

 1. Add Group selector to add contact window

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

 1. When '''searching a contact''' and starting a session to it by double clicking: (NameError: global name 'row' is not defined)

= Preferences =

 1. '''Change audio devices''' does not enter effect unless rebooting

 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). Fo  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)

Hide the following settings, they should not be editable from GUI:

 1. global.chat.accept_types

 1. global.chat.accept_wrapped_types

 1. global.sip.ignore_missing_ack

 1. account.presence.subscribe_rls_services

 1. account.presence.subscribe_xcap_diff

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

= Engine =

 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

 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

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

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

 1. Show '''activity note''' on second row

= 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'''