SIP SIMPLE Client SDK

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

= Sessions =

 1. Increase '''font size''' with one unit 

 1. Align '''session duration counter''' to the left

 1. Center the '''drawer title''' Voice Calls when the drawer resizes and make it look like a title bar

 1. '''Maximize drawer''' when it opens the first time

 1. Center align '''Hangup All and Conference buttons''' while resizing the drawer

 1. '''Expand the session frame horizontally''' to fill on the drawer to its full width

 1. Draw just '''one line''' between two sessions

 1. Display the '''negotiated codec''' next to audio and use the font colors from email snapshot

 1. '''Cannot terminate''' an outgoing active session

 1. '''Cannot cancel''' an outgoing call

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

 1. Ended session must '''disappear from the drawer''' after 5 seconds

 1. Incoming sessions '''do no cancel''' and the session fails when attempting to reject/accept it later, pop up window must disappear when an incoming call has been cancelled 

 1. Active sessions must '''shuffle to drawer top'''

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

 1. Closed session keep updating the session '''duration counter'''

 1. Answered sessions to not display audio '''session information'''

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

 1. When the active session is selected with the mouse, what is typed must be transmitted as DTMF tones

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

 1. Highlight the active session with a '''bold''' border

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

 1. If all sessions have ended, the '''drawer must be closed'''

 1. Add an item to the '''Session menu''' to show the drawer, if it was closed

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

= Chat window =

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

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

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

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

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

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

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

= Main interface =

 1. When select a different account, mark it as '''default'''

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

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

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

 1. '''Hide display name''', show only the SIP address in the account selection

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

 1. '''Replace''' 3333 with 3333@sip2sip.info and 4444 with 4444@sip2sip.info

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

 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: (NameError: global name 'row' is not defined)

 1. Mute button does not work

= Preferences =

 1. Changing audio devices does not enter effect unless rebooting

 1. When enable a new account, the main interface displays '''no account selected''' anymore

 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. In advanced sections, display '''Enabled:''' on the left side of the checkbox instead of the right

 1. Delete the '''little title bar on top of the account list'''

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

 1. Hide the vertical and horizontal '''scroll bars''' when no scrollbar is needed

 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. 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

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.publish_interval

 1. account.presence.subscribe_interval

 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 tabe

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

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

= Middleware changes =

 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 certificate before starting the engine

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

 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 you must use the default standard port of 2855, 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 '''account.audio''' section