SIP SIMPLE Client SDK

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

= Critical issues =

Must be solved first.

 1. '''Audio sessions''' to do not work:

  * When making audio session: (AttributeError: 'Session' object has no attribute 'end_time') and (AttributeError: 'Session' object has no attribute 'on_hold')

  * Failed sessions remain indefinitely in the drawer

  * Voice calls do not update the status of the drawer

  * There is no microphone input, nothing is heard on the other side

  * Incoming session display different elements than outgoing, no session information or buttons are displayed

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

 1. When text is pasted, as opposed to typed on the keyboard, the '''pasted content''' is sent over the media session but is not displayed in the history window

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

 1. '''session duration''' for both incoming and outgoing is missing

 1. '''Expand the session frame horizontally''' to fill on the drawer to its full width and draw just one line between two voice calls

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

= Engine =

 * 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

 * 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

= Chat window =

 * Change title of the chat window to '''remote@domain'''

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

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

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

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

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

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

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

 * For each message that appears in the history window:

  * First 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

  * Then display the content of the message

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

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

= Contacts =

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

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

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

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

 * Show the '''Display name''' if set for the Contact, otherwise the SIP address

 * Add new '''contact window fields''': SIP address, Display name and Group

 * '''Delete the contact''' when pressing delete, show a confirm window

 * When '''searching a contact''' and starting a session to it: (NameError: global name 'row' is not defined)

= Preferences =

 * Change drop down box of general chapters to a tool bar with icons

 * Delete the '''little bar on top of the account list''', is redundant

 * '''Delete Enabled checkbox''' in the account section, leave only the checkbox in the account list

 * Properly '''align''' horizontally the Advanced settings

 * Display the '''bonjour account''' at the end of the list

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

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

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

 * Add a '''Reset option''' to reset advanced settings to default values

 * Cannot '''delete Bonjour''' account: (TypeError: cannot delete BonjourAccount instance with default id). This is correct, do  not allow the user to attempt to delete it. Bonjour can only be enabled/disabled but not deleted.

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

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

 * Allow ordering of accounts in the account list

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

u

{{{

2009-07-19 10:39:45.775 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.]

2009-07-19 10:37:41.634 Blink[26992:10b] Traceback (most recent call last):

2009-07-19 10:37:41.635 Blink[26992:10b]   File "/Users/adigeo/work/blink/build/Debug/Blink.app/Contents/Resources/BlinkAppDelegate.py", line 40, in applicationDidFinishLaunching_

2009-07-19 10:37:41.653 Blink[26992:10b]     self.backend.init(options)

2009-07-19 10:37:41.654 Blink[26992:10b]   File "/Users/adigeo/work/blink/build/Debug/Blink.app/Contents/Resources/SIPManager.py", line 108, in init

2009-07-19 10:37:41.672 Blink[26992:10b]     self._app.start(config_backend=config_be)

2009-07-19 10:37:41.673 Blink[26992:10b]   File "/Library/Python/2.5/site-packages/sipsimple/api.py", line 85, in start

2009-07-19 10:37:41.693 Blink[26992:10b]     trace_sip=settings.logging.trace_sip,

2009-07-19 10:37:41.705 Blink[26992:10b]   File "/Library/Python/2.5/site-packages/sipsimple/engine.py", line 75, in start

2009-07-19 10:37:41.747 Blink[26992:10b]     self._ua = PJSIPUA(self._handle_event, **init_options)

2009-07-19 10:37:41.749 Blink[26992:10b]   File "core.ua.pxi", line 66, in sipsimple.core.PJSIPUA.__init__ (sipsimple/core.c:23189)

2009-07-19 10:37:41.751 Blink[26992:10b]   File "core.lib.pxi", line 106, in sipsimple.core.PJSIPEndpoint.__cinit__ (sipsimple/core.c:6231)

2009-07-19 10:37:41.753 Blink[26992:10b]   File "core.lib.pxi", line 167, in sipsimple.core.PJSIPEndpoint._start_tls_transport (sipsimple/core.c:6884)

2009-07-19 10:37:41.754 Blink[26992:10b] PJSIPError: Could not create TLS transport: Error loading/verifying SSL CA list file (PJSIP_TLS_ECACERT)

2009-07-19 10:37:41.754 Blink[26992:10b] None

}}}

Hide the following settings:

 * global.chat.accept_types

 * global.chat.accept_wrapped_types

 * global.sip.ignore_missing_ack

 * account.presence.publish_interval

 * account.presence.subscribe_interval

 * account.presence.subscribe_rls_services

 * account.presence.subscribe_xcap_diff

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

= Sessions =

 * Center the '''drawer title''' and make it look like a title bar

 * Show the '''drawer maximized''' horizontally

 * Center align '''Hangup All and Conference buttons'''

 * 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

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

 * Highlight the active session with a '''bold''' border

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

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

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

 * We need a solution to enter DTMF tones to an audio session, now the keyboard focus is on the search bar

= Presence bar =

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

 * Show '''own photo''' to the left

 * Show '''Display name''' next to the photo on top row

 * Show '''presence activity''' next to name on top row

 * Show '''activity note''' on second row

= Debug window =

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

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

= GUI settings =

Remember the following settings between restarts:

 * '''Main window''' size and screen position

 * '''Debug window''' and the selected tabe

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

 * SIP '''accounts order''' list

= Middleware changes =

 * Move '''account.outbound_proxy''' to '''account.sip.outbound_proxy'''

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

 * 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

 * Append dummy to the lists of available audio devices

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

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

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

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

 * Delete '''account.audio''' section