SIP SIMPLE Client SDK

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

= Audio sessions =

 1. Set SIP User Agent name on start to '''blink-<version>'''

 1. With every new incoming session the cancel window shrinks in size (not solved)

 1. Do not open session drawer when the main interface is collapsed 

 1. '''Mute''' microphone when pressing mute button

 1. Add  a '''record button''' left to Hold button. When pressed toggle recording the audio session

 1. Display RTP '''packet loss''' with red above the session buttons when > 0.5 %, sample every 10 seconds

 1. Display '''Ringing...''' when get ringing indication from remote party

 1. Closing many active sessions cause many tones to be played back. Throttle tones playback

 1. Call '''set_audio_devices()''' when changing the audio device or tail_length in global Preferences

 1. Calls made with bonjour account that do not resolve in DNS do not end, try dial abcd

 1. Add a '''triangle button''' to expand vertically the audio session frame with extra room for audio session information:

    * Local RTP: 130.129.86.171:64369

    * Remote RTP: 81.23.228.129:58228

    * Remote UA: Asterisk PBX 

    * Packet loss=0.0% (must be updated every 5 seconds) 

    * Jitter RX/TX=0/21 ms (must be updated every 5 seconds)

= Chat sessions =

 1. Replace Connect/Close buttons with a single button

 1. When disconnected print system message: '''MSRP chat session terminated (reason)''' <- not working

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

 1. To close a chat tab add a '''small x button''' to it

 1. When using a relay during stream initialization print system message: '''MSRP session reserved at relay host:port'''

 1. When click on the Audio button, use add_stream to append an audio stream to existing chat session

 1. If an audio stream exists part of the session, add the audio control buttons '''Record|Hold|Terminate'''

 1. Use the '''same look and feel''' for the toolbar buttons as the main interface (greyish buttons) 

 1. Implement the is-composing payload parser in middleware, see http://sipsimpleclient.com/ticket/40

 1. Display '''is-composing''' payload

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

 1. '''Auto-accept''' chat only sessions when caller is in the contacts list  (tricky because of aliases). Use the SIP URI user@domain from the From header of the INVITE to match the contact

= Main interface =

 1. '''Bonjour account''' does not show/hide when activate/deactivate in preferences

 1. Expand/collapse brings the buddy list back in the wrong position, 20 pixels too high over own photo

 1. Start audio session when double click on the search results

 1. Show search results from the system '''address book'''

= Log drawer =

 1. Lower the font size and use red font in case of errors

 1. Display more useful messages (e.g. NAT type, DNS lookups, session status)

= Contacts =

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

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

 1. Rename menu item Contact with Contacts

 1. Rename menu item Session with Sessions

 1. Add edit Contact in the Contact menu

 1. Hide '''Add contact''' if contact exists / does not work yet

 1. If search when gui is collapsed and then expand then the Add contact button overlaps other text in the window

= Preferences =

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

 1. Audio device must also list 'Default system input ' and 'Default system output'

 1. Do not allow dragging of window size with lower left corner

 1. '''bonjour account''' has only Display name as main property

 1. Hide the minus button for Bonjour, it cannot be deleted

 1. When password field is changed, re-Register the account 

 1. When Register flag is toggled for account, (de)Register the account accordingly

 1. Codec lists must be an ordered list with at least one active codec

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

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

 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. 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.

= Conferencing =

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

Start conference as a mixer

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

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

 1. Disable the hold buttons of the conferenced sessions

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

Stop  conference as a mixer

 1. Set the conference bridge to stop mixing audio

 1. Send a re-INVITE to disable is-focus attribute of the Contact header

 1. Enable the hold buttons

 1. Draw the selection rectangle around the selected session

New sessions as a mixer

 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

Conference participant

 1. SUBSCRIBE to the conference event package when receiving a re-INVITE with isfocus true

 1. Expand vertically the session frame to display the names of the participants received in subsequent NOTIFY that contain the list of the participants in the conference

= Audio history =

 1. Use the same session drawer to display previous sessions

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

   1. Missed calls (incoming un-answered sessions must display the SIP address in red) 

   1. Duration line must display also the Start time: HH:MM:SS (YYYY-MM-DD HH:MM )

   1. Instead of audio session information display: Incoming|Outgoing

   1. Replace the current session buttons with a callback button

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

 1. Show most recent sessions on top

= 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. 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:

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

= Menu structure =

 1. File

 1. Edit

 1. View

  1. Audio sessions drawer 

  1. Previous audio sessions

  1. Previous chat sessions

 1. Audio

  1. Output device selection

  1. Input device selection

  1. Alert device selection

  1. Mute (mute mic input)

  1. Silent (mute output)

 1. Presence

  1. Activity

   1. Available

   1. Phone call

   1. Meeting

   1. Lunch

   1. Dinner

   1. Do not disturb (when enabled reject automatically all audio calls)

   1. Travel

   1. Offline

   1. Holiday

 1. Contacts

  1. Add contact

  1. Delete contact

  1. Edit contact

  1. Add group

  1. Delete group

  1. Edit group

 1. Sessions

  1. New audio session ...

  1. New chat session ...

  1. New multi party chat session ...

  1. Close all audio sessions

  1. Conference audio sessions

= Presence bar =

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

 1. ~~ Show '''own photo''' to the left ~~

 1. Acquire photo from webcam and cache it

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

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

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

   1. Available

   1. Phone call

   1. Meeting

   1. Lunch

   1. Dinner

   1. Do not disturb

   1. Travel

   1. Offline

   1. Holiday

Add a Presence menu item before Session.

= 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

= 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. Enable logging of notifications to file

 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. 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

 1.  Add a new setting '''account.chat.server''' default = chatserver.domain

= General =

 1. ~~Enable logging based on log settings~~

 1. ~~'''Set global.user_data_directory''' to ~/Library/Application\ Support/Blink/~~

 1. Scratchy noises when playing tone generator tones

 1. Crash when more than 32 tones in the tone generator queue

= IP connectivity checks =

If the Internet connectivity goes up and down the client must adjust to this

by taling appropiate actions. It is not enough to detect only what kind of

NAT we are behind but also what kind of IP connectivity we have.

== No IP network ==

Posible checks:

 * No real IP

 * DNS is down

 * No default route

== LAN without Internet ==

(e.g. wifi hotspot before login)

Posible checks:

 * Ethernet is up

 * DNS is up

== Internet access ==

Posible checks:

 * Ethernet is up

 * DNS is up

 * STUN is up

 * NAT type