SIP SIMPLE Client SDK

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

= Audio =

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

 1. Rename menu item Session with Sessions

 1. Play the disconnect tone if the session fails to establish

 1. When dial a second call while in another call, if the user clicks on the previous call before the new session is established the session info is broken

 1. New incoming sessions, the cancel window shrinks in size after a few clicks or cancels from remote (still not solved)

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

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

 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 =

 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 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

 1. Add file transfer function when click on File transfer

= Main interface =

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

 1. If drag window size when collapsed, it does not resize correctly the same collapsed position

 1. When selecting bonjour as account, hide the main buddy list. The idea with bonjour is that other clients on the local LAN will show up in the list

 1. Show only errors in the log drawer

= Contacts =

 1. Add the following per contact settings and fields in the '''Add contact window''':

      * Double click action: Drop down box: Audio session, Chat session

      * Subscribe to Presence (checkbox, default true)

      * Subscribe to Dialog (checkbox, default false)

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

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

 1. Append domain part if missing. E.g. when adding abcd@chatserver we must still append the domain name after chatserver

 1. Rename menu item Contact with Contacts

 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' (or is it a middleware setting?)

 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 supported by PJSIP 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.

= 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

= Presence bar =

 1. Show '''Display name''' next to the photo on top row. When edit the name save account.display_name setting

 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

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

 1. Acquire photo from webcam and save it

= Debug =

 1. Add colors to SIP trace content

 1. Print DNS lookups and results in the SIP trace window formated like in sip_audio_session script

 1. High CPU usage

= GUI =

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

 1. Save position of Debug window and its checkbos settings