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diff --git a/TODO b/TODO
index 99d2d310..bf3e4e0f 100644
--- a/TODO
+++ b/TODO
@@ -1,132 +1,133 @@
Python PJSIP bindings
---------------------
- Handle UPDATE for re-INVITE
- High level Invitation class that handles SDP as well
- Cleanup Registration, Publication and Subscription to behave more like Invitation
- Means to query RTP statistics
- Handle sending and received of multipart bodies for Invitation
- Fix high CPU usage
- Properly decrease reference counts of pjsip_tx_data structs
- Split up .pyx file
- Use STUN to determine and log the type of NAT we are behind (PJNATH_NAT_DETECT)
- Add option to use the STUN response for Contact header (default false)
- Fix remainig TODOs for PUBLISH/REGISTER
- Make all init_options passed to PJSIPUA attributes, set local_ip at runtime
- Add feature to send OPTIONS method and parse the replies
- Add ability for application to request handling of arbitrary methods such as
OPTIONS, REGISTER and REFER, SUBSCRIBE (but not INVITE or NOTIFY...) and have it reply.
- Registration requestion URI should be settable by application
- Remove any DNS or lookups from pjsip (lookup is always done outside PJSIP)
- Write API documentation
Audio
-----
+- Add option and command to mute the microphone (useful for listen in conferences)
- Play a soft beep tone while on hold every 30 seconds
- Play a beep after we closed a session
- Play a special beep when receiving a call while engaged in another call
- Implement Automatic Gain Control in PJSIP media engine
- zRTP
Command line clients
--------------------
Logging, based on trace_x configuration options:
- Append PJSIP messages to .sipclient/log/user@domain/pjsip_trace.txt
- Append MSRP messages to .sipclient/log/user@domain/msrp_trace.txt
- Append XCAP messages to .sipclient/log/user@domain/xcap_trace.txt
- Add timestamp and packet counters as for SIP trace for all other prorocols
- Toggle sip logging at run time
- Add key command to start a session when starting in receiving mode (c)
- Create sip_subscribe_conference_info script
- xcap_pidf_manipulation script
- Publish service and status when starting/stopping rtp/msrp session script
- Set the shell return code to 0 (success) for 2XX, 3XX and 1 (failure) for
other codes. For invite scripts the code is based on the response for the
initial INVITE, not for the BYE If RTP was not received with 6 seconds after
call setup, consider the call has failed.
- Add parameter to hangup after a number of seconds for sip_audio_session
- Handle multiple incomming sessions (waiting for the re-INVITE)
- Conference established sessions
- Handle 301/302 redirect by prompting the user to accept/reject the redirection
- Combine msrp/rtp in one script
Applications
------------
- MSRP focus
- MSRP mixer
- Conference-info event package
- MSRP multiparty chat
- Is-composing payload
- Lookup capabilities for SIP (RFC3263)
- Message-summary payload
- xcap-diff payload
- Buddylist library
- Bonjour discovery
- Lookup for ENUM
- Write documentation
Sessions
--------
- Comfort noise generator
- Video (H.264)
- Real time text over RTP (RFC4103)
- Desktop sharing (VNC protocol)
- Bit pipe over MSRP
End-user experience
-------------------
- Mute other applications that access the soundcard on receiving INVITE
Enrollment
----------
- End-point certificate generator
- Account creation
- UA-profile XCAP application
Middleware
----------
- rename lookup_srv to rename_rfc3263, add NAPTR lookup to it and pass a
list of supported transports by the SIP library to select from (default is
udp).
- Configuration framework based on ua-profile event package
- Event bus
- Environment detection: ambient and geo-location detection
User interfaces
---------------
- Stand alone library
- Command line clients
- GUI
Packaging
---------
- MacOSX installer
- Windows installer
Others
------
- Add doc strings to all classes and functions
- IPv6 support
- Survive local IP address changes
- P2PSIP overlay

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