Fixing the development branch #163
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…ginal
headers.
[ADD] Added `__gen_uri` in `SIPClient` for generating URIs for headers.
[ADD] Added better error handling for database issues in `VoIPSocket`.
[ADD] Added `get_database_dump` in `VoIPSocket`.
[CHANGE] Renamed `SIPClient._gen_from_to` to `__gen_from_to_via_request`.
[CHANGE] Renamed `SIPClient._gen_user_agent` to `__gen_user_agent`.
[CHANGE] `SIPClient.invite` now also returns a `VoIPConnection`.
[CHANGE] Changed signature of `SIPClient.trying_timeout_check` to include the
`VoIPConnection` so the timeout check can use the same dialog.
[CHANGE] Changed `VoIPCall` to accept its `VoIPConnection` so it can interact
with its dialog.
[FIX] Fixed SIP.recv error causing a crash it data was not received at all.
[FIX] Fixed issue with `SIPClient.gen_invite` using malformed headers.
[FIX] Fixed all instances of not capturing the return from `SIPClient.sendto`
causing multiple `VoIPConnection`s to be created for the same dialog.
[FIX] Fixed issue with `VoIPSocket.determine_tags` returning a null local tag.
[FIX] Fixed issue where `pyVoIP.VoIP` was not installing.
[REMOVE] Removed commented code.
[ADD] Added ability to close a `VoIPConnection`. [CHANGE] Cleaned up register and deregister. [CHANGE] Changed `trying_timeout_check` to `__receive` and changed usage. [CHANGE] `VoIPConnection.recv` now raises a timeout error for UDP. [FIX] Fixed listening table not updating properly on receiving the remote tag.
[ADD] Added basic Network Address Translation (NAT) feature. [ADD] Added __gen_via in SIP.client. [ADD] Added __gen_contact in SIP.client. [ADD] Added some IPv6 handling. [CHANGE] Changed Docker start script to PowerShell. [FIX] Fixed not receiving replies due to network routing. [FIX] Fixed registration tests failing due to max contacts. [FIX] Fixed uri's not denoting sips when using TLS [FIX] Fixed Via headers from other hosts being incorrectly changed. [FIX] Fixed timeout errors being raised incorrectly. [FIX] Fixed deregister_connection for TCP and TLS sockets. [FIX] Fixed all registration tests.
[FIX] Fixed flake8 error
[ADD] Added receiver thread to VoIPCall for handling future requests. [ADD] Added SIP State DB location global variable for debugging. [ADD] Added AddressType enum to support NAT-based determine_tags functions. [ADD] Added check_host function in NAT to check the AddressType of a host. [ADD] Added filtering to func test if on Windows. [ADD] Added helper variables to ensure func tests run correctly on linux. [CHANGE] Cleaned up response checks in SIP client. [CHANGE] Broke up VoIPCall init into multiple functions. [CHANGE] Changed determine_tags to use NAT. [CHANGE] Changed SIPMessage to always raise SIPParseError upon any error. [CHANGE] Added pprint option to get_database_dump. [FIX] Fixed port not showing when not 5060 in Contact header. [FIX] Fixed invites not working for UDP. [FIX] Fixed missing NSD update in VoIPPhone stop. [FIX] Fixed messages not sending when they are a UDP response. [FIX] Fixed UDP socket recv function not receiving messages missing tags. [FIX] Fixed SQL connection issues.
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Fixes #76 |
[FIX] Fixed multiple SQL issues. [FIX] Fixed line length. [FIX] Fixed __get_connection.
[CHANGE] Added VoIPConnection to callback in VoIPPhone
[CHANGE] Changed _callback_MSG_Invite to use VoIPConnection instead of creating
a new dialog
[CHANGE] Changed _create_call to be PEP8 compliant
[CHANGE] Moved TCP/TLS recv and UDP recv code to different functions for better
readability and maintainability
[FIX] Fixed errors if calling RTP.stop when not started
[FIX] Fixed issue with code not responding to SIP_STATE_DB_LOCATION
[REMOVE] Removed call re-negotiation code as it seemed to not work as intended
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I can use development branch right? or is it broken? |
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Many major issues have been fixed, merging so development will become usable again. |
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Fixes #162