Upgrades

Release notice for Ingate Firewall® 6.3.3 and Ingate SIParator® 6.3.3

The new version can be found here

Release notice for Ingate SIParator(R)/Firewall(R) 6.3.3

Release name: Ingate SIParator(R)/Firewall(R) 6.3.3
Release date: June 21, 2021

The new version and documentation can be found at:
https://account.ingate.com/

This is a bug fix release with important stability and security improvements.
We recommend everyone to upgrade.


Note: The first time a unit is upgraded from 6.2.x to 6.3.x it will EMPTY
the persistent log partition (any subsequent down->upgrades will not require
this). Meaning that *all logs* will be lost. This inconvenience is necessary
for improved file system support. If you want to save the unit logs prior
to the upgrade, use the "Export the Log" functionality found in the tab
"Logging and Tools -> Display Log". Exported logs cannot be imported in the
new version.


* SIP *****************************************************************

*** Fixed buffer over-read when generating responses.

*** Fixed use-after-free during SDP media processing when
    Reuse port numbers.

*** Fixed Reuse port numbers together with the B2BUA.

*** Fixed 'Use session identifier when comparing endpoint SDPs'
    together with the B2BUA.

*** SIP Trunk: Make REGISTERs honor From header domain.

*** SIP Trunk: Fixed route incoming based on To header.

*** SIP Trunk: Fixed attended transfer when forwarding REFER.

*** B2BUA: Don't use UPDATE as refresh method if not allowed by config.

*** B2BUA: Relay delayed re-INVITE after UPDATE.

*** B2BUA: Fixed ACK/PRACK w/ SDP when hiding Record-Route.

*** B2BUA: Forward a retry ACK w/ SDP answer to retransmitted 200 offer
    (o/a type-2 scenario), instead of empty ACK.

*** WebRTC: Improve ICE check in 0.0.0.0 hold handling.

*** WebRTC: Fixed ICE handling when no ICE candidates.

*** WebRTC: Fixed DTLS handling when changing media port.

*** WebRTC: Remove WebSocket registrations on connection close.

*** Transcoding: Fixed DTMF during codec transcoding.

*** Transcoding: Always pass through telephone-event on codec transcoding.

*** Transcoding: Fixed adding OPUS codec parameters during transcoding.

*** Transcoding: Remove trailing semicolon in fmtp.

*** Transcoding: Fixed SRTP transcoding together with B2BUA ICE pass-through.

*** Fixed 2543 hold with B2BUA.

*** Fixed ACK/PRACK w/ SDP but wo/ Route (sip.use-first-response-offer).

*** Fixed parsing of session-param in crypto attributes.

*** Fixed media destination update when the media proxy is enabled but
    not needed.

*** Keep request-uri parameters when doing DNS override with modify ruri.

*** SIP didn't always get a gw on startup/failover when using multiple
    default gateways.

*** Do not configure Remote SIP Connectivity if no licenses.


* Other ***************************************************************

*** Fixed DHCPv6 prefix length.

*** Include Standby Unit Access Relay ports in the used-ports check.

*** Q-TURN server media streams were not set up properly in all scenarios.