UpgradesRelease notice for Ingate Firewall® 4.10.1 and Ingate SIParator® 4.10.1
| Release name: |
Ingate Firewall® 4.10.1
Ingate SIParator® 4.10.1 |
The new version can be found here
Release notes for Ingate Firewall(R) 4.10.1 and Ingate SIParator(R) 4.10.1
Release name: Ingate Firewall(R) 4.10.1
Ingate SIParator(R) 4.10.1
Release date: June 28, 2012
The new version and User Manuals can be found at:
http://www.ingate.com/Upgrades.php
This is a major release with many new functions, features and fixed issues.
We recommend everybody to upgrade.
* New SIP Related Features *********************************************
*** Gin
*** Support Globally Routable UA URI (GRUU)
New setting: "SIP Services"->Interoperability->"Resolve public GRUU locally"
[Tracking ID: 1263]
*** Method-based SIP routing
New "methods" parameter in Dial Plan Forward To Reg Expr field
[Tracking ID: 1873, 4728]
*** Support MSRP for session-mode instant messaging
[Tracking ID: 1895]
*** Support Path header (RFC 3327)
New setting:
"SIP Services"->"Interoperatbility"->"Always add Path Header in REGISTERS".
[Tracking ID: 1938]
*** Check route-set within dialogs.
New setting: "SIP Services->Interoperability->Fix Bad Route set"
[Tracking ID: 2261]
*** Setting for allowing media encryption key change in reINVITE
New setting:
"SIP Services"->"Sessions and Media"->"Media Configuration"->
"Support forked media streams"
[Tracking ID: 3779]
*** Separate RFC 4904 SIP trunk parameters into Remote and Local
New settings:
"SIP Trunks"->"Trunk 1"->"Local Trunk Group Parameters (RFC 4904)"
"SIP Trunks"->"Trunk 1"->"Local Trunk Group Parameters usage:"
[Tracking ID: 4483]
*** Treat all 5xx and 6xx responses except 503 from monitored SIP servers as success
[Tracking ID: 4662]
*** New interoperability settings.
"SIP Services"->Interoperability->"Disable re-INVITEs"
"SIP Services"->Interoperability->"Disable Supported Header in B2BUA"
"SIP Services"->Interoperability->"Force RTP Packetization Time"
"SIP Services"->Interoperability->"Sequential Register Delay"
[Tracking ID: 4673, 4674, 4675]
*** New Radius attributes added and a new setting making it possible
to send accounting data only from specific interfaces
New setting:
SIP Traffic->Authenticaton and Accounting->Radius Accounting->Include media statistics.
[Tracking ID: 4745]
*** Support for INVITE with Replaces also for early dialogs,
i.e. calls which are in the calling state and not answered yet
[Tracking ID: 4710]
*** Support for URI parameters, for example ;user=phone, in more settings
on the SIP Trunk page: "SIP Trunk Service Provider Domain", "From domain"
and "PBX Domain Name"
[Tracking ID: 4712]
*** Gin Registration for Multiple Phone Numbers (RFC 6140)
Required by SIP Connect 1.1 registration mode
New setting on SIP Trunk pages
[Tracking ID: 4713]
*** Support for using display name (dname) in Generic Header Manipulation,
needed to allow rewriting the From URI host on SIP Trunk pages,
while keeping the display name
[Tracking ID: 4716]
*** Send "pong" if a "ping" is received, RFC 5626
New setting in (SIP Services->Remote SIP Connectivity->NAT keepalive method)
[Tracking ID: 4718]
*** Recognize inbound requests that arrives on client connections
with port number in request-uri that is not a server port
[Tracking ID: 4719]
*** Forward Diversion header in 3xx response from the B2BUA
New setting on the Interoperability page:
"Forward headers in 3xx responses in the B2BUA"
[Tracking ID: 4720]
*** Allow regex subexpressions in "From PBX number/user" and
"Forward to PBX Account" on the SIP Trunk page
[Tracking ID: 4725]
*** Support from header replacement using regexp field in
the Dial Plan also when forwarding to a trunk
[Tracking ID: 4727]
*** Option to start local ringback when call makes progress.
New option "Play local ringback when new target rings or makes progress"
added to setting "Local Ringback Played at Call Transfer" on page
SIP session and media
[Tracking ID: 4729]
*** Option to limit networks and computers from which
the Ingate accepts SIP Signaling
New setting "SIP Signaling Access Control" on SIP Services -> Basic
[Tracking ID: 4730]
*** Maximum number of simultaneous calls per Trunk Line
New setting:
"Maximum number of simultaneous calls per Trunk Line" on the SIP trunk page.
[Tracking ID: 4737]
*** Support SIP Connect registration backoff timers when registration fails
New setting:
"SIP Traffic"->"SIP Accounts"->"Use exponential backoff timer on retry"
[Tracking ID: 4741]
*** Support self-made GRUU as URI encoding options
New URI encoding options in SIP Services Interoperability
[Tracking ID: 4742]
*** Support creation of certificates containing SubjectAltName
[Tracking ID: 4743]
*** Support adding of intermediate certificates to an existing
private certificate
[Tracking ID: 4744]
*** Support authorization of Trust Network based on TLS certificate (Mutual TLS)
New setting "SIP Traffic"->"Authentication and Accounting"->
P-Asserted-Identity->"Trusted Domains".
[Tracking ID: 4751]
*** Support use of FQDN in Contact and Record-Route of outgoing SIP messages
Should be used if the TLS peer certificate only contains the FQDN,
and not the IP address in CN and/or SubjectAltName
New column:
"Use CN FQDN" in "SIP Services"->"Signaling Encryption"->
"TLS Connections On Different IP Addresses"
[Tracking ID: 4753]
*** Option to terminate transferor when transfer target plays early media
New setting: "SIP Services"->"Interoperability"->"Terminate
Transferor on 183"
[Tracking ID: 4762]
*** Enhancement to SIP survival
Added survival methods: Swisscom and Display name
[Tracking ID: 4773]
*** Ignore old ICMP errors when monitoring SIP servers (OPTIONS)
[Tracking ID: 4775]
*** Add predefined trusted groups
Trusted groups "authenticated", and "friendly" added to Trust Domains
settings, and can be used in forward to regexp in the dial-plan.
[Tracking ID: 4784]
*** Improved call-id detection for B2BUA invocation
From manipulation did not work in PBX Line, forward-to field
[Tracking ID: 4785]
*** Support for tel URI in From header when matching in dial plan and trunk
[Tracking ID: 4787]
*** Added GUI option to use same URI encoding for registration
and other messages.
[Tracking ID: 4789]
*** Sending periodic 1xx responses to extend cancel timeout
[Tracking ID: 4795]
* Other New Features ***************************************************
*** Max sip traversals increased on hardware 1660 to 1000
[Tracking ID: 4763]
* Fixed SIP Related Issues *********************************************
*** Improved handling of in-dialog messages with incorrect
request URI:s in calls with large initial INVITE
[Tracking ID: 4599]
*** Fix translation of Contact in 200 OK to register when in LAN SIParator mode.
[Tracking ID: 4707]
*** Fix SIP signalling source address after failover
[Tracking ID: 4721]
*** Allow UTF-8 characters in display name accessed with .dname
variable substitution
[Tracking ID: 4738]
*** Support media transcoding for incoming INVITE without SDP offer
[Tracking ID: 4739]
*** Improvements to media encryption to allow more call scenarios
[Tracking ID: 4750]
*** Fix missing angle bracket in To header when rewritten for REGISTER response
[Tracking ID: 4761]
*** Correct SDP used in session refresh re-invite after previous
offerless INVITE (SDP in ACK)
[Tracking ID: 4764]
*** Ignore failure ECONNREFUSED if OPTIONS monitoring to get at least
one OPTIONS out on the network for each retried transaction.
*** DMZ/LAN SIParator no-way media in spiral scenario
[Tracking ID: 4788]
*** Set remote target when receiving 2xx even if dialog already exist
Fixes a problem with a lost ACK
*** Added log messages for call to unregistered users.
*** Stop search if matching a unregistered user
Previously dial plan and everything else was traversed,
but still responding 480 when it was a call to unregistered user
* Fixed Other Issues **************************************************
*** IPSec VPN stability fixes. Updated to Openwan 2.6.37 + patches.
[Tracking ID: 4782]
*** General stability improvement
[Tracking ID: 4740]
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