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Welcome to a webinar about Ingate software version 4.7. This webinar is a listen only audio broadcast. Please feel free to pose any questions via webex chat, which will be answered in the Q&A session at the end of the presentation.
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Welcome to a webinar about Ingate software version 4.7 This webinar is a listen only audio broadcast. Please feel free to pose any questions via webex chat, which will be answered in the Q&A session at the end of the presentation. A conference bridge will be available after the session for more detailed technical questions
Janne Magnusson, Director of Development and Product Management
Ingate software release 4.7 • GA release version: 4.7.1 • Release day: March 23rd, 2009 • 186 new features, improvements and bug fixes • For customers with SW subscription the release upgrade available for download from: http://www.ingate.com/Upgrades.php • Release notes available at: http://www.ingate.com/Relnote.php?ver=471 • For customers without SW subscription contracts pls contact Ingate at sales@ingate.com
Ingate software release 4.7 • Applicable for: • 1180/18 • 1190/19 • 1200/20 (EOL Oct 2009) • 1450/45 • 1400/40 (EOL Jan 2010) • 1500/50 • 1550/55 • 1600/60 • 1880/88 • 1900/90 • Older models reached EOL and are officially not supported in 4.7
Release 4.6 and older • The release 4.6 will be supported until the release of 4.8 but no longer then September 23rd 2009. • Older releases are not supported.
Media Quality Scoring Quality for media received by Ingate is measured and reported in the CDR ticket over RADIUS.
Media Quality Scoring The following information are computed (separate for each direction) • packets sent • bytes sent • jitter • number of missing packets • maximum number of consecutive missing packets. • (estimated MOS value will be available in June -09) Key benefit: • Real time reporting and monitoring of voice quality • Enables preemptive actions to increase voice quality • Creates a record of all calls with quality information • Makes it easier to determine where quality issues occur
WAN SIParator in front of existing firewall Key Benefits • The SIParator can enforce Traffic shaping on the WAN link to prioritize VoIP traffic over Data for optimal voice quality • Connected directly to the LAN or via the firewall DMZ. • No configuration changes to the existing firewall • All data traffic are routed untouched to the existing firewall • Same SIP functionality as for all other SIParator configurations
WAN SIParator QoS Features (requires the Ingate QoS module) • Call Admission Control • Reject new calls when all available bandwidth is used • Modification of TOS (Type of Service) bits • Traffic prioritization • Reserve bandwidth available for VoIP • Prioritization of data traffic based on Service and TOS value • Works for both inbound and outbound traffic
WAN SIParator QoS Features • Configurable for each interface and direction • Dynamic, free bandwidth configured for SIP are used for Data when available • Separate category for emergency calls Emergency calls SIP and Media Available for Data Available bandwidth
Multiple Internet connections Key Benefits • Allows usage of a dedicated IP connection for VoIP only • The dedicated VoIP connection can handle all or a subset of the SIP traffic. • Remote users can use the ordinary Data connection to access the IP-PBX system
Multiple Internet Connectionsexample: different VLANs • Both pipes can be uses for SIP allowing remote users to connect to the enterprise VoIP system via the Internet
Multiple Internet Connectionsexample: Remote User • Both pipes can be uses for SIP allowing remote users to connect to the enterprise VoIP system via the Internet
Handles remote clients behind NATs with poor SIP support Key Benefits • Enforce Ingate Remote SIP Connectivity for all remote phones regardless of the NAT they are behind • Solves issues with many “SIP capable” home and hotel routers with rudimentary SIP support
Multiple IP addresses for SIP signaling Key Benefits • Ingate can use different IP addresses internally when communicating with the IP PBX. • Solves issues for some IP PBXes not accepting SIP Trunks and Remote Users from the same source IP (e.g. Asterisk, Nortel SCS and 3Com VCX) • One or two IP addresses are used on the external side of the Ingate
Configurable Display Name and P-Asserted-Identity on Accounts Key Benefits • Allows configurable Display Name and P-Asserted-Identity per SIP Account used for SIP Trunking • Increase interoperability
Handling of INVITE without offer IP-PBX Ingate ITSP 1: | |<---INV:A,B---| 2: |<----INV:B-----| | 3: |-----200:B---->| | 4: | |----200:B---->| 5: |-----re-I:---->| | 6: | |----re-I:B--->| 7: | |<---200:B-----| 8: |<----200:B-----| | 9: |-----ACK:B---->| | 10: | |----ACK------>| The Ingate B2BUA handles INVITEs and re-INVITEs without SDP and adds SDP where SDP-less INVITEs are unsupported. Key Benefits • Allows combinations of IP-PBXes and ITSP to interact that otherwise wouldn’t.
User Database and SIP Trunk accounts separated Key Benefits • Enables usage for user authentication via RADIUS and SIP Trunking accounts concurrently. • More intuitive and user friendly.
Other improvements • Proxy ARP (applicable for the Firewall only) • Reduced number of port changes for media • User Routing for those authenticated via RADIUS • Additional configuration parameters for IPSec VPN ….and more, for the full list of updates see: http://www.ingate.com/Relnote.php?ver=471
Presentation available for download including a recording of this webinar http://www.ingate.com/Webinar_Release_4.7.php.
Q&A • Chat in Webex now • Technical discussion after the call at: sip:conf2@ingate.com alt. 469-656-3573 x870 alt. +4613210870 • Customers with support contract send E-mail to: support@ingate.com • Customers without support contract pls contact sales@ingate.com